From tculjaga at gmail.com Sun Nov 1 00:37:20 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sun, 1 Nov 2009 08:37:20 +0100 Subject: [Freeswitch-users] mod_t38gateway In-Reply-To: References: <65d96fc80910301452x2a831733h92b05861d6c94123@mail.gmail.com> Message-ID: <65d96fc80911010037g64c207dat2face4fca8de041a@mail.gmail.com> i tought so :PP T. On Sun, Nov 1, 2009 at 6:34 AM, Michael Jerris wrote: > This is a non working module, just a shell for development. > > Mike > > On Oct 30, 2009, at 5:52 PM, Tihomir Culjaga wrote: > > > does anybody know how does it work and how to use it in a dialplan? > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/6dc0ef76/attachment-0001.html From tculjaga at gmail.com Sun Nov 1 00:49:29 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sun, 1 Nov 2009 08:49:29 +0100 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179 In-Reply-To: References: <87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com> <87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com> <509453DC-57BF-43D0-B704-FF0F2BB58EC0@jerris.com> <87f2f3b90910312224s6968f278i86e35469c781d60c@mail.gmail.com> Message-ID: <65d96fc80911010049w65d59a34i615462f7d6f229a9@mail.gmail.com> and it works :P On Sun, Nov 1, 2009 at 6:38 AM, Michael Jerris wrote: > see rupa's explanation below. > > > On Nov 1, 2009, at 1:24 AM, Michael Collins wrote: > > How would you do an expression like: if $x < 24 in a condition tag? Just > curious. I would like to make sure that is properly documented. > -MC > > >> >> >> On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker wrote: >> >>> ${cond(${myvar} > 15 ? ERR : OK)} >>> >>> >>> If both sides of the comparison operator are numeric then it does >>> numeric comparison otherwise it does lexical string comparison. >>> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/a83e3912/attachment.html From ivan at myrvold.org Sun Nov 1 02:40:05 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Sun, 1 Nov 2009 11:40:05 +0100 Subject: [Freeswitch-users] SIP provider with extern rtp server In-Reply-To: <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org> References: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com> <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org> <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org> <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org> Message-ID: No one have any idea why this is not working? I have combed through the log, but couldn't find any clue there. Incoming calls from my sip provider is working perfect, but for outgoing calls it looks like Freeswitch is not letting the incoming rtp to the local sip phone. Ivan On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote: > Yes, now I got a more detailed trace. Thank you for helping me with > this. > > A new pastebin at http://pastebin.freeswitch.org/10905 > > Ivan > > Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable: > >> fsctl loglevel debug >> console loglevel debug >> sofia profile internal siptrace on >> sofia profile external siptrace on >> sofia loglevel all 9 >> ^^^^^^^^^^^^^^^^^^^^^ >> >> Then run your call, then do this: >> >> sofia loglevel all 0 >> sofia profile external siptrace off >> sofia profile internal siptrace off >> fsctl loglevel warning >> console loglevel warning >> >> On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold >> wrote: >>> I have already set debug to 9, on both profiles. >>> >>> Ivan >>> >>> >>> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable: >>> >>>> See that 200 OK that keeps coming in over and over and over and >>>> over >>>> again? That's because they never received your ACK. If you can >>>> turn on >>>> sofia loglevel to 9 and then watch where you send the ACK, you will >>>> probably have your answer to why the other system did not receive >>>> it. >>>> If you're still not sure what's going on, post another pastebin >>>> with >>>> sofia loglevel set to 9. >>>> >>>> >>>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold >>>> wrote: >>>>> Oh, what happened to it? >>>>> Anyway, here is a new pb: >>>>> http://pastebin.freeswitch.org/10867 >>>>> Ivan >>>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins: >>>>> >>>>> >>>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold >>>>> wrote: >>>>>> >>>>>> Here is a debug log from a call from an internal phone out to an >>>>>> external (my iPhone with nbr 91316356): >>>>>> http://pastebin.freeswitch.org/108578 >>>>>> >>>>>> Ivan >>>>>> >>>>> Uh... you wanna try that PB number again? >>>>> -MC >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch- >>>>> users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch- >>>>> users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>> -- >>>> Eliot Gable >>>> >>>> "We do not inherit the Earth from our ancestors: we borrow it from >>>> our >>>> children." ~David Brower >>>> >>>> "I decided the words were too conservative for me. We're not >>>> borrowing >>>> from our children, we're stealing from them--and it's not even >>>> considered to be a crime." ~David Brower >>>> >>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >>>> live; not live to eat.) ~Marcus Tullius Cicero >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>>> users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Eliot Gable >> >> "We do not inherit the Earth from our ancestors: we borrow it from >> our >> children." ~David Brower >> >> "I decided the words were too conservative for me. We're not >> borrowing >> from our children, we're stealing from them--and it's not even >> considered to be a crime." ~David Brower >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >> live; not live to eat.) ~Marcus Tullius Cicero >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > From yehavi.bourvine at gmail.com Sun Nov 1 06:24:18 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 1 Nov 2009 16:24:18 +0200 Subject: [Freeswitch-users] Rejecting a call from JavaScript Message-ID: Hello, We would like to handle an incoming call to a busy phone according to user's prefference: Some want waiting call, some want to just reject the call, and others want to send the call to voicemail. We have a small JavaScript which tests the status of the destination and the user's will and tries to act accordingly. Our problem is how to send busy. I tried session.hangup("USER_BUSY") but it always sends "temporary unavailable" which causes the orignator to think that the destination is out of order. What is the correct way to do so? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/c428b1bd/attachment.html From diego.viola at gmail.com Sun Nov 1 06:37:26 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 1 Nov 2009 14:37:26 +0000 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <200910312224.47236.chris@cloudtel.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> Message-ID: <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> Don't put me on the same leval as DelphiWorld please, I was just curious about how this SIP stack compares to sofia. Diego On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: > My favorite part of this 'civilized' discussion on IRC was when DelphiWord and > diegoviola sat around tryin to take the piss outta stkn on this issue for > seemingly no reason. Thanks for making the channel a cool place, guys ;) > > On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: >> Anthony Minessale a ?crit : >> > Meftah, >> > Feel free. >> > >> > thanks >> > >> > P.S. >> > >> > STKN was the guy who made the first mod_pjsip for FS that we abandoned >> > years ago. So you should believe him. >> > Both him and I agreed it was not working out. ?So if you don't believe >> > me, find out for yourself. >> > anthony, why i don't believe ?you? >> >> never say that. >> i believe you and all Freeswitch Staf and thank you and to all >> Freeswitch Staf. >> >> > On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb > > > wrote: >> > >> > ? ? hi Anthony >> > ? ? i agry >> > ? ? i say that because STKN hate all my suggestions. >> > ? ? about pjsip, i will contribute aditional module in the contrib. >> > ? ? thanks Anthony >> > >> > ? ? Anthony Minessale a ?crit : >> >> ? ? Meftah, >> >> >> >> ? ? He is 100% correct. ?Please do not insult my volunteer >> >> ? ? developers. Without help from him you would not have any >> >> ? ? FreeSWITCH right now so please drop this subject we are not using >> >> ? ? pjsip. >> >> >> >> >> >> >> >> ? ? On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb >> >> ? ? > wrote: >> >> >> >> ? ? ? ? hi, >> >> ? ? ? ? Pjsip support ICE, STUN and TURN! >> >> ? ? ? ? to STKN: >> >> ? ? ? ? if you don't pjsip, please stop talking or exit the discution >> >> ? ? ? ? we want to kype Freeswitch Clean and universal >> >> >> >> ? ? ? ? Stefan Knoblich a ?crit : >> >>> ? ? ? ? Michael S Collins wrote: >> >>>> ? ? ? ? I can guarantee that the FS devs are well aware of pj-sip. If >> >>>> it was/ is a viable alternative then it would be considered. The fact >> >>>> that it isn't being used is a pretty good indication that it isn't >> >>>> suitable for FS at this time. >> >>>> >> >>>> ? ? ? ? -MV >> >>>> >> >>>> ? ? ? ? Sent from my iPhone >> >>> >> >>> ? ? ? ? We already mentioned some of the reasons why it did get >> >>> ? ? ? ? dropped 3 years ago (first two points from memory, last two >> >>> ? ? ? ? from old IRC logs): [License incompatible (GPL), but i think >> >>> ? ? ? ? tony tried to negotiate on alternate license terms] Not >> >>> ? ? ? ? possible to have multiple SIP profiles (due to global >> >>> ? ? ? ? variables being used in the lib). A race-condition under >> >>> ? ? ? ? high load, that couldn't be resolved back then (with the >> >>> ? ? ? ? help of the pjsip developers). And the sofia module "just >> >>> ? ? ? ? working" and surviving the scalability tests, so all efforts >> >>> ? ? ? ? were focussed on mod_sofia and pjsip got dropped. stkn >> >>> ? ? ? ? _______________________________________________ >> >>> ? ? ? ? FreeSWITCH-users mailing list >> >>> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >> >>> ? ? ? ? >> >>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user >> >>>s http://www.freeswitch.org >> >>> >> >>> >> >>> >> >>> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >> >>> virus signature database 4539 (20091024) __________ >> >>> >> >>> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >> >>> >> >>> ? ? ? ? http://www.eset.com >> >> >> >> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >> >> ? ? ? ? virus signature database 4539 (20091024) __________ >> >> >> >> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >> >> >> >> ? ? ? ? http://www.eset.com >> >> >> >> ? ? ? ? _______________________________________________ >> >> ? ? ? ? FreeSWITCH-users mailing list >> >> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >> >> ? ? ? ? >> >> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> ? ? -- >> >> ? ? Anthony Minessale II >> >> >> >> ? ? FreeSWITCH http://www.freeswitch.org/ >> >> ? ? ClueCon http://www.cluecon.com/ >> >> ? ? Twitter: http://twitter.com/FreeSWITCH_wire >> >> >> >> ? ? AIM: anthm >> >> ? ? MSN:anthony_minessale at hotmail.com >> >> ? ? >> >> ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> ? ? >> >> ? ? IRC: irc.freenode.net #freeswitch >> >> >> >> ? ? FreeSWITCH Developer Conference >> >> ? ? sip:888 at conference.freeswitch.org >> >> ? ? >> >> ? ? iax:guest at conference.freeswitch.org/888 >> >> ? ? >> >> ? ? googletalk:conf+888 at conference.freeswitch.org >> >> ? ? >> >> ? ? pstn:213-799-1400 >> >> >> >> ? ? _______________________________________________ FreeSWITCH-users >> >> ? ? mailing list FreeSWITCH-users at lists.freeswitch.org >> >> ? ? >> >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org __________ Information from ESET NOD32 >> >> Antivirus, version of virus signature database 4539 (20091024) >> >> __________ The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> > >> > ? ? __________ Information from ESET NOD32 Antivirus, version of virus >> > ? ? signature database 4539 (20091024) __________ >> > >> > ? ? The message was checked by ESET NOD32 Antivirus. >> > >> > ? ? http://www.eset.com >> > >> > ? ? _______________________________________________ >> > ? ? FreeSWITCH-users mailing list >> > ? ? FreeSWITCH-users at lists.freeswitch.org >> > ? ? >> > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> > googletalk:conf+888 at conference.freeswitch.org >> > >> > pstn:213-799-1400 >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > __________ Information from ESET NOD32 Antivirus, version of virus >> > signature database 4539 (20091024) __________ >> > >> > The message was checked by ESET NOD32 Antivirus. >> > >> > http://www.eset.com >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From JCasale at activenetwerx.com Sun Nov 1 06:43:58 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Sun, 1 Nov 2009 14:43:58 +0000 Subject: [Freeswitch-users] multihomed help Message-ID: I am setting up fs in pfsense. Following the multihomed tutorial (I also have a dedicated wan/lan int) if I set directory/default.xml domain=10.0.0.1 (my lan int ip) it breaks everything, but if I set in conf/vars.xml I now get audio working correctly, *9999 etc plays MOH. Is there still something I have done wrong? conf/sip_profiles/internal.xml has the int lan ip's entered, and external.xml does not have any edits. Thanks for any info! jlc From jonas.gauffin at gmail.com Sun Nov 1 07:06:04 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Sun, 1 Nov 2009 16:06:04 +0100 Subject: [Freeswitch-users] Small bug in switch_ivr_record_file (in trunk) Message-ID: switch_ivr_play_say.c, line 486. file = switch_core_session_sprintf(session, "%s%s%s%s", switch_str_nil(tfile), tfile ? "]" : "", prefix, SWITCH_PATH_SEPARATOR, file); There should be five %s, not four. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/0b9018b1/attachment.html From jonas.gauffin at gmail.com Sun Nov 1 07:14:57 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Sun, 1 Nov 2009 16:14:57 +0100 Subject: [Freeswitch-users] Small bug in switch_ivr_record_file (in trunk) In-Reply-To: References: Message-ID: Same bug in switch_ivr_async.c, method switch_ivr_record_session. On Sun, Nov 1, 2009 at 4:06 PM, Jonas Gauffin wrote: > switch_ivr_play_say.c, line 486. > > file = switch_core_session_sprintf(session, "%s%s%s%s", > switch_str_nil(tfile), tfile ? "]" : "", prefix, SWITCH_PATH_SEPARATOR, > file); > > There should be five %s, not four. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/b823c318/attachment.html From gromovd at gmail.com Sun Nov 1 07:23:14 2009 From: gromovd at gmail.com (Dmitry Gromov) Date: Sun, 1 Nov 2009 11:23:14 -0400 Subject: [Freeswitch-users] SIP Proxy with direct media path In-Reply-To: <20091101052119.GA28137@jdc.jasonjgw.net> References: <20091101052119.GA28137@jdc.jasonjgw.net> Message-ID: Hi! On Sun, Nov 1, 2009 at 01:21, Jason White wrote: > > http://wiki.freeswitch.org/ is the best there is. > > It is being improved by the community over time. > > You can also take advantage of the IRC channel, the FreeSWITCH conference > and > of course the mailing list. > > > Thank you very much for the reply... I had some experience with FreeSwitch about a year ago, but did not have more time to check all the features. And there wasn't much documentation available then. I'll give it another try this time. Thanks, Dmitry -- DG NJ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/818192da/attachment.html From gmaruzz at celliax.org Sun Nov 1 08:04:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 1 Nov 2009 17:04:36 +0100 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> Message-ID: <7b197bef0911010804n7bcf1d56n180ab963a13a1e39@mail.gmail.com> On Sun, Nov 1, 2009 at 3:37 PM, Diego Viola wrote: > Don't put me on the same leval as DelphiWorld please, I was just > curious about how this SIP stack compares to sofia. Smile Diego, smile. We're all just jocking! :) -gm > > Diego > > On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: >> My favorite part of this 'civilized' discussion on IRC was when DelphiWord and >> diegoviola sat around tryin to take the piss outta stkn on this issue for >> seemingly no reason. Thanks for making the channel a cool place, guys ;) >> >> On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: >>> Anthony Minessale a ?crit : >>> > Meftah, >>> > Feel free. >>> > >>> > thanks >>> > >>> > P.S. >>> > >>> > STKN was the guy who made the first mod_pjsip for FS that we abandoned >>> > years ago. So you should believe him. >>> > Both him and I agreed it was not working out. ?So if you don't believe >>> > me, find out for yourself. >>> > anthony, why i don't believe ?you? >>> >>> never say that. >>> i believe you and all Freeswitch Staf and thank you and to all >>> Freeswitch Staf. >>> >>> > On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb >> > > wrote: >>> > >>> > ? ? hi Anthony >>> > ? ? i agry >>> > ? ? i say that because STKN hate all my suggestions. >>> > ? ? about pjsip, i will contribute aditional module in the contrib. >>> > ? ? thanks Anthony >>> > >>> > ? ? Anthony Minessale a ?crit : >>> >> ? ? Meftah, >>> >> >>> >> ? ? He is 100% correct. ?Please do not insult my volunteer >>> >> ? ? developers. Without help from him you would not have any >>> >> ? ? FreeSWITCH right now so please drop this subject we are not using >>> >> ? ? pjsip. >>> >> >>> >> >>> >> >>> >> ? ? On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb >>> >> ? ? > wrote: >>> >> >>> >> ? ? ? ? hi, >>> >> ? ? ? ? Pjsip support ICE, STUN and TURN! >>> >> ? ? ? ? to STKN: >>> >> ? ? ? ? if you don't pjsip, please stop talking or exit the discution >>> >> ? ? ? ? we want to kype Freeswitch Clean and universal >>> >> >>> >> ? ? ? ? Stefan Knoblich a ?crit : >>> >>> ? ? ? ? Michael S Collins wrote: >>> >>>> ? ? ? ? I can guarantee that the FS devs are well aware of pj-sip. If >>> >>>> it was/ is a viable alternative then it would be considered. The fact >>> >>>> that it isn't being used is a pretty good indication that it isn't >>> >>>> suitable for FS at this time. >>> >>>> >>> >>>> ? ? ? ? -MV >>> >>>> >>> >>>> ? ? ? ? Sent from my iPhone >>> >>> >>> >>> ? ? ? ? We already mentioned some of the reasons why it did get >>> >>> ? ? ? ? dropped 3 years ago (first two points from memory, last two >>> >>> ? ? ? ? from old IRC logs): [License incompatible (GPL), but i think >>> >>> ? ? ? ? tony tried to negotiate on alternate license terms] Not >>> >>> ? ? ? ? possible to have multiple SIP profiles (due to global >>> >>> ? ? ? ? variables being used in the lib). A race-condition under >>> >>> ? ? ? ? high load, that couldn't be resolved back then (with the >>> >>> ? ? ? ? help of the pjsip developers). And the sofia module "just >>> >>> ? ? ? ? working" and surviving the scalability tests, so all efforts >>> >>> ? ? ? ? were focussed on mod_sofia and pjsip got dropped. stkn >>> >>> ? ? ? ? _______________________________________________ >>> >>> ? ? ? ? FreeSWITCH-users mailing list >>> >>> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >>> >>> ? ? ? ? >>> >>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user >>> >>>s http://www.freeswitch.org >>> >>> >>> >>> >>> >>> >>> >>> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >>> >>> virus signature database 4539 (20091024) __________ >>> >>> >>> >>> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >>> >>> >>> >>> ? ? ? ? http://www.eset.com >>> >> >>> >> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >>> >> ? ? ? ? virus signature database 4539 (20091024) __________ >>> >> >>> >> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >>> >> >>> >> ? ? ? ? http://www.eset.com >>> >> >>> >> ? ? ? ? _______________________________________________ >>> >> ? ? ? ? FreeSWITCH-users mailing list >>> >> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >>> >> ? ? ? ? >>> >> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> >> >>> >> ? ? -- >>> >> ? ? Anthony Minessale II >>> >> >>> >> ? ? FreeSWITCH http://www.freeswitch.org/ >>> >> ? ? ClueCon http://www.cluecon.com/ >>> >> ? ? Twitter: http://twitter.com/FreeSWITCH_wire >>> >> >>> >> ? ? AIM: anthm >>> >> ? ? MSN:anthony_minessale at hotmail.com >>> >> ? ? >>> >> ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >> ? ? >>> >> ? ? IRC: irc.freenode.net #freeswitch >>> >> >>> >> ? ? FreeSWITCH Developer Conference >>> >> ? ? sip:888 at conference.freeswitch.org >>> >> ? ? >>> >> ? ? iax:guest at conference.freeswitch.org/888 >>> >> ? ? >>> >> ? ? googletalk:conf+888 at conference.freeswitch.org >>> >> ? ? >>> >> ? ? pstn:213-799-1400 >>> >> >>> >> ? ? _______________________________________________ FreeSWITCH-users >>> >> ? ? mailing list FreeSWITCH-users at lists.freeswitch.org >>> >> ? ? >>> >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org __________ Information from ESET NOD32 >>> >> Antivirus, version of virus signature database 4539 (20091024) >>> >> __________ The message was checked by ESET NOD32 Antivirus. >>> >> http://www.eset.com >>> > >>> > ? ? __________ Information from ESET NOD32 Antivirus, version of virus >>> > ? ? signature database 4539 (20091024) __________ >>> > >>> > ? ? The message was checked by ESET NOD32 Antivirus. >>> > >>> > ? ? http://www.eset.com >>> > >>> > ? ? _______________________________________________ >>> > ? ? FreeSWITCH-users mailing list >>> > ? ? FreeSWITCH-users at lists.freeswitch.org >>> > ? ? >>> > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > -- >>> > Anthony Minessale II >>> > >>> > FreeSWITCH http://www.freeswitch.org/ >>> > ClueCon http://www.cluecon.com/ >>> > Twitter: http://twitter.com/FreeSWITCH_wire >>> > >>> > AIM: anthm >>> > MSN:anthony_minessale at hotmail.com >>> > >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> > >>> > IRC: irc.freenode.net #freeswitch >>> > >>> > FreeSWITCH Developer Conference >>> > sip:888 at conference.freeswitch.org >>> > >>> > iax:guest at conference.freeswitch.org/888 >>> > >>> > googletalk:conf+888 at conference.freeswitch.org >>> > >>> > pstn:213-799-1400 >>> > ------------------------------------------------------------------------ >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> > >>> > __________ Information from ESET NOD32 Antivirus, version of virus >>> > signature database 4539 (20091024) __________ >>> > >>> > The message was checked by ESET NOD32 Antivirus. >>> > >>> > http://www.eset.com >>> >>> __________ Information from ESET NOD32 Antivirus, version of virus >>> signature database 4539 (20091024) __________ >>> >>> The message was checked by ESET NOD32 Antivirus. >>> >>> http://www.eset.com >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From tayeb.meftah at gmail.com Sun Nov 1 09:46:19 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sun, 01 Nov 2009 17:46:19 +0000 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> Message-ID: <4AEDC96B.5090009@gmail.com> hi diego, what you mean? so my level is nothing? my level is bad? my level is zero? thank to gmaruzz/MikeJ that understand me quickly/easyly Diego Viola a ?crit : > Don't put me on the same leval as DelphiWorld please, I was just > curious about how this SIP stack compares to sofia. > > Diego > > On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: > >> My favorite part of this 'civilized' discussion on IRC was when DelphiWord and >> diegoviola sat around tryin to take the piss outta stkn on this issue for >> seemingly no reason. Thanks for making the channel a cool place, guys ;) >> >> On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: >> >>> Anthony Minessale a ?crit : >>> >>>> Meftah, >>>> Feel free. >>>> >>>> thanks >>>> >>>> P.S. >>>> >>>> STKN was the guy who made the first mod_pjsip for FS that we abandoned >>>> years ago. So you should believe him. >>>> Both him and I agreed it was not working out. So if you don't believe >>>> me, find out for yourself. >>>> anthony, why i don't believe you? >>>> >>> never say that. >>> i believe you and all Freeswitch Staf and thank you and to all >>> Freeswitch Staf. >>> >>> >>>> On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb >>> > wrote: >>>> >>>> hi Anthony >>>> i agry >>>> i say that because STKN hate all my suggestions. >>>> about pjsip, i will contribute aditional module in the contrib. >>>> thanks Anthony >>>> >>>> Anthony Minessale a ?crit : >>>> >>>>> Meftah, >>>>> >>>>> He is 100% correct. Please do not insult my volunteer >>>>> developers. Without help from him you would not have any >>>>> FreeSWITCH right now so please drop this subject we are not using >>>>> pjsip. >>>>> >>>>> >>>>> >>>>> On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb >>>>> > wrote: >>>>> >>>>> hi, >>>>> Pjsip support ICE, STUN and TURN! >>>>> to STKN: >>>>> if you don't pjsip, please stop talking or exit the discution >>>>> we want to kype Freeswitch Clean and universal >>>>> >>>>> Stefan Knoblich a ?crit : >>>>> >>>>>> Michael S Collins wrote: >>>>>> >>>>>>> I can guarantee that the FS devs are well aware of pj-sip. If >>>>>>> it was/ is a viable alternative then it would be considered. The fact >>>>>>> that it isn't being used is a pretty good indication that it isn't >>>>>>> suitable for FS at this time. >>>>>>> >>>>>>> -MV >>>>>>> >>>>>>> Sent from my iPhone >>>>>>> >>>>>> We already mentioned some of the reasons why it did get >>>>>> dropped 3 years ago (first two points from memory, last two >>>>>> from old IRC logs): [License incompatible (GPL), but i think >>>>>> tony tried to negotiate on alternate license terms] Not >>>>>> possible to have multiple SIP profiles (due to global >>>>>> variables being used in the lib). A race-condition under >>>>>> high load, that couldn't be resolved back then (with the >>>>>> help of the pjsip developers). And the sofia module "just >>>>>> working" and surviving the scalability tests, so all efforts >>>>>> were focussed on mod_sofia and pjsip got dropped. stkn >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user >>>>>> s http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> __________ Information from ESET NOD32 Antivirus, version of >>>>>> virus signature database 4539 (20091024) __________ >>>>>> >>>>>> The message was checked by ESET NOD32 Antivirus. >>>>>> >>>>>> http://www.eset.com >>>>>> >>>>> __________ Information from ESET NOD32 Antivirus, version of >>>>> virus signature database 4539 (20091024) __________ >>>>> >>>>> The message was checked by ESET NOD32 Antivirus. >>>>> >>>>> http://www.eset.com >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> >>>>> pstn:213-799-1400 >>>>> >>>>> _______________________________________________ FreeSWITCH-users >>>>> mailing list FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org __________ Information from ESET NOD32 >>>>> Antivirus, version of virus signature database 4539 (20091024) >>>>> __________ The message was checked by ESET NOD32 Antivirus. >>>>> http://www.eset.com >>>>> >>>> __________ Information from ESET NOD32 Antivirus, version of virus >>>> signature database 4539 (20091024) __________ >>>> >>>> The message was checked by ESET NOD32 Antivirus. >>>> >>>> http://www.eset.com >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> >>>> iax:guest at conference.freeswitch.org/888 >>>> >>>> googletalk:conf+888 at conference.freeswitch.org >>>> >>>> pstn:213-799-1400 >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> __________ Information from ESET NOD32 Antivirus, version of virus >>>> signature database 4539 (20091024) __________ >>>> >>>> The message was checked by ESET NOD32 Antivirus. >>>> >>>> http://www.eset.com >>>> >>> __________ Information from ESET NOD32 Antivirus, version of virus >>> signature database 4539 (20091024) __________ >>> >>> The message was checked by ESET NOD32 Antivirus. >>> >>> http://www.eset.com >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/bc8f9c19/attachment-0001.html From diego.viola at gmail.com Sun Nov 1 08:56:23 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 1 Nov 2009 16:56:23 +0000 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <7b197bef0911010804n7bcf1d56n180ab963a13a1e39@mail.gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> <7b197bef0911010804n7bcf1d56n180ab963a13a1e39@mail.gmail.com> Message-ID: <86a32abc0911010856s173a45a4m9e1d2b0fee9f165a@mail.gmail.com> =D On Sun, Nov 1, 2009 at 4:04 PM, Giovanni Maruzzelli wrote: > On Sun, Nov 1, 2009 at 3:37 PM, Diego Viola wrote: >> Don't put me on the same leval as DelphiWorld please, I was just >> curious about how this SIP stack compares to sofia. > > Smile Diego, smile. We're all just jocking! ?:) > > -gm > > >> >> Diego >> >> On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: >>> My favorite part of this 'civilized' discussion on IRC was when DelphiWord and >>> diegoviola sat around tryin to take the piss outta stkn on this issue for >>> seemingly no reason. Thanks for making the channel a cool place, guys ;) >>> >>> On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: >>>> Anthony Minessale a ?crit : >>>> > Meftah, >>>> > Feel free. >>>> > >>>> > thanks >>>> > >>>> > P.S. >>>> > >>>> > STKN was the guy who made the first mod_pjsip for FS that we abandoned >>>> > years ago. So you should believe him. >>>> > Both him and I agreed it was not working out. ?So if you don't believe >>>> > me, find out for yourself. >>>> > anthony, why i don't believe ?you? >>>> >>>> never say that. >>>> i believe you and all Freeswitch Staf and thank you and to all >>>> Freeswitch Staf. >>>> >>>> > On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb >>> > > wrote: >>>> > >>>> > ? ? hi Anthony >>>> > ? ? i agry >>>> > ? ? i say that because STKN hate all my suggestions. >>>> > ? ? about pjsip, i will contribute aditional module in the contrib. >>>> > ? ? thanks Anthony >>>> > >>>> > ? ? Anthony Minessale a ?crit : >>>> >> ? ? Meftah, >>>> >> >>>> >> ? ? He is 100% correct. ?Please do not insult my volunteer >>>> >> ? ? developers. Without help from him you would not have any >>>> >> ? ? FreeSWITCH right now so please drop this subject we are not using >>>> >> ? ? pjsip. >>>> >> >>>> >> >>>> >> >>>> >> ? ? On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb >>>> >> ? ? > wrote: >>>> >> >>>> >> ? ? ? ? hi, >>>> >> ? ? ? ? Pjsip support ICE, STUN and TURN! >>>> >> ? ? ? ? to STKN: >>>> >> ? ? ? ? if you don't pjsip, please stop talking or exit the discution >>>> >> ? ? ? ? we want to kype Freeswitch Clean and universal >>>> >> >>>> >> ? ? ? ? Stefan Knoblich a ?crit : >>>> >>> ? ? ? ? Michael S Collins wrote: >>>> >>>> ? ? ? ? I can guarantee that the FS devs are well aware of pj-sip. If >>>> >>>> it was/ is a viable alternative then it would be considered. The fact >>>> >>>> that it isn't being used is a pretty good indication that it isn't >>>> >>>> suitable for FS at this time. >>>> >>>> >>>> >>>> ? ? ? ? -MV >>>> >>>> >>>> >>>> ? ? ? ? Sent from my iPhone >>>> >>> >>>> >>> ? ? ? ? We already mentioned some of the reasons why it did get >>>> >>> ? ? ? ? dropped 3 years ago (first two points from memory, last two >>>> >>> ? ? ? ? from old IRC logs): [License incompatible (GPL), but i think >>>> >>> ? ? ? ? tony tried to negotiate on alternate license terms] Not >>>> >>> ? ? ? ? possible to have multiple SIP profiles (due to global >>>> >>> ? ? ? ? variables being used in the lib). A race-condition under >>>> >>> ? ? ? ? high load, that couldn't be resolved back then (with the >>>> >>> ? ? ? ? help of the pjsip developers). And the sofia module "just >>>> >>> ? ? ? ? working" and surviving the scalability tests, so all efforts >>>> >>> ? ? ? ? were focussed on mod_sofia and pjsip got dropped. stkn >>>> >>> ? ? ? ? _______________________________________________ >>>> >>> ? ? ? ? FreeSWITCH-users mailing list >>>> >>> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >>>> >>> ? ? ? ? >>>> >>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user >>>> >>>s http://www.freeswitch.org >>>> >>> >>>> >>> >>>> >>> >>>> >>> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >>>> >>> virus signature database 4539 (20091024) __________ >>>> >>> >>>> >>> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >>>> >>> >>>> >>> ? ? ? ? http://www.eset.com >>>> >> >>>> >> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >>>> >> ? ? ? ? virus signature database 4539 (20091024) __________ >>>> >> >>>> >> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >>>> >> >>>> >> ? ? ? ? http://www.eset.com >>>> >> >>>> >> ? ? ? ? _______________________________________________ >>>> >> ? ? ? ? FreeSWITCH-users mailing list >>>> >> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >>>> >> ? ? ? ? >>>> >> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> >>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> ? ? -- >>>> >> ? ? Anthony Minessale II >>>> >> >>>> >> ? ? FreeSWITCH http://www.freeswitch.org/ >>>> >> ? ? ClueCon http://www.cluecon.com/ >>>> >> ? ? Twitter: http://twitter.com/FreeSWITCH_wire >>>> >> >>>> >> ? ? AIM: anthm >>>> >> ? ? MSN:anthony_minessale at hotmail.com >>>> >> ? ? >>>> >> ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> >> ? ? >>>> >> ? ? IRC: irc.freenode.net #freeswitch >>>> >> >>>> >> ? ? FreeSWITCH Developer Conference >>>> >> ? ? sip:888 at conference.freeswitch.org >>>> >> ? ? >>>> >> ? ? iax:guest at conference.freeswitch.org/888 >>>> >> ? ? >>>> >> ? ? googletalk:conf+888 at conference.freeswitch.org >>>> >> ? ? >>>> >> ? ? pstn:213-799-1400 >>>> >> >>>> >> ? ? _______________________________________________ FreeSWITCH-users >>>> >> ? ? mailing list FreeSWITCH-users at lists.freeswitch.org >>>> >> ? ? >>>> >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> >>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org __________ Information from ESET NOD32 >>>> >> Antivirus, version of virus signature database 4539 (20091024) >>>> >> __________ The message was checked by ESET NOD32 Antivirus. >>>> >> http://www.eset.com >>>> > >>>> > ? ? __________ Information from ESET NOD32 Antivirus, version of virus >>>> > ? ? signature database 4539 (20091024) __________ >>>> > >>>> > ? ? The message was checked by ESET NOD32 Antivirus. >>>> > >>>> > ? ? http://www.eset.com >>>> > >>>> > ? ? _______________________________________________ >>>> > ? ? FreeSWITCH-users mailing list >>>> > ? ? FreeSWITCH-users at lists.freeswitch.org >>>> > ? ? >>>> > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> > >>>> > >>>> > -- >>>> > Anthony Minessale II >>>> > >>>> > FreeSWITCH http://www.freeswitch.org/ >>>> > ClueCon http://www.cluecon.com/ >>>> > Twitter: http://twitter.com/FreeSWITCH_wire >>>> > >>>> > AIM: anthm >>>> > MSN:anthony_minessale at hotmail.com >>>> > >>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> > >>>> > IRC: irc.freenode.net #freeswitch >>>> > >>>> > FreeSWITCH Developer Conference >>>> > sip:888 at conference.freeswitch.org >>>> > >>>> > iax:guest at conference.freeswitch.org/888 >>>> > >>>> > googletalk:conf+888 at conference.freeswitch.org >>>> > >>>> > pstn:213-799-1400 >>>> > ------------------------------------------------------------------------ >>>> > >>>> > _______________________________________________ >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> > >>>> > __________ Information from ESET NOD32 Antivirus, version of virus >>>> > signature database 4539 (20091024) __________ >>>> > >>>> > The message was checked by ESET NOD32 Antivirus. >>>> > >>>> > http://www.eset.com >>>> >>>> __________ Information from ESET NOD32 Antivirus, version of virus >>>> signature database 4539 (20091024) __________ >>>> >>>> The message was checked by ESET NOD32 Antivirus. >>>> >>>> http://www.eset.com >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From diego.viola at gmail.com Sun Nov 1 09:09:16 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 1 Nov 2009 17:09:16 +0000 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <4AEDC96B.5090009@gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> <4AEDC96B.5090009@gmail.com> Message-ID: <86a32abc0911010909lcddf914u1b477b22ee65ed75@mail.gmail.com> Hi Meftah, No, of course is not, and it will never be, I actually quite admire how you are able to do what you do. I just wanted to say that I don't want people to put me into this problem, because I don't have anything to do with it, I was just curious about how pjsip compares to sofia, etc. That's all. Apologies if I didn't expressed myself correctly. Regards, Diego On Sun, Nov 1, 2009 at 5:46 PM, Meftah Tayeb wrote: > hi diego, > what you mean? > so my level is nothing? > my level is bad? > my level is zero? > thank to gmaruzz/MikeJ that understand me quickly/easyly > Diego Viola a ?crit?: > > Don't put me on the same leval as DelphiWorld please, I was just > curious about how this SIP stack compares to sofia. > > Diego > > On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: > > > My favorite part of this 'civilized' discussion on IRC was when DelphiWord > and > diegoviola sat around tryin to take the piss outta stkn on this issue for > seemingly no reason. Thanks for making the channel a cool place, guys ;) > > On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: > > > Anthony Minessale a ?crit : > > > Meftah, > Feel free. > > thanks > > P.S. > > STKN was the guy who made the first mod_pjsip for FS that we abandoned > years ago. So you should believe him. > Both him and I agreed it was not working out. ?So if you don't believe > me, find out for yourself. > anthony, why i don't believe ?you? > > > never say that. > i believe you and all Freeswitch Staf and thank you and to all > Freeswitch Staf. > > > > On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb > wrote: > > ? ? hi Anthony > ? ? i agry > ? ? i say that because STKN hate all my suggestions. > ? ? about pjsip, i will contribute aditional module in the contrib. > ? ? thanks Anthony > > ? ? Anthony Minessale a ?crit : > > > ? ? Meftah, > > ? ? He is 100% correct. ?Please do not insult my volunteer > ? ? developers. Without help from him you would not have any > ? ? FreeSWITCH right now so please drop this subject we are not using > ? ? pjsip. > > > > ? ? On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb > ? ? > wrote: > > ? ? ? ? hi, > ? ? ? ? Pjsip support ICE, STUN and TURN! > ? ? ? ? to STKN: > ? ? ? ? if you don't pjsip, please stop talking or exit the discution > ? ? ? ? we want to kype Freeswitch Clean and universal > > ? ? ? ? Stefan Knoblich a ?crit : > > > ? ? ? ? Michael S Collins wrote: > > > ? ? ? ? I can guarantee that the FS devs are well aware of pj-sip. If > it was/ is a viable alternative then it would be considered. The fact > that it isn't being used is a pretty good indication that it isn't > suitable for FS at this time. > > ? ? ? ? -MV > > ? ? ? ? Sent from my iPhone > > > ? ? ? ? We already mentioned some of the reasons why it did get > ? ? ? ? dropped 3 years ago (first two points from memory, last two > ? ? ? ? from old IRC logs): [License incompatible (GPL), but i think > ? ? ? ? tony tried to negotiate on alternate license terms] Not > ? ? ? ? possible to have multiple SIP profiles (due to global > ? ? ? ? variables being used in the lib). A race-condition under > ? ? ? ? high load, that couldn't be resolved back then (with the > ? ? ? ? help of the pjsip developers). And the sofia module "just > ? ? ? ? working" and surviving the scalability tests, so all efforts > ? ? ? ? were focussed on mod_sofia and pjsip got dropped. stkn > ? ? ? ? _______________________________________________ > ? ? ? ? FreeSWITCH-users mailing list > ? ? ? ? FreeSWITCH-users at lists.freeswitch.org > ? ? ? ? > ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > s http://www.freeswitch.org > > > > ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of > virus signature database 4539 (20091024) __________ > > ? ? ? ? The message was checked by ESET NOD32 Antivirus. > > ? ? ? ? http://www.eset.com > > > ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of > ? ? ? ? virus signature database 4539 (20091024) __________ > > ? ? ? ? The message was checked by ESET NOD32 Antivirus. > > ? ? ? ? http://www.eset.com > > ? ? ? ? _______________________________________________ > ? ? ? ? FreeSWITCH-users mailing list > ? ? ? ? FreeSWITCH-users at lists.freeswitch.org > ? ? ? ? > ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ? ? -- > ? ? Anthony Minessale II > > ? ? FreeSWITCH http://www.freeswitch.org/ > ? ? ClueCon http://www.cluecon.com/ > ? ? Twitter: http://twitter.com/FreeSWITCH_wire > > ? ? AIM: anthm > ? ? MSN:anthony_minessale at hotmail.com > ? ? > ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > ? ? > ? ? IRC: irc.freenode.net #freeswitch > > ? ? FreeSWITCH Developer Conference > ? ? sip:888 at conference.freeswitch.org > ? ? > ? ? iax:guest at conference.freeswitch.org/888 > ? ? > ? ? googletalk:conf+888 at conference.freeswitch.org > ? ? > ? ? pstn:213-799-1400 > > ? ? _______________________________________________ FreeSWITCH-users > ? ? mailing list FreeSWITCH-users at lists.freeswitch.org > ? ? > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org __________ Information from ESET NOD32 > Antivirus, version of virus signature database 4539 (20091024) > __________ The message was checked by ESET NOD32 Antivirus. > http://www.eset.com > > > ? ? __________ Information from ESET NOD32 Antivirus, version of virus > ? ? signature database 4539 (20091024) __________ > > ? ? The message was checked by ESET NOD32 Antivirus. > > ? ? http://www.eset.com > > ? ? _______________________________________________ > ? ? FreeSWITCH-users mailing list > ? ? FreeSWITCH-users at lists.freeswitch.org > ? ? > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > __________ Information from ESET NOD32 Antivirus, version of virus signature > database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature > database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Sun Nov 1 09:25:07 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 1 Nov 2009 17:25:07 +0000 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <86a32abc0911010909lcddf914u1b477b22ee65ed75@mail.gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> <4AEDC96B.5090009@gmail.com> <86a32abc0911010909lcddf914u1b477b22ee65ed75@mail.gmail.com> Message-ID: <86a32abc0911010925s19986a84ga20466aef0379cd4@mail.gmail.com> I never said your level is bad or anything, I just said that I don't want people to involve me into that problem. Diego On Sun, Nov 1, 2009 at 5:09 PM, Diego Viola wrote: > Hi Meftah, > > No, of course is not, and it will never be, I actually quite admire > how you are able to do what you do. > > I just wanted to say that I don't want people to put me into this > problem, because I don't have anything to do with it, I was just > curious about how pjsip compares to sofia, etc. That's all. > > Apologies if I didn't expressed myself correctly. > > Regards, > > Diego > > On Sun, Nov 1, 2009 at 5:46 PM, Meftah Tayeb wrote: >> hi diego, >> what you mean? >> so my level is nothing? >> my level is bad? >> my level is zero? >> thank to gmaruzz/MikeJ that understand me quickly/easyly >> Diego Viola a ?crit?: >> >> Don't put me on the same leval as DelphiWorld please, I was just >> curious about how this SIP stack compares to sofia. >> >> Diego >> >> On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: >> >> >> My favorite part of this 'civilized' discussion on IRC was when DelphiWord >> and >> diegoviola sat around tryin to take the piss outta stkn on this issue for >> seemingly no reason. Thanks for making the channel a cool place, guys ;) >> >> On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: >> >> >> Anthony Minessale a ?crit : >> >> >> Meftah, >> Feel free. >> >> thanks >> >> P.S. >> >> STKN was the guy who made the first mod_pjsip for FS that we abandoned >> years ago. So you should believe him. >> Both him and I agreed it was not working out. ?So if you don't believe >> me, find out for yourself. >> anthony, why i don't believe ?you? >> >> >> never say that. >> i believe you and all Freeswitch Staf and thank you and to all >> Freeswitch Staf. >> >> >> >> On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb > > wrote: >> >> ? ? hi Anthony >> ? ? i agry >> ? ? i say that because STKN hate all my suggestions. >> ? ? about pjsip, i will contribute aditional module in the contrib. >> ? ? thanks Anthony >> >> ? ? Anthony Minessale a ?crit : >> >> >> ? ? Meftah, >> >> ? ? He is 100% correct. ?Please do not insult my volunteer >> ? ? developers. Without help from him you would not have any >> ? ? FreeSWITCH right now so please drop this subject we are not using >> ? ? pjsip. >> >> >> >> ? ? On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb >> ? ? > wrote: >> >> ? ? ? ? hi, >> ? ? ? ? Pjsip support ICE, STUN and TURN! >> ? ? ? ? to STKN: >> ? ? ? ? if you don't pjsip, please stop talking or exit the discution >> ? ? ? ? we want to kype Freeswitch Clean and universal >> >> ? ? ? ? Stefan Knoblich a ?crit : >> >> >> ? ? ? ? Michael S Collins wrote: >> >> >> ? ? ? ? I can guarantee that the FS devs are well aware of pj-sip. If >> it was/ is a viable alternative then it would be considered. The fact >> that it isn't being used is a pretty good indication that it isn't >> suitable for FS at this time. >> >> ? ? ? ? -MV >> >> ? ? ? ? Sent from my iPhone >> >> >> ? ? ? ? We already mentioned some of the reasons why it did get >> ? ? ? ? dropped 3 years ago (first two points from memory, last two >> ? ? ? ? from old IRC logs): [License incompatible (GPL), but i think >> ? ? ? ? tony tried to negotiate on alternate license terms] Not >> ? ? ? ? possible to have multiple SIP profiles (due to global >> ? ? ? ? variables being used in the lib). A race-condition under >> ? ? ? ? high load, that couldn't be resolved back then (with the >> ? ? ? ? help of the pjsip developers). And the sofia module "just >> ? ? ? ? working" and surviving the scalability tests, so all efforts >> ? ? ? ? were focussed on mod_sofia and pjsip got dropped. stkn >> ? ? ? ? _______________________________________________ >> ? ? ? ? FreeSWITCH-users mailing list >> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >> ? ? ? ? >> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user >> s http://www.freeswitch.org >> >> >> >> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >> virus signature database 4539 (20091024) __________ >> >> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >> >> ? ? ? ? http://www.eset.com >> >> >> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >> ? ? ? ? virus signature database 4539 (20091024) __________ >> >> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >> >> ? ? ? ? http://www.eset.com >> >> ? ? ? ? _______________________________________________ >> ? ? ? ? FreeSWITCH-users mailing list >> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >> ? ? ? ? >> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> ? ? -- >> ? ? Anthony Minessale II >> >> ? ? FreeSWITCH http://www.freeswitch.org/ >> ? ? ClueCon http://www.cluecon.com/ >> ? ? Twitter: http://twitter.com/FreeSWITCH_wire >> >> ? ? AIM: anthm >> ? ? MSN:anthony_minessale at hotmail.com >> ? ? >> ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> ? ? >> ? ? IRC: irc.freenode.net #freeswitch >> >> ? ? FreeSWITCH Developer Conference >> ? ? sip:888 at conference.freeswitch.org >> ? ? >> ? ? iax:guest at conference.freeswitch.org/888 >> ? ? >> ? ? googletalk:conf+888 at conference.freeswitch.org >> ? ? >> ? ? pstn:213-799-1400 >> >> ? ? _______________________________________________ FreeSWITCH-users >> ? ? mailing list FreeSWITCH-users at lists.freeswitch.org >> ? ? >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org __________ Information from ESET NOD32 >> Antivirus, version of virus signature database 4539 (20091024) >> __________ The message was checked by ESET NOD32 Antivirus. >> http://www.eset.com >> >> >> ? ? __________ Information from ESET NOD32 Antivirus, version of virus >> ? ? signature database 4539 (20091024) __________ >> >> ? ? The message was checked by ESET NOD32 Antivirus. >> >> ? ? http://www.eset.com >> >> ? ? _______________________________________________ >> ? ? FreeSWITCH-users mailing list >> ? ? FreeSWITCH-users at lists.freeswitch.org >> ? ? >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus signature >> database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> >> >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus signature >> database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From anthony.minessale at gmail.com Sun Nov 1 10:51:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 12:51:57 -0600 Subject: [Freeswitch-users] Small bug in switch_ivr_record_file (in trunk) In-Reply-To: References: Message-ID: <191c3a030911011051w63f1e9e0ibb03be66ac8d53eb@mail.gmail.com> thank you fixed in r15308 We do prefer jira for this kind of thing so we can track and make accurate change logs. On Sun, Nov 1, 2009 at 9:14 AM, Jonas Gauffin wrote: > Same bug in switch_ivr_async.c, method switch_ivr_record_session. > > > On Sun, Nov 1, 2009 at 4:06 PM, Jonas Gauffin wrote: > >> switch_ivr_play_say.c, line 486. >> >> file = switch_core_session_sprintf(session, "%s%s%s%s", >> switch_str_nil(tfile), tfile ? "]" : "", prefix, SWITCH_PATH_SEPARATOR, >> file); >> >> There should be five %s, not four. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/213a1ae9/attachment.html From anthony.minessale at gmail.com Sun Nov 1 10:54:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 12:54:34 -0600 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <86a32abc0911010925s19986a84ga20466aef0379cd4@mail.gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> <4AEDC96B.5090009@gmail.com> <86a32abc0911010909lcddf914u1b477b22ee65ed75@mail.gmail.com> <86a32abc0911010925s19986a84ga20466aef0379cd4@mail.gmail.com> Message-ID: <191c3a030911011054j131f6710s742c8033c3ac71eb@mail.gmail.com> Meftah and Diego, If you continue to argue on our mailing lists I will be forced to moderate you to stop you from bothering other people. I do not want to see one more reply to this thread from either of you. Please do not reply to apologize, simply stop sending any more email to this topic. On Sun, Nov 1, 2009 at 11:25 AM, Diego Viola wrote: > I never said your level is bad or anything, I just said that I don't > want people to involve me into that problem. > > Diego > > On Sun, Nov 1, 2009 at 5:09 PM, Diego Viola wrote: > > Hi Meftah, > > > > No, of course is not, and it will never be, I actually quite admire > > how you are able to do what you do. > > > > I just wanted to say that I don't want people to put me into this > > problem, because I don't have anything to do with it, I was just > > curious about how pjsip compares to sofia, etc. That's all. > > > > Apologies if I didn't expressed myself correctly. > > > > Regards, > > > > Diego > > > > On Sun, Nov 1, 2009 at 5:46 PM, Meftah Tayeb > wrote: > >> hi diego, > >> what you mean? > >> so my level is nothing? > >> my level is bad? > >> my level is zero? > >> thank to gmaruzz/MikeJ that understand me quickly/easyly > >> Diego Viola a ?crit : > >> > >> Don't put me on the same leval as DelphiWorld please, I was just > >> curious about how this SIP stack compares to sofia. > >> > >> Diego > >> > >> On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: > >> > >> > >> My favorite part of this 'civilized' discussion on IRC was when > DelphiWord > >> and > >> diegoviola sat around tryin to take the piss outta stkn on this issue > for > >> seemingly no reason. Thanks for making the channel a cool place, guys ;) > >> > >> On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: > >> > >> > >> Anthony Minessale a ?crit : > >> > >> > >> Meftah, > >> Feel free. > >> > >> thanks > >> > >> P.S. > >> > >> STKN was the guy who made the first mod_pjsip for FS that we abandoned > >> years ago. So you should believe him. > >> Both him and I agreed it was not working out. So if you don't believe > >> me, find out for yourself. > >> anthony, why i don't believe you? > >> > >> > >> never say that. > >> i believe you and all Freeswitch Staf and thank you and to all > >> Freeswitch Staf. > >> > >> > >> > >> On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb >> > wrote: > >> > >> hi Anthony > >> i agry > >> i say that because STKN hate all my suggestions. > >> about pjsip, i will contribute aditional module in the contrib. > >> thanks Anthony > >> > >> Anthony Minessale a ?crit : > >> > >> > >> Meftah, > >> > >> He is 100% correct. Please do not insult my volunteer > >> developers. Without help from him you would not have any > >> FreeSWITCH right now so please drop this subject we are not using > >> pjsip. > >> > >> > >> > >> On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb > >> > wrote: > >> > >> hi, > >> Pjsip support ICE, STUN and TURN! > >> to STKN: > >> if you don't pjsip, please stop talking or exit the discution > >> we want to kype Freeswitch Clean and universal > >> > >> Stefan Knoblich a ?crit : > >> > >> > >> Michael S Collins wrote: > >> > >> > >> I can guarantee that the FS devs are well aware of pj-sip. If > >> it was/ is a viable alternative then it would be considered. The fact > >> that it isn't being used is a pretty good indication that it isn't > >> suitable for FS at this time. > >> > >> -MV > >> > >> Sent from my iPhone > >> > >> > >> We already mentioned some of the reasons why it did get > >> dropped 3 years ago (first two points from memory, last two > >> from old IRC logs): [License incompatible (GPL), but i think > >> tony tried to negotiate on alternate license terms] Not > >> possible to have multiple SIP profiles (due to global > >> variables being used in the lib). A race-condition under > >> high load, that couldn't be resolved back then (with the > >> help of the pjsip developers). And the sofia module "just > >> working" and surviving the scalability tests, so all efforts > >> were focussed on mod_sofia and pjsip got dropped. stkn > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > >> s http://www.freeswitch.org > >> > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of > >> virus signature database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of > >> virus signature database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> > > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> > > > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > >> > > > >> iax:guest at conference.freeswitch.org/888 > >> > >> googletalk:conf+888 at conference.freeswitch.org > >> > > > >> pstn:213-799-1400 > >> > >> _______________________________________________ FreeSWITCH-users > >> mailing list FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org __________ Information from ESET NOD32 > >> Antivirus, version of virus signature database 4539 (20091024) > >> __________ The message was checked by ESET NOD32 Antivirus. > >> http://www.eset.com > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of virus > >> signature database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> > > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> > > > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > >> > > > >> iax:guest at conference.freeswitch.org/888 > >> > >> googletalk:conf+888 at conference.freeswitch.org > >> > > > >> pstn:213-799-1400 > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of virus > >> signature database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of virus > >> signature database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of virus > signature > >> database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> > >> > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of virus > signature > >> database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/88ae4771/attachment-0001.html From anthony.minessale at gmail.com Sun Nov 1 11:27:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 13:27:31 -0600 Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session In-Reply-To: <8ccbff060910311945j5be2fa85j95e7e1b422dcc3a9@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com> <191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com> <8ccbff060910311945j5be2fa85j95e7e1b422dcc3a9@mail.gmail.com> Message-ID: <191c3a030911011127g543d8464g8a7933f9e4860951@mail.gmail.com> #4 makes no sense to me. Are you just trying to create a call that the channel does not participate in? Then for sure you want to use bgapi to cause the originate to happen in the background. Also your gcore report has to be taken while you have the stuck channels to see why they are stuck. I can promise you this problem should be filed under misuse/abuse. On Sat, Oct 31, 2009 at 8:45 PM, Dome Charoenyost wrote: > How to use bgapi in my flow. > 1. user call did > 2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey) > 3. Hangup (by use or timeout) > 4. FS callback to user and bridge to IVR > > I'm not sure bgapi can do after channel hangup. > > Dome C. > > > 2009/11/1 Anthony Minessale : > > Use bgapi originate ... > > > > On Oct 31, 2009 9:09 PM, "Dome Charoenyost" wrote: > > > > Yes. i user api_hangup_hook for do callback. > > > > > > may be need originate_timeout > > > > Dome C. > > > > > > 2009/11/1 Rupa Schomaker : > > > >> fscore_pb has been updated. > > Next time put in the pastebin url. This > >> one was 10911. > > Ok, a ... > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/ad4a0733/attachment.html From anthony.minessale at gmail.com Sun Nov 1 11:31:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 13:31:59 -0600 Subject: [Freeswitch-users] Rejecting a call from JavaScript In-Reply-To: References: Message-ID: <191c3a030911011131h6310f312u555af860889488c8@mail.gmail.com> try session.execute("hangup", "user_busy"); On Sun, Nov 1, 2009 at 8:24 AM, Yehavi Bourvine wrote: > Hello, > > We would like to handle an incoming call to a busy phone according > to user's prefference: Some want waiting call, some want to just reject the > call, and others want to send the call to voicemail. > > We have a small JavaScript which tests the status of the destination and > the user's will and tries to act accordingly. Our problem is how to send > busy. I tried session.hangup("USER_BUSY") but it always sends "temporary > unavailable" which causes the orignator to think that the destination is out > of order. > > What is the correct way to do so? > > Thanks! __Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/8865a324/attachment.html From anthony.minessale at gmail.com Sun Nov 1 11:40:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 13:40:21 -0600 Subject: [Freeswitch-users] CDR CSV variables In-Reply-To: <609781.88322.qm@web37502.mail.mud.yahoo.com> References: <609781.88322.qm@web37502.mail.mud.yahoo.com> Message-ID: <191c3a030911011140q505868c3p5a9c94b5decbf857@mail.gmail.com> you can make up your own variable and set whatever you want in there then add it to the template. On Fri, Oct 30, 2009 at 10:04 AM, DJB wrote: > I wonder if I don't want to have b-leg in cdr csv, is there any variables > that can give me the actual gateway ip address that is actually went out. > > For instance, if I have this in my dialplan: > > the only value that I can think of from cdr csv is to get > remote_ip_last_arg, but it would contains the whole line of both ip > addresses. > > Thank you, > Dorn B. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/f53517af/attachment.html From dome at tel.co.th Sun Nov 1 11:53:01 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 2 Nov 2009 02:53:01 +0700 Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session In-Reply-To: <191c3a030911011127g543d8464g8a7933f9e4860951@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com> <191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com> <8ccbff060910311945j5be2fa85j95e7e1b422dcc3a9@mail.gmail.com> <191c3a030911011127g543d8464g8a7933f9e4860951@mail.gmail.com> Message-ID: <8ccbff060911011153h63bed954qe154b716a2bf8099@mail.gmail.com> 2009/11/2 Anthony Minessale : > #4 makes no sense to me. This solution call "miss callback". some contry cost for toll free number is so high. but call to user cheaper. so when user want to use international call. they call to my DID and my system callback to them. > Are you just trying to create a call that the channel does not participate > in? > Then for sure you want to use bgapi to cause the originate to? happen in the > background. Give me example how bgapi work when user hangup call Dome C. > > Also your gcore report has to be taken while you have the stuck channels to > see why they are stuck. > I can promise you this problem should be filed under misuse/abuse. > > > On Sat, Oct 31, 2009 at 8:45 PM, Dome Charoenyost wrote: >> >> How to use bgapi in my flow. >> 1. user call did >> 2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey) >> 3. Hangup (by use or timeout) >> 4. FS callback to user and bridge to IVR >> >> I'm not sure ?bgapi can do after channel hangup. >> >> Dome C. >> >> >> 2009/11/1 Anthony Minessale : >> > Use bgapi originate ... >> > >> > On Oct 31, 2009 9:09 PM, "Dome Charoenyost" wrote: >> > >> > Yes. i user api_hangup_hook for do callback. >> > >> > >> > may be need originate_timeout >> > >> > Dome C. >> > >> > >> > 2009/11/1 Rupa Schomaker : >> > >> >> fscore_pb has been updated. > > Next time put in the pastebin url. >> >> ?This >> >> one was 10911. > > Ok, a ... >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Sun Nov 1 12:04:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 14:04:41 -0600 Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session In-Reply-To: <8ccbff060911011153h63bed954qe154b716a2bf8099@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com> <191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com> <8ccbff060910311945j5be2fa85j95e7e1b422dcc3a9@mail.gmail.com> <191c3a030911011127g543d8464g8a7933f9e4860951@mail.gmail.com> <8ccbff060911011153h63bed954qe154b716a2bf8099@mail.gmail.com> Message-ID: <191c3a030911011204r6a74b4e2u108c2943d14d0ee4@mail.gmail.com> if you set the variable to: "originate foo bar baz" "bgapi originate foo bar baz" bgapi is itself an api command that takes the argument string and runs it in a separate thread and returns instantly. This will stop it from hanging at that point. On Sun, Nov 1, 2009 at 1:53 PM, Dome Charoenyost wrote: > 2009/11/2 Anthony Minessale : > > #4 makes no sense to me. > This solution call "miss callback". some contry cost for toll free > number is so high. but call to user cheaper. so when user want to > use international call. they call to my DID and my system callback to them. > > > Are you just trying to create a call that the channel does not > participate > > in? > > Then for sure you want to use bgapi to cause the originate to happen in > the > > background. > Give me example how bgapi work when user hangup call > > > > Dome C. > > > > > Also your gcore report has to be taken while you have the stuck channels > to > > see why they are stuck. > > I can promise you this problem should be filed under misuse/abuse. > > > > > > On Sat, Oct 31, 2009 at 8:45 PM, Dome Charoenyost > wrote: > >> > >> How to use bgapi in my flow. > >> 1. user call did > >> 2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey) > >> 3. Hangup (by use or timeout) > >> 4. FS callback to user and bridge to IVR > >> > >> I'm not sure bgapi can do after channel hangup. > >> > >> Dome C. > >> > >> > >> 2009/11/1 Anthony Minessale : > >> > Use bgapi originate ... > >> > > >> > On Oct 31, 2009 9:09 PM, "Dome Charoenyost" wrote: > >> > > >> > Yes. i user api_hangup_hook for do callback. > >> > > >> > > >> > may be need originate_timeout > >> > > >> > Dome C. > >> > > >> > > >> > 2009/11/1 Rupa Schomaker : > >> > > >> >> fscore_pb has been updated. > > Next time put in the pastebin url. > >> >> This > >> >> one was 10911. > > Ok, a ... > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/38463b48/attachment-0001.html From anthony.minessale at gmail.com Sun Nov 1 12:20:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 14:20:55 -0600 Subject: [Freeswitch-users] SIP provider with extern rtp server In-Reply-To: References: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com> <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org> <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org> <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org> Message-ID: <191c3a030911011220m22d7b515kda412e1fd9408f59@mail.gmail.com> Session-Expires: -1;refresher=uas nta: 200 OK has fatal syntax errors This is a know-bug in asterisk. see: https://issues.asterisk.org/view.php?id=15621 On Sun, Nov 1, 2009 at 4:40 AM, Ivan C Myrvold wrote: > No one have any idea why this is not working? I have combed through > the log, but couldn't find any clue there. > Incoming calls from my sip provider is working perfect, but for > outgoing calls it looks like Freeswitch is not letting the incoming > rtp to the local sip phone. > > Ivan > > On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote: > > > Yes, now I got a more detailed trace. Thank you for helping me with > > this. > > > > A new pastebin at http://pastebin.freeswitch.org/10905 > > > > Ivan > > > > Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable: > > > >> fsctl loglevel debug > >> console loglevel debug > >> sofia profile internal siptrace on > >> sofia profile external siptrace on > >> sofia loglevel all 9 > >> ^^^^^^^^^^^^^^^^^^^^^ > >> > >> Then run your call, then do this: > >> > >> sofia loglevel all 0 > >> sofia profile external siptrace off > >> sofia profile internal siptrace off > >> fsctl loglevel warning > >> console loglevel warning > >> > >> On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold > >> wrote: > >>> I have already set debug to 9, on both profiles. > >>> > >>> Ivan > >>> > >>> > >>> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable: > >>> > >>>> See that 200 OK that keeps coming in over and over and over and > >>>> over > >>>> again? That's because they never received your ACK. If you can > >>>> turn on > >>>> sofia loglevel to 9 and then watch where you send the ACK, you will > >>>> probably have your answer to why the other system did not receive > >>>> it. > >>>> If you're still not sure what's going on, post another pastebin > >>>> with > >>>> sofia loglevel set to 9. > >>>> > >>>> > >>>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold > >>>> wrote: > >>>>> Oh, what happened to it? > >>>>> Anyway, here is a new pb: > >>>>> http://pastebin.freeswitch.org/10867 > >>>>> Ivan > >>>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins: > >>>>> > >>>>> > >>>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold > >>>>> wrote: > >>>>>> > >>>>>> Here is a debug log from a call from an internal phone out to an > >>>>>> external (my iPhone with nbr 91316356): > >>>>>> http://pastebin.freeswitch.org/108578 > >>>>>> > >>>>>> Ivan > >>>>>> > >>>>> Uh... you wanna try that PB number again? > >>>>> -MC > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > >>>>> freeswitch- > >>>>> users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > >>>>> freeswitch- > >>>>> users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>> -- > >>>> Eliot Gable > >>>> > >>>> "We do not inherit the Earth from our ancestors: we borrow it from > >>>> our > >>>> children." ~David Brower > >>>> > >>>> "I decided the words were too conservative for me. We're not > >>>> borrowing > >>>> from our children, we're stealing from them--and it's not even > >>>> considered to be a crime." ~David Brower > >>>> > >>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > >>>> live; not live to eat.) ~Marcus Tullius Cicero > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>>> users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Eliot Gable > >> > >> "We do not inherit the Earth from our ancestors: we borrow it from > >> our > >> children." ~David Brower > >> > >> "I decided the words were too conservative for me. We're not > >> borrowing > >> from our children, we're stealing from them--and it's not even > >> considered to be a crime." ~David Brower > >> > >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > >> live; not live to eat.) ~Marcus Tullius Cicero > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/4b8be022/attachment.html From dujinfang at gmail.com Sun Nov 1 17:37:26 2009 From: dujinfang at gmail.com (Seven Du) Date: Mon, 2 Nov 2009 09:37:26 +0800 Subject: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session In-Reply-To: <23f91030910311924of53ae2ega246cf367f4a98a7@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms> <191c3a030910311426of22a4fdyb6be7cabfb992bd6@mail.gmail.com> <23f91030910311924of53ae2ega246cf367f4a98a7@mail.gmail.com> Message-ID: <23f91030911011737l70a39195xb7a5e69df51f4865@mail.gmail.com> Just suspicious would be possible that happened on sqlite stage? I manually deleted the channels from sqlite and nothing bad happend. just FYI. ---------- Forwarded message ---------- From: Seven Du Date: Sun, 1 Nov 2009 10:24:32 +0800 Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session To: Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound and outbound. I must be very careful if I upgrade to trunk and turn on rwlock debug because it's on production and the the problem not happening that much so would be hard to trace. But I will find time to test and report back. FYI, I also noticed that in some zombile channels, couple of INVITEs sent but never got a response. However, it sent out an ACK at last. From lei.tlfly at gmail.com Sun Nov 1 18:26:59 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Mon, 2 Nov 2009 10:26:59 +0800 Subject: [Freeswitch-users] Get error "415 Unsupported Media Type" when receiving call from softswitch Message-ID: <50c41b4e0911011826s3431a0cex79d4ba7eee79c872@mail.gmail.com> Hi all, I get a "415 Unsupported Media Type" when FS receiving call from a softswitch. I captured some packets, It seems that the softswitch use SIP-I protocol, does FS can handle SIP-I message? ===here is the invite messagefrom softswitch INVITE sip:xxxxx at xxxx:5060;user=phone SIP/2.0 Contact: MIME-version: 1.0 Content-Type: multipart/mixed;boundary=Alcatel-boundary To: From: xxxx;tag=73D332463135364195291201 P-Asserted-Identity: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE Supported: 100rel,timer,replaces,diversion Expires: 155 Session-Expires: 1800 Min-SE: 90 Call-ID: 01FD034872814000000230A1 at sip-3 Max-Forwards: 70 CSeq: 1 INVITE Timestamp: 10645 Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E Content-Length: 542 --Alcatel-boundary Content-Type: application/sdp v=0 o=- 5 8 IN IP4 xxxxxx s=SDP Data c=IN IP4 xxxxx t=0 0 m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97 a=rtpmap:110 speex/8000/1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 TELEPHONE-EVENT/8000 a=fmtp:110 mode=3 a=rtpmap:111 speex/8000/1 a=rtpmap:3 GSM/8000 a=ptime:30 a=fmtp:111 mode=2 a=rtpmap:112 speex/8000/1 --Alcatel-boundary Content-Type: application/ISUP;version=N/A .. .... ...0.D... ...V..0......1...n..p... --Alcatel-boundary-- =====response message from FS SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E From: xxxxx ;tag=73D332463135364195291201 To: ;tag=m6XS44BKF4pSS Call-ID: 01FD034872814000000230A1 at sip-3 CSeq: 1 INVITE Timestamp: 10645 0.000000 User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/d8ab7972/attachment-0001.html From lei.tlfly at gmail.com Sun Nov 1 20:09:25 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Mon, 2 Nov 2009 12:09:25 +0800 Subject: [Freeswitch-users] Get error "415 Unsupported Media Type" when receiving call from softswitch In-Reply-To: <50c41b4e0911011826s3431a0cex79d4ba7eee79c872@mail.gmail.com> References: <50c41b4e0911011826s3431a0cex79d4ba7eee79c872@mail.gmail.com> Message-ID: <50c41b4e0911012009y2b2bbfa3qa4713854843c968f@mail.gmail.com> FYI, Here is the log when I set sofia loglevel all 9 ============== tport_wakeup_pri(00DFE3E8): events IN tport_recv_event(00DFE3E8) tport(00DFE3E8) msg 01B2E0C0 from (udp/MyIP:5060) has 1315 bytes, veclen = 1 tport(00DFE3E8): msg 01B2E0C0 (1315 bytes) from udp/SSIP:5060/sip next=000 00000 nta: received INVITE sip:AAA at MyIP:5060;user=phone SIP/2.0 (CSeq 1) nta: canonizing sip:AAA at MyIP:5060 with contact nta: INVITE (1) going to a default leg nta: timer shortened to 200 ms tport_tsend(00DFE3E8) tpn = UDP/SSIP:5061 tport_resolve addrinfo = SSIP:5061 tport(00DFE3E8): not found by name UDP/SSIP:5061 tport_vsend(00DFE3E8): 652 bytes of 652 to udp/SSIP:5061 tport_vsend returned 652 nta: sent 415 Unsupported Media Type for INVITE (1) tport_wakeup_pri(00DFE3E8): events IN tport_recv_event(00DFE3E8) tport(00DFE3E8) msg 0192C510 from (udp/MyIP:5060) has 388 bytes, veclen = 1 tport(00DFE3E8): msg 0192C510 (388 bytes) from udp/SSIP:5060/sip next=0000 0000 nta: received ACK sip:AAA at MyIP:5060;user=phone SIP/2.0 (CSeq 1) nta: ACK (1) is going to INVITE (1) nta: timer set next to 937 ms nta: timer J fired, terminate 200 response incoming_reclaim_all(00000000, 00000000, 02E2FEB8) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 3859 ms nta: timer I fired, terminate 415 response incoming_reclaim_all(00000000, 00000000, 02E2FEB8) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set 2009/11/2 Lei Tang > Hi all, I get a "415 Unsupported Media Type" when FS receiving call from a > softswitch. I captured some packets, It seems that the softswitch use SIP-I > protocol, does FS can handle SIP-I message? > > ===here is the invite messagefrom softswitch > INVITE sip:xxxxx at xxxx:5060;user=phone SIP/2.0 > Contact: > MIME-version: 1.0 > Content-Type: multipart/mixed;boundary=Alcatel-boundary > To: > From: xxxx;tag=73D332463135364195291201 > P-Asserted-Identity: > Allow: > INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE > Supported: 100rel,timer,replaces,diversion > Expires: 155 > Session-Expires: 1800 > Min-SE: 90 > Call-ID: 01FD034872814000000230A1 at sip-3 > Max-Forwards: 70 > CSeq: 1 INVITE > Timestamp: 10645 > Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E > Content-Length: 542 > > --Alcatel-boundary > Content-Type: application/sdp > > v=0 > o=- 5 8 IN IP4 xxxxxx > s=SDP Data > c=IN IP4 xxxxx > t=0 0 > m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97 > a=rtpmap:110 speex/8000/1 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:96 TELEPHONE-EVENT/8000 > a=fmtp:110 mode=3 > a=rtpmap:111 speex/8000/1 > a=rtpmap:3 GSM/8000 > a=ptime:30 > a=fmtp:111 mode=2 > a=rtpmap:112 speex/8000/1 > > --Alcatel-boundary > Content-Type: application/ISUP;version=N/A > > .. .... > ...0.D... > ...V..0......1...n..p... > --Alcatel-boundary-- > > =====response message from FS > SIP/2.0 415 Unsupported Media Type > Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E > From: xxxxx ;tag=73D332463135364195291201 > To: ;tag=m6XS44BKF4pSS > Call-ID: 01FD034872814000000230A1 at sip-3 > CSeq: 1 INVITE > Timestamp: 10645 0.000000 > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 > Accept: application/sdp > Accept-Encoding: > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Length: 0 > -- > Lei.Tang > lei.tlfly at gmail.com > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/daaead31/attachment.html From yehavi.bourvine at gmail.com Sun Nov 1 20:15:32 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 2 Nov 2009 06:15:32 +0200 Subject: [Freeswitch-users] Rejecting a call from JavaScript In-Reply-To: <191c3a030911011131h6310f312u555af860889488c8@mail.gmail.com> References: <191c3a030911011131h6310f312u555af860889488c8@mail.gmail.com> Message-ID: Thanks! It works! __Yehavi: 2009/11/1 Anthony Minessale > try session.execute("hangup", "user_busy"); > > > On Sun, Nov 1, 2009 at 8:24 AM, Yehavi Bourvine < > yehavi.bourvine at gmail.com> wrote: > >> Hello, >> >> We would like to handle an incoming call to a busy phone according >> to user's prefference: Some want waiting call, some want to just reject the >> call, and others want to send the call to voicemail. >> >> We have a small JavaScript which tests the status of the destination and >> the user's will and tries to act accordingly. Our problem is how to send >> busy. I tried session.hangup("USER_BUSY") but it always sends "temporary >> unavailable" which causes the orignator to think that the destination is out >> of order. >> >> What is the correct way to do so? >> >> Thanks! __Yehavi: >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/81bb8333/attachment.html From anthony.minessale at gmail.com Sun Nov 1 20:56:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 22:56:55 -0600 Subject: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session In-Reply-To: <191c3a030911012056j2991b3c6y3cda1b705b5f9df9@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms> <191c3a030910311426of22a4fdyb6be7cabfb992bd6@mail.gmail.com> <23f91030910311924of53ae2ega246cf367f4a98a7@mail.gmail.com> <23f91030911011737l70a39195xb7a5e69df51f4865@mail.gmail.com> <191c3a030911012056j2991b3c6y3cda1b705b5f9df9@mail.gmail.com> Message-ID: <191c3a030911012056y66441b8aya672a16fabe89f33@mail.gmail.com> We already concluded its your unacceptabe use of originate in hangup hook right? On Nov 1, 2009 7:45 PM, "Seven Du" wrote: Just suspicious would be possible that happened on sqlite stage? I manually deleted the channels from sqlite and nothing bad happend. just FYI. ---------- Forwarded message ---------- From: Seven Du Date: Sun, 1 Nov 2009 10:24:32 +0800 Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session To: Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound and outbound. I must be very careful if I upgrade to trunk and turn on rwlock debug because it's on production and the the problem not happening that much so would be hard to trace. But I will find time to test and report back. FYI, I also noticed that in some zombile channels, couple of INVITEs sent but never got a response. However, it sent out an ACK at last. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at list... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/2da278e1/attachment.html From lei.tlfly at gmail.com Mon Nov 2 01:24:52 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Mon, 2 Nov 2009 17:24:52 +0800 Subject: [Freeswitch-users] Get error "415 Unsupported Media Type" whenreceiving call from softswitch In-Reply-To: <50c41b4e0911020117w27ad3ca2w99590117cb1925ad@mail.gmail.com> References: <50c41b4e0911011826s3431a0cex79d4ba7eee79c872@mail.gmail.com> <2CEBE489DC2CE140B7983073B17FB3D6E5B075@301081ANEX2.global.avaya.com> <50c41b4e0911020117w27ad3ca2w99590117cb1925ad@mail.gmail.com> Message-ID: <50c41b4e0911020124s27b7203bj5d5224288e1bb790@mail.gmail.com> Hi all, The problem is solved. I ask the softswitch to send only sdp in INVITE message, then It works. I think sofia doesn't support multipart content currently. is it right? 2009/11/2 Lei Tang > Hi Daniel. > Sure. pls email me to tlfly at hotmail.com. > > 2009/11/2 Zeng, Qinglan (Daniel) > > Lei, >> >> This is Daniel Zeng and I got your email address from the maillist of FS. >> I have personal interestings on FS and if possible can we have a talk on >> this? >> >> Thanks >> Daniel Zeng >> >> ------------------------------ >> *From:* Lei Tang [mailto:lei.tlfly at gmail.com] >> *Sent:* Monday, November 02, 2009 10:27 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Get error "415 Unsupported Media Type" >> whenreceiving call from softswitch >> >> Hi all, I get a "415 Unsupported Media Type" when FS receiving call from >> a softswitch. I captured some packets, It seems that the softswitch use >> SIP-I protocol, does FS can handle SIP-I message? >> >> ===here is the invite messagefrom softswitch >> INVITE sip:xxxxx at xxxx:5060;user=phone SIP/2.0 >> Contact: >> MIME-version: 1.0 >> Content-Type: multipart/mixed;boundary=Alcatel-boundary >> To: >> From: xxxx;tag=73D332463135364195291201 >> P-Asserted-Identity: >> Allow: >> INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE >> Supported: 100rel,timer,replaces,diversion >> Expires: 155 >> Session-Expires: 1800 >> Min-SE: 90 >> Call-ID: 01FD034872814000000230A1 at sip-3 >> Max-Forwards: 70 >> CSeq: 1 INVITE >> Timestamp: 10645 >> Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E >> Content-Length: 542 >> >> --Alcatel-boundary >> Content-Type: application/sdp >> >> v=0 >> o=- 5 8 IN IP4 xxxxxx >> s=SDP Data >> c=IN IP4 xxxxx >> t=0 0 >> m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97 >> a=rtpmap:110 speex/8000/1 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:96 TELEPHONE-EVENT/8000 >> a=fmtp:110 mode=3 >> a=rtpmap:111 speex/8000/1 >> a=rtpmap:3 GSM/8000 >> a=ptime:30 >> a=fmtp:111 mode=2 >> a=rtpmap:112 speex/8000/1 >> >> --Alcatel-boundary >> Content-Type: application/ISUP;version=N/A >> >> .. .... >> ...0.D... >> ...V..0......1...n..p... >> --Alcatel-boundary-- >> >> =====response message from FS >> SIP/2.0 415 Unsupported Media Type >> Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E >> From: xxxxx ;tag=73D332463135364195291201 >> To: ;tag=m6XS44BKF4pSS >> Call-ID: 01FD034872814000000230A1 at sip-3 >> CSeq: 1 INVITE >> Timestamp: 10645 0.000000 >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 >> Accept: application/sdp >> Accept-Encoding: >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Length: 0 >> -- >> Lei.Tang >> lei.tlfly at gmail.com >> > > > > -- > Lei.Tang > lei.tlfly at gmail.com > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/3504a8c7/attachment-0001.html From dujinfang at gmail.com Mon Nov 2 02:03:39 2009 From: dujinfang at gmail.com (Seven Du) Date: Mon, 2 Nov 2009 18:03:39 +0800 Subject: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session In-Reply-To: <191c3a030911012056y66441b8aya672a16fabe89f33@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms> <191c3a030910311426of22a4fdyb6be7cabfb992bd6@mail.gmail.com> <23f91030910311924of53ae2ega246cf367f4a98a7@mail.gmail.com> <23f91030911011737l70a39195xb7a5e69df51f4865@mail.gmail.com> <191c3a030911012056j2991b3c6y3cda1b705b5f9df9@mail.gmail.com> <191c3a030911012056y66441b8aya672a16fabe89f33@mail.gmail.com> Message-ID: <23f91030911020203i182536bfn2e51604e5909a5c3@mail.gmail.com> No, I'm Seven and never used hangup hook. you must had though I was Dome. Sorry, I'm not tend to hijack this thread, just though it's the same topic. 2009/11/2 Anthony Minessale > We already concluded its your unacceptabe use of originate in hangup hook > right? > > On Nov 1, 2009 7:45 PM, "Seven Du" wrote: > > Just suspicious would be possible that happened on sqlite stage? I > manually deleted the channels from sqlite and nothing bad happend. > just FYI. > > ---------- Forwarded message ---------- > From: Seven Du > Date: Sun, 1 Nov 2009 10:24:32 +0800 > Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session > To: > > Thank you Anthony. We are on r14696 and no non-standard mods loaded. we > even > unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both > inbound > and outbound. I must be very careful if I upgrade to trunk and turn on > rwlock debug because it's on production and the the problem not happening > that much so would be hard to trace. But I will find time to test and > report > back. > > FYI, I also noticed that in some zombile channels, couple of INVITEs sent > but never got a response. However, it sent out an ACK at last. > > _______________________________________________ FreeSWITCH-users mailing > list FreeSWITCH-users at list... > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/6e73cdf0/attachment.html From mariusz_kolo at wp.pl Mon Nov 2 03:32:56 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=) Date: Mon, 02 Nov 2009 12:32:56 +0100 Subject: [Freeswitch-users] Problem with hangin bri In-Reply-To: <87f2f3b90910281528t633b765x7c2bb858f41ce154@mail.gmail.com> References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com> <87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com> <4AE8B3B6.9000106@wp.pl> <87f2f3b90910281528t633b765x7c2bb858f41ce154@mail.gmail.com> Message-ID: <4AEEC368.5020204@wp.pl> Hi pastebin: http://pastebin.freeswitch.org/10926 and http://pastebin.freeswitch.org/10927 .We invoke calls from one voip phone to cell phone, and vice versa, but when i make inbound and outbound connection in nearly same time something goes wrong with chanells Thanks Michael Collins pisze: > Thanks. Can you collect debug logs of this happening? See > http://wiki.freeswitch.org/wiki/Reporting_Bugs for helpful tips on > collecting debug information. Use pastebin to dump all the log info > and reply here with the link. We don't have too many BRI users but I > believe there are a few so hopefully we can help you get up and running. > -MC > > 2009/10/28 Mariusz Ko?odziejczyk > > > Hi > > I'm also working on this project, so i can answer your questions > > Which version of FreeSWITCH are you running? > > FreeSWITCH Version 1.0.trunk (15246) > > Which PRI library are you using? > openzap Native stack > > openzap.conf > > [span zt BRI1] > trunk_type => bri > b-channel => 1-2 > d-channel=> 3 > > openzap.conf > > > > > > > > > > > > > > > > > Which BRI card are you using? > > Producer: http://www.phoniceq.com/ > card model: http://quadbri.phoniceq.com/ > > Card instalation process (instruction from producer) > > 1) download bristuff staff from > > http://junghanns.net/downloads/bristuff-0.4.0-RC3h.tar.gz > or > http://junghanns.net/downloads/bristuff-0.3.0-PRE-1y-z.tar.gz > > unpack it and go to bristuff-* > > 2) download patcher from > http://quadbri.phoniceq.com/driver/bristuff/qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch > > patch it using > > patch -p0 < qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch > > 3) you can check card using zttest (result should be 99.x) > > Producer has said, that we are first client, it wants to use this > card in freeswitch > > we are using 1 port (S/T interface). Our NT is "NT1 plus 2b1q" > > > Thanks > > Michael Collins pisze: > > Okay, obligatory questions: > > Which version of FreeSWITCH are you running? > > Which PRI library are you using? > > Which BRI card are you using? > > > > -MC > > > > On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawli?ski > > > >> wrote: > > > > Hi, > > I have some problems with bri status. I have 3 chanel isdn > modem, > > and zaptel compatible quad bri card. I can invoke calls from my > > voip phone to cell phone, and vice versa, but when i make > inbound > > and outbound connection in nearly same time something goes wrong > > with chanells and after few calls all of them has hangup status. > > > > There is log about that in attachement, see "is already in use > > waiting for it to become available." phrase. Time of this > event is > > about 14:43:35. Unload and Load open_zap module helped, but its > > not an solution because of lost connections. > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Mariusz Ko?odziejczyk > > Advanced Developing Architecture S.C. > > tel. : +48 609 381 316 > e-mail : mariusz_kolo at wp.pl > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Mariusz Ko?odziejczyk Advanced Developing Architecture S.C. tel. : +48 609 381 316 e-mail : mariusz_kolo at wp.pl From dipen at entvoice.com Mon Nov 2 03:53:51 2009 From: dipen at entvoice.com (dipen at entvoice.com) Date: Mon, 2 Nov 2009 06:53:51 -0500 Subject: [Freeswitch-users] Java example Message-ID: <44498.1257162831@entvoice.com> Hi, Can you please paste me your sample java dialplan code that work for you ? ..coz m also facing the same problem. My mod_java is loaded properly. Also /usr/lib/jvm/java-1.5.0-gcj-4.3-1.5.0.0/jre/lib/i386/client/libjvm.so and freeswitch.jar in java.conf.xml is specified properly. I have written a java code to print HIIIIIIIIII on the console but its not printing. Level mentioned is INFO. on FS console it just prints EXECUTE sofia/internal/1004 at 192.168.1.144:5061 java(testing.class) I am attaching my java code herewith. Can u just tell me where more i should do the modification to get my dialplan work. Waiting for your kind reply. Thanks & Regards, Dipen Velani On Fri 19/12/08 4:09 AM , kriko wrote: > Seems like my dialplan was a bit problematic, it works now. > Thanks. > On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale > wrote: > > did you turn up your console log level high enough to see it? The > default > > level is "INFO" > > > > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > -- > kriko > _______________________________________________ > Freeswitch-users mailing list > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ---- Msg sent via @Mail - http://atmail.com/ -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: java_sample_code.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/9b737d5a/attachment.txt From dipen at entvoice.com Mon Nov 2 03:53:51 2009 From: dipen at entvoice.com (dipen at entvoice.com) Date: Mon, 2 Nov 2009 06:53:51 -0500 Subject: [Freeswitch-users] Java example Message-ID: <44498.1257162831@entvoice.com> Hi, Can you please paste me your sample java dialplan code that work for you ? ..coz m also facing the same problem. My mod_java is loaded properly. Also /usr/lib/jvm/java-1.5.0-gcj-4.3-1.5.0.0/jre/lib/i386/client/libjvm.so and freeswitch.jar in java.conf.xml is specified properly. I have written a java code to print HIIIIIIIIII on the console but its not printing. Level mentioned is INFO. on FS console it just prints EXECUTE sofia/internal/1004 at 192.168.1.144:5061 java(testing.class) I am attaching my java code herewith. Can u just tell me where more i should do the modification to get my dialplan work. Waiting for your kind reply. Thanks & Regards, Dipen Velani On Fri 19/12/08 4:09 AM , kriko wrote: > Seems like my dialplan was a bit problematic, it works now. > Thanks. > On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale > wrote: > > did you turn up your console log level high enough to see it? The > default > > level is "INFO" > > > > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > -- > kriko > _______________________________________________ > Freeswitch-users mailing list > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ---- Msg sent via @Mail - http://atmail.com/ -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: java_sample_code.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/9b737d5a/attachment-0001.txt From info at daccii.it Mon Nov 2 05:04:24 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Mon, 02 Nov 2009 14:04:24 +0100 Subject: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone after the first call using OpenZAP (analog spans) In-Reply-To: <4AEC303F.4020207@daccii.it> References: <4AE8B20D.4040608@daccii.it> <191c3a030910290912x9489784s4cc8e224363f3796@mail.gmail.com> <87f2f3b90910291026u2c62ded6v5b9458a9c2e85877@mail.gmail.com> <4AEC303F.4020207@daccii.it> Message-ID: <4AEED8D8.6050303@daccii.it> Hi, i've done more and more tests ... the result is the same :\ I've tried previous freeswitch version (1.0.2, 1.0.3), lastest stable (1.0.4) and with svn (updated at revision 15315 while openzap revision is 847). I've tried with ubuntu zaptel modules (1.4.10), with and without octvqe soft echo, with another card, that uses wctdm instead of opvxa1200, with lastest manually compiled zaptel modules (1.4.12.1) but, yet, nothing to do. I tested openzap test utilities but them works well (only a little change to testanalog.c to use it tones instead of us). The problem is ever the same, i can do the first call but i can't do more. I put logs and config files into freeswitch pastebin, here links: - FULL STARTUP http://pastebin.freeswitch.org/10930 - ZAPTEL STARTUP http://pastebin.freeswitch.org/10929 - FIRST CALL http://pastebin.freeswitch.org/10928 - SECOND CALL http://pastebin.freeswitch.org/10931 - INCOMING CALL http://pastebin.freeswitch.org/10939 - SHUTDOWN http://pastebin.freeswitch.org/10932 - OPENVOX DMESG http://pastebin.freeswitch.org/10933 - zaptel.conf http://pastebin.freeswitch.org/10934 - openzap.conf http://pastebin.freeswitch.org/10935 - zt.conf http://pastebin.freeswitch.org/10936 - openzap.conf.xml http://pastebin.freeswitch.org/10937 - openzap dialplan for outgoing/incoming calls http://pastebin.freeswitch.org/10938 As you can see from logs freeswitch doesn't reknow free dial tone after the first call, but i don't understand why. I'll join on irc in short, hope to find some help :) Best Regards, Daniele Albano Daniele Salvatore - Lavoro ha scritto: > Hi, > > i've done more tests, with svn too, but nothing to do. The strange thing > is when freeswitch shutdown and close zaptel channel i get back the dial > tone from my provider (i'm checking trought ztmonitor 1 -vv). Can be > something is missing when span get closed? > > PS1: with svn i can do only one call, but i can receive calls without > problems > PS2: i joined yesterday on irc but i had few time ... hope to join monday > > > Best Regards, > Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/cbc035a6/attachment-0001.vcf From hjqlopez at hotmail.com Mon Nov 2 06:36:50 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Mon, 2 Nov 2009 09:36:50 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media Message-ID: Thanks for you answers guys, I test the parameters you suggested but still no audio due to the lack of reINVITE.? By the way I'm using 1.0.4 but I also tried 1.0.5pre3. One particular condition is that there is no on-hold before the Blind Transfer. Regards, Humberto >? >? >> My scenario is as follows: >> >> inbound-bypass-media is set in the profile because we dont want FS handling >> the media. >> >> 1. A calls B >> 2. FS sends to B the A's SDP >> 3. B answers >> 4. FS sends to A the B's SDP >> 5. Media going directly between A and B >> 6. B REFERs the call to C (blind transfer with no reINVITE for Hold) >> 7. FS accepts(202) the REFER and sends the NOTIFY >> 7a. B and FS send the BYE >> 8. FS sends an INVITE? to C with A's SDP >> 9. C answers >> 10. FS doesn't send a reINVITE to A to let it know about C's SDP >> >> >> Is that the expected FS behavior or is this a bug? _________________________________________________________________ Ready for a deal-of-a-lifetime? See fantastic offers on Windows 7, in one convenient place. http://go.microsoft.com/?linkid=9691634 From mike at jerris.com Mon Nov 2 06:45:26 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 2 Nov 2009 09:45:26 -0500 Subject: [Freeswitch-users] Get error "415 Unsupported Media Type" whenreceiving call from softswitch In-Reply-To: <50c41b4e0911020124s27b7203bj5d5224288e1bb790@mail.gmail.com> References: <50c41b4e0911011826s3431a0cex79d4ba7eee79c872@mail.gmail.com> <2CEBE489DC2CE140B7983073B17FB3D6E5B075@301081ANEX2.global.avaya.com> <50c41b4e0911020117w27ad3ca2w99590117cb1925ad@mail.gmail.com> <50c41b4e0911020124s27b7203bj5d5224288e1bb790@mail.gmail.com> Message-ID: <724DAF7A-537D-4588-AE73-7B06076DE78E@jerris.com> That is correct. Mike On Nov 2, 2009, at 4:24 AM, Lei Tang wrote: > Hi all, > The problem is solved. I ask the softswitch to send only sdp in > INVITE message, then It works. > I think sofia doesn't support multipart content currently. is it > right? From mike at jerris.com Mon Nov 2 06:49:22 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 2 Nov 2009 09:49:22 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <177068F1-7F95-4AB2-AF60-E1B367B49213@jerris.com> Please re-try with latest svn trunk. Mike On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote: > > Thanks for you answers guys, > > I test the parameters you suggested > but still no audio due to the lack of reINVITE. By the way I'm using > 1.0.4 but I also tried 1.0.5pre3. > > One particular condition is that there is no on-hold before the > Blind Transfer. > > Regards, > > Humberto > >> >> From anthony.minessale at gmail.com Mon Nov 2 07:32:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Nov 2009 09:32:54 -0600 Subject: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session In-Reply-To: <23f91030911020203i182536bfn2e51604e5909a5c3@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms> <191c3a030910311426of22a4fdyb6be7cabfb992bd6@mail.gmail.com> <23f91030910311924of53ae2ega246cf367f4a98a7@mail.gmail.com> <23f91030911011737l70a39195xb7a5e69df51f4865@mail.gmail.com> <191c3a030911012056j2991b3c6y3cda1b705b5f9df9@mail.gmail.com> <191c3a030911012056y66441b8aya672a16fabe89f33@mail.gmail.com> <23f91030911020203i182536bfn2e51604e5909a5c3@mail.gmail.com> Message-ID: <191c3a030911020732r4b043c52r975ea31a7b35e4d0@mail.gmail.com> Every time you have stuck channels at the last state it means something took control of the thread and did not release it. revisions other that current SVN trunk are not possible to debug because over one thousand changes have occurred since then. On Mon, Nov 2, 2009 at 4:03 AM, Seven Du wrote: > No, I'm Seven and never used hangup hook. you must had though I was Dome. > > Sorry, I'm not tend to hijack this thread, just though it's the same topic. > > 2009/11/2 Anthony Minessale > >> We already concluded its your unacceptabe use of originate in hangup hook >> right? >> >> On Nov 1, 2009 7:45 PM, "Seven Du" wrote: >> >> Just suspicious would be possible that happened on sqlite stage? I >> manually deleted the channels from sqlite and nothing bad happend. >> just FYI. >> >> ---------- Forwarded message ---------- >> From: Seven Du >> Date: Sun, 1 Nov 2009 10:24:32 +0800 >> Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session >> To: >> >> Thank you Anthony. We are on r14696 and no non-standard mods loaded. we >> even >> unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both >> inbound >> and outbound. I must be very careful if I upgrade to trunk and turn on >> rwlock debug because it's on production and the the problem not happening >> that much so would be hard to trace. But I will find time to test and >> report >> back. >> >> FYI, I also noticed that in some zombile channels, couple of INVITEs sent >> but never got a response. However, it sent out an ACK at last. >> >> _______________________________________________ FreeSWITCH-users mailing >> list FreeSWITCH-users at list... >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/61fc2204/attachment.html From ujjval at simplesignal.com Mon Nov 2 07:54:33 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Mon, 2 Nov 2009 07:54:33 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> , <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> Message-ID: <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> Yes, I think I did. However here is what furthur testing revelas. If I dial in from AT&T cell phone, I do not see any DTMF using Don's IVR.xml.conf to call my conf app. But when I dial the same number using a Verizon Cell, it works. When I dial a number that is provisioned to call the Conf App directly from the public.xml dialplan...it works even with the same AT&T cell phone... Strange behaviour ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris [mike at jerris.com] Sent: Saturday, October 31, 2009 11:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Have you answered the call? On Oct 30, 2009, at 11:34 AM, Rob Forman wrote: > Hm, strange. I haven't seen that before. Can you pastebin your logs > at debug level? > > On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > >> It's strange... a tcpdump tells me that there is no DTMF from my >> provider when using IVR, but when I call into a TN that goes >> directly into the Conference App, I see DTMF from the provider. >> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Mon Nov 2 08:08:29 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Nov 2009 10:08:29 -0600 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> , <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> Message-ID: <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> you know I have heard this before... It seems to ONLY be AT&T /b On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote: > Yes, I think I did. However here is what furthur testing revelas. If > I dial in from AT&T cell phone, I do not see any DTMF using Don's > IVR.xml.conf to call my conf app. But when I dial the same number > using a Verizon Cell, it works. > > When I dial a number that is provisioned to call the Conf App > directly from the public.xml dialplan...it works even with the same > AT&T cell phone... > > Strange behaviour From djbinter at yahoo.com Mon Nov 2 08:52:34 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 2 Nov 2009 08:52:34 -0800 (PST) Subject: [Freeswitch-users] CDR CSV variables In-Reply-To: <191c3a030911011140q505868c3p5a9c94b5decbf857@mail.gmail.com> References: <609781.88322.qm@web37502.mail.mud.yahoo.com> <191c3a030911011140q505868c3p5a9c94b5decbf857@mail.gmail.com> Message-ID: <430543.96247.qm@web37502.mail.mud.yahoo.com> Anthony, Yes, if you can advise, how would I detect whether it's going out to 192.168.1.4 or 192.168.1.5 without having to activate b-leg of the CDRs. Thank you, Dorn B. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Sun, November 1, 2009 11:40:21 AM Subject: Re: [Freeswitch-users] CDR CSV variables you can make up your own variable and set whatever you want in there then add it to the template. On Fri, Oct 30, 2009 at 10:04 AM, DJB wrote: I wonder if I don't want to have b-leg in cdr csv, is there any variables that can give me the actual gateway ip address that is actually went out. > >>For instance, if I have this in my dialplan: >> >>the only value that I can think of from cdr csv is to get remote_ip_last_arg, but it would contains the whole line of both ip addresses. > >>Thank you, >>Dorn B. > > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/f0173889/attachment-0001.html From djbinter at yahoo.com Mon Nov 2 08:56:34 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 2 Nov 2009 08:56:34 -0800 (PST) Subject: [Freeswitch-users] Freeswitch in signaling path only In-Reply-To: <312533.76921.qm@web37505.mail.mud.yahoo.com> References: <429925.60305.qm@web37508.mail.mud.yahoo.com> <200910301300.01766.chris@cloudtel.com> <312533.76921.qm@web37505.mail.mud.yahoo.com> Message-ID: <612508.14697.qm@web37508.mail.mud.yahoo.com> Any suggestion from anyone please? Thank you, Dorn B ----- Original Message ---- From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Fri, October 30, 2009 11:31:12 AM Subject: Re: [Freeswitch-users] Freeswitch in signaling path only Now i have as follows, but it's still the same result. By the way, I am running: FreeSWITCH Version 1.0.4 (exported) . . . session:execute("set","hangup_after_bridge=true") session:execute("set","continue_on_fail=true") session:execute("set","originate_timeout=2") session:execute("set","originate_retries=3") session:execute("set","progress_timeout=15") . . . while row do local gw_ip_address = row.gw_ip_address local cust_name = row.cust_name session:execute("set", "accountcode=" ..cust_name .. "") session:execute("set","bypass_media=true") session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.146") session:execute("set","bypass_media=true") session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.105") -- Block for testing -- session:execute("bridge","sofia/external/" .. called_num .. "@" .. gw_ip_address .."") row = cur:fetch (row, "a") end Here is the debug for switch_ivr_originate.c: 2009-10-30 11:09:52.877832 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:09:53.17811 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.1 46] 2009-10-30 11:09:54.285453 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:09:54.422426 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 105] 2009-10-30 11:09:55.694761 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:09:55.836036 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 146] 2009-10-30 11:09:57.107697 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:09:57.254664 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 105] 2009-10-30 11:12:03.129097 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:12:03.273055 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 146] 2009-10-30 11:12:04.546410 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:12:04.682661 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 105] 2009-10-30 11:12:15.781701 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 16 [NORMAL_CLEARING] 2009-10-30 11:12:33.349162 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:12:33.470989 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 146] 2009-10-30 11:12:34.724641 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2009-10-30 11:12:34.730634 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2009-10-30 11:12:34.750637 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] FIRST ROUTE: XX.XX.XX.146 and I tried to failed the first route and it gave 500 back, then it goes to the next one. SECOND ROUTE: XX.XX.XX.105 Thank you, Dorn B. ----- Original Message ---- From: Chris Burns To: freeswitch-users at lists.freeswitch.org Sent: Fri, October 30, 2009 10:00:01 AM Subject: Re: [Freeswitch-users] Freeswitch in signaling path only Do you have a debug level message from switch_ivr_originate.c in your log? "Channel is already up, delaying proxy mode 'till both legs are answered." Set bypass_media b4 each bridge. It is unsetting on you and setting bypass_media_after_bridge because you already answered the channel running the lua script. On October 30, 2009 12:03:29 pm DJB wrote: > I am wondering why I cannot do as condition#2. > > For Lua in dialplan, when I have the followings: > > > --WORKING-- > (Condition#1) > . > . > session:execute("set","bypass_media=true") > session:execute("set","hangup_after_bridge=true") > session:execute("set","continue_on_fail=true") > . > . > session:execute("bridge","sofia/external/" .. called_num .. > "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") . > . > > --NOT WORKING-- > (Condition#2) > Note: FS tries to be in media path and send re-invite. > . > . > session:execute("set","bypass_media=true") > session:execute("set","hangup_after_bridge=true") > session:execute("set","continue_on_fail=true") > . > . > session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1") > session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2") > . > . > > Thank you, > Dorn B. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Mon Nov 2 09:07:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Nov 2009 11:07:26 -0600 Subject: [Freeswitch-users] Freeswitch in signaling path only In-Reply-To: <612508.14697.qm@web37508.mail.mud.yahoo.com> References: <429925.60305.qm@web37508.mail.mud.yahoo.com> <200910301300.01766.chris@cloudtel.com> <312533.76921.qm@web37505.mail.mud.yahoo.com> <612508.14697.qm@web37508.mail.mud.yahoo.com> Message-ID: <191c3a030911020907o7718bc3ci7ebfee40519e01cb@mail.gmail.com> yes please use our latest SVN trunk if you plan to report issues because our first questions always is "can you reproduce this issue with our latest code" And please don't reply twice to the same email asking us to hurry up an answer you faster its only monday morning here. On Mon, Nov 2, 2009 at 10:56 AM, DJB wrote: > Any suggestion from anyone please? > > Thank you, > Dorn B > > > ----- Original Message ---- > From: DJB > To: freeswitch-users at lists.freeswitch.org > Sent: Fri, October 30, 2009 11:31:12 AM > Subject: Re: [Freeswitch-users] Freeswitch in signaling path only > > Now i have as follows, but it's still the same result. By the way, I am > running: FreeSWITCH Version 1.0.4 (exported) > . > . > . > session:execute("set","hangup_after_bridge=true") > session:execute("set","continue_on_fail=true") > session:execute("set","originate_timeout=2") > session:execute("set","originate_retries=3") > session:execute("set","progress_timeout=15") > . > . > . > while row do > local gw_ip_address = row.gw_ip_address > local cust_name = row.cust_name > session:execute("set", "accountcode=" ..cust_name .. "") > session:execute("set","bypass_media=true") > session:execute("bridge","sofia/external/" .. called_num .. > "@XX.XX.XX.XX.146") > session:execute("set","bypass_media=true") > session:execute("bridge","sofia/external/" .. called_num .. > "@XX.XX.XX.XX.105") > -- Block for testing -- session:execute("bridge","sofia/external/" .. > called_num .. "@" .. gw_ip_address .."") > row = cur:fetch (row, "a") > end > > Here is the debug for switch_ivr_originate.c: > > 2009-10-30 11:09:52.877832 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:09:53.17811 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.1 > 46] > 2009-10-30 11:09:54.285453 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:09:54.422426 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 105] > 2009-10-30 11:09:55.694761 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:09:55.836036 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 146] > 2009-10-30 11:09:57.107697 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:09:57.254664 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 105] > 2009-10-30 11:12:03.129097 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:12:03.273055 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 146] > 2009-10-30 11:12:04.546410 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:12:04.682661 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 105] > 2009-10-30 11:12:15.781701 [DEBUG] switch_ivr_originate.c:2138 Originate > Resulted in Error Cause: 16 [NORMAL_CLEARING] > 2009-10-30 11:12:33.349162 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:12:33.470989 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 146] > 2009-10-30 11:12:34.724641 [DEBUG] switch_ivr_originate.c:2138 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > 2009-10-30 11:12:34.730634 [DEBUG] switch_ivr_originate.c:2138 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > 2009-10-30 11:12:34.750637 [DEBUG] switch_ivr_originate.c:2138 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > > > FIRST ROUTE: XX.XX.XX.146 > and I tried to failed the first route and it gave 500 back, then it goes to > the next one. > SECOND ROUTE: XX.XX.XX.105 > > > Thank you, > Dorn B. > > > > ----- Original Message ---- > From: Chris Burns > To: freeswitch-users at lists.freeswitch.org > Sent: Fri, October 30, 2009 10:00:01 AM > Subject: Re: [Freeswitch-users] Freeswitch in signaling path only > > Do you have a debug level message from switch_ivr_originate.c in your log? > "Channel is already up, delaying proxy mode 'till both legs are answered." > > Set bypass_media b4 each bridge. It is unsetting on you and setting > bypass_media_after_bridge because you already answered the channel running > the lua script. > > On October 30, 2009 12:03:29 pm DJB wrote: > > I am wondering why I cannot do as condition#2. > > > > For Lua in dialplan, when I have the followings: > > > > > > --WORKING-- > > (Condition#1) > > . > > . > > session:execute("set","bypass_media=true") > > session:execute("set","hangup_after_bridge=true") > > session:execute("set","continue_on_fail=true") > > . > > . > > session:execute("bridge","sofia/external/" .. called_num .. > > "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") . > > . > > > > --NOT WORKING-- > > (Condition#2) > > Note: FS tries to be in media path and send re-invite. > > . > > . > > session:execute("set","bypass_media=true") > > session:execute("set","hangup_after_bridge=true") > > session:execute("set","continue_on_fail=true") > > . > > . > > session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1") > > session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2") > > . > > . > > > > Thank you, > > Dorn B. > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/8a3ef38e/attachment.html From anthony.minessale at gmail.com Mon Nov 2 09:09:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Nov 2009 11:09:44 -0600 Subject: [Freeswitch-users] CDR CSV variables In-Reply-To: <430543.96247.qm@web37502.mail.mud.yahoo.com> References: <609781.88322.qm@web37502.mail.mud.yahoo.com> <191c3a030911011140q505868c3p5a9c94b5decbf857@mail.gmail.com> <430543.96247.qm@web37502.mail.mud.yahoo.com> Message-ID: <191c3a030911020909r39a3687fq733189943bb07cd7@mail.gmail.com> if you enable debug on the cdr_csv module you will get a big dump of all the data you have available and you may be able to pick something out that indicates which one it was. On Mon, Nov 2, 2009 at 10:52 AM, DJB wrote: > Anthony, > > Yes, if you can advise, how would I detect whether it's going out to > 192.168.1.4 or 192.168.1.5 without having to activate b-leg of the CDRs. > > Thank you, > Dorn B. > > ------------------------------ > *From:* Anthony Minessale > > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Sun, November 1, 2009 11:40:21 AM > *Subject:* Re: [Freeswitch-users] CDR CSV variables > > you can make up your own variable and set whatever you want in there then > add it to the template. > > > On Fri, Oct 30, 2009 at 10:04 AM, DJB wrote: > >> I wonder if I don't want to have b-leg in cdr csv, is there any variables >> that can give me the actual gateway ip address that is actually went out. >> >> For instance, if I have this in my dialplan: >> >> the only value that I can think of from cdr csv is to get >> remote_ip_last_arg, but it would contains the whole line of both ip >> addresses. >> >> Thank you, >> Dorn B. >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/b1b99b88/attachment-0001.html From djbinter at yahoo.com Mon Nov 2 09:39:28 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 2 Nov 2009 09:39:28 -0800 (PST) Subject: [Freeswitch-users] Freeswitch in signaling path only In-Reply-To: <191c3a030911020907o7718bc3ci7ebfee40519e01cb@mail.gmail.com> References: <429925.60305.qm@web37508.mail.mud.yahoo.com> <200910301300.01766.chris@cloudtel.com> <312533.76921.qm@web37505.mail.mud.yahoo.com> <612508.14697.qm@web37508.mail.mud.yahoo.com> <191c3a030911020907o7718bc3ci7ebfee40519e01cb@mail.gmail.com> Message-ID: <147606.16838.qm@web37506.mail.mud.yahoo.com> I am really sorry. I did not mean to rush or anything. I've had a problem with my email many times, so I just want to make sure that my email gets there. Regards, Dorn B. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Mon, November 2, 2009 9:07:26 AM Subject: Re: [Freeswitch-users] Freeswitch in signaling path only yes please use our latest SVN trunk if you plan to report issues because our first questions always is "can you reproduce this issue with our latest code" And please don't reply twice to the same email asking us to hurry up an answer you faster its only monday morning here. On Mon, Nov 2, 2009 at 10:56 AM, DJB wrote: >Any suggestion from anyone please? > > >>Thank you, >>Dorn B > > >>----- Original Message ---- > >From: DJB >>To: freeswitch-users at lists.freeswitch.org > >Sent: Fri, October 30, 2009 11:31:12 AM >>Subject: Re: [Freeswitch-users] Freeswitch in signaling path only > >>Now i have as follows, but it's still the same result. By the way, I am running: FreeSWITCH Version 1.0.4 (exported) >>. >>. >>. >>session:execute("set","hangup_after_bridge=true") >>session:execute("set","continue_on_fail=true") >>session:execute("set","originate_timeout=2") >>session:execute("set","originate_retries=3") >>session:execute("set","progress_timeout=15") >>. >>. >>. >>while row do >>local gw_ip_address = row.gw_ip_address >>local cust_name = row.cust_name >>session:execute("set", "accountcode=" ..cust_name .. "") >>session:execute("set","bypass_media=true") >>session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.146") >>session:execute("set","bypass_media=true") >>session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.105") >>-- Block for testing -- session:execute("bridge","sofia/external/" .. called_num .. "@" .. gw_ip_address .."") >>row = cur:fetch (row, "a") >>end > >>Here is the debug for switch_ivr_originate.c: > >>2009-10-30 11:09:52.877832 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:09:53.17811 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.1 >>46] >>2009-10-30 11:09:54.285453 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:09:54.422426 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>105] >>2009-10-30 11:09:55.694761 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:09:55.836036 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>146] >>2009-10-30 11:09:57.107697 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:09:57.254664 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>105] >>2009-10-30 11:12:03.129097 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:12:03.273055 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>146] >>2009-10-30 11:12:04.546410 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:12:04.682661 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>105] >>2009-10-30 11:12:15.781701 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 16 [NORMAL_CLEARING] >>2009-10-30 11:12:33.349162 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:12:33.470989 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>146] >>2009-10-30 11:12:34.724641 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >>2009-10-30 11:12:34.730634 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >>2009-10-30 11:12:34.750637 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > > >>FIRST ROUTE: XX.XX.XX.146 >>and I tried to failed the first route and it gave 500 back, then it goes to the next one. >>SECOND ROUTE: XX.XX.XX.105 > > >>Thank you, >>Dorn B. > > > >>----- Original Message ---- >>From: Chris Burns >>To: freeswitch-users at lists.freeswitch.org >>Sent: Fri, October 30, 2009 10:00:01 AM >>Subject: Re: [Freeswitch-users] Freeswitch in signaling path only > >>Do you have a debug level message from switch_ivr_originate.c in your log? >>"Channel is already up, delaying proxy mode 'till both legs are answered." > >>Set bypass_media b4 each bridge. It is unsetting on you and setting >>bypass_media_after_bridge because you already answered the channel running >>the lua script. > >>On October 30, 2009 12:03:29 pm DJB wrote: >>> I am wondering why I cannot do as condition#2. >>> >>> For Lua in dialplan, when I have the followings: >>> >>> >>> --WORKING-- >>> (Condition#1) >>> . >>> . >>> session:execute("set","bypass_media=true") >>> session:execute("set","hangup_after_bridge=true") >>> session:execute("set","continue_on_fail=true") >>> . >>> . >>> session:execute("bridge","sofia/external/" .. called_num .. >>> "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") . >>> . >>> >>> --NOT WORKING-- >>> (Condition#2) >>> Note: FS tries to be in media path and send re-invite. >>> . >>> . >>> session:execute("set","bypass_media=true") >>> session:execute("set","hangup_after_bridge=true") >>> session:execute("set","continue_on_fail=true") >>> . >>> . >>> session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1") >>> session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2") >>> . >>> . >>> >>> Thank you, >>> Dorn B. >>> >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/9fc97856/attachment.html From shiyanov at gmail.com Mon Nov 2 10:51:59 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Mon, 2 Nov 2009 21:51:59 +0300 Subject: [Freeswitch-users] Java example In-Reply-To: <44498.1257162831@entvoice.com> References: <44498.1257162831@entvoice.com> Message-ID: Here is rather big and, let's say, complete example of mod_java usage: https://starpound.svn.sourceforge.net/svnroot/starpound/trunk/src/fs2agi The goal of this project is to be a proxy between FreeSwitch and server application which knows Asterisk AGI. On Mon, Nov 2, 2009 at 2:53 PM, wrote: > > Hi, > > Can you please paste me your sample java dialplan code that work for you ? > ..coz m also facing the same problem. > > My mod_java is loaded properly. > Also /usr/lib/jvm/java-1.5.0-gcj-4.3-1.5.0.0/jre/lib/i386/client/libjvm.so > and freeswitch.jar in java.conf.xml is specified properly. > > I have written a java code to print HIIIIIIIIII on the console but its not > printing. Level mentioned is INFO. > > on FS console it just prints > EXECUTE sofia/internal/1004 at 192.168.1.144:5061 java(testing.class) > > > I am attaching my java code herewith. > Can u just tell me where more i should do the modification to get my > dialplan work. > > Waiting for your kind reply. > > Thanks & Regards, > Dipen Velani > > On Fri 19/12/08 4:09 AM , kriko wrote: > > > Seems like my dialplan was a bit problematic, it works now. > > Thanks. > > On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale > > wrote: > > > did you turn up your console log level high enough to see it? The > > default > > > level is "INFO" > > > > > > > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > > > > > > > > > > -- > > kriko > > _______________________________________________ > > Freeswitch-users mailing list > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ---- Msg sent via @Mail - http://atmail.com/ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/ed780c43/attachment-0001.html From brian at freeswitch.org Mon Nov 2 11:07:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Nov 2009 13:07:08 -0600 Subject: [Freeswitch-users] Java example In-Reply-To: References: <44498.1257162831@entvoice.com> Message-ID: <85845D7B-9D9D-4BFA-ACCA-0F28DA4EBA9E@freeswitch.org> Is starpound involved in the FS Community? /b On Nov 2, 2009, at 12:51 PM, Artem Shiyanov wrote: > Here is rather big and, let's say, complete example of mod_java usage: > https://starpound.svn.sourceforge.net/svnroot/starpound/trunk/src/fs2agi > The goal of this project is to be a proxy between FreeSwitch and > server application which knows Asterisk AGI. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/3c5a6fc8/attachment.html From ujjval at simplesignal.com Mon Nov 2 11:51:39 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Mon, 2 Nov 2009 11:51:39 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> Message-ID: <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> Rob: Once I have the Moderator and Participants logged on, how do I invoke the moderator previlidges, LIk esay muting everyone/someone or kicking someone out of the Conf and the like? Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO? 80112 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Friday, October 30, 2009 9:34 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hm, strange. I haven't seen that before. Can you pastebin your logs at debug level? On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > It's strange... a tcpdump tells me that there is no DTMF from my > provider when using IVR, but when I call into a TN that goes > directly into the Conference App, I see DTMF from the provider. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 7:23 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > I've never had any problem with that. Is your logging at debug level > so you can see the RECV DTFM in the log/fs_cli? Are you calling from > a SIP phone on the pbx, or via a PSTN provider? Maybe your provider > isn't passing them through. > > Make sure your logging is turned up then try something simpler, like > calling the echo application, and see if DTFM comes through. > > Rob > > On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: > >> Rob: >> >> For some reason, I don't see the DTMF appear on the fs_CLI when >> using the below configuration....so it basically timesout. >> >> UK >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, October 26, 2009 9:21 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Thx a lot Rob, reading the wiki your way or using IVR seems correct.. >> =============== >> The wiki also says that the wait-mod might be "used in conjunction >> with an IVR where the moderators are authenticated with an extra >> pass- >> code", which is what I did. I guess that's why I didn't understand >> the point of the +pin. >> ====================== >> >> I will try it out. >> >> Again thx a lot for your help. Will keep everyone posted. >> >> ________________________________________ >> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >> Sent: Friday, October 23, 2009 12:22 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I just re-tested with the pin in my dial plan: >> >> >> >> And it doesn't challenge me for the pin. I just drop right in. I >> figured this is how it was intended, since the wiki says the pin is >> set initially and only challenged in later attempts [by future >> callers]: >> >> "The first time a conference name (confname) is used, it will be >> created on demand, and the pin will be set to what ever is specified >> at that time: the pin in the data string if specified, or if not, the >> "pin" setting in the conference profile, and if that is also >> unspecified, then there is no pin protection. Any later attempt to >> join the conference must specify the same pin number, if one existed >> when it was created. " >> >> >> The wiki also says that the wait-mod might be "used in conjunction >> with an IVR where the moderators are authenticated with an extra >> pass- >> code", which is what I did. I guess that's why I didn't understand >> the point of the +pin. >> >> I'm sure there's a scenario where its used and useful, the wiki just >> doesn't explain it. >> >> Rob >> >> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >> >>> Well first off you're not defining a pine here... >>> >>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>> pin >>> number] >>> >>> That might be why its not asking for a pin. >>> >>> /b >>> >>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>> >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ivan at myrvold.org Mon Nov 2 12:58:22 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Mon, 2 Nov 2009 21:58:22 +0100 Subject: [Freeswitch-users] SIP provider with extern rtp server In-Reply-To: <191c3a030911011220m22d7b515kda412e1fd9408f59@mail.gmail.com> References: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com> <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org> <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org> <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org> <191c3a030911011220m22d7b515kda412e1fd9408f59@mail.gmail.com> Message-ID: <03E9A3AC-F501-4FAE-8199-A4AEC4D60891@myrvold.org> That was it. My sip provider applied the patch to his Asterisk server that was referenced in the link you was so kind to provide, and again everything worked as it should. Thank you very much! Ivan Den 1. nov. 2009 kl. 21:20 skrev Anthony Minessale: > Session-Expires: -1;refresher=uas > > nta: 200 OK has fatal syntax errors > > This is a know-bug in asterisk. > > see: https://issues.asterisk.org/view.php?id=15621 > > > > On Sun, Nov 1, 2009 at 4:40 AM, Ivan C Myrvold > wrote: > No one have any idea why this is not working? I have combed through > the log, but couldn't find any clue there. > Incoming calls from my sip provider is working perfect, but for > outgoing calls it looks like Freeswitch is not letting the incoming > rtp to the local sip phone. > > Ivan > > On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote: > > > Yes, now I got a more detailed trace. Thank you for helping me with > > this. > > > > A new pastebin at http://pastebin.freeswitch.org/10905 > > > > Ivan > > > > Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable: > > > >> fsctl loglevel debug > >> console loglevel debug > >> sofia profile internal siptrace on > >> sofia profile external siptrace on > >> sofia loglevel all 9 > >> ^^^^^^^^^^^^^^^^^^^^^ > >> > >> Then run your call, then do this: > >> > >> sofia loglevel all 0 > >> sofia profile external siptrace off > >> sofia profile internal siptrace off > >> fsctl loglevel warning > >> console loglevel warning > >> > >> On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold > >> wrote: > >>> I have already set debug to 9, on both profiles. > >>> > >>> Ivan > >>> > >>> > >>> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable: > >>> > >>>> See that 200 OK that keeps coming in over and over and over and > >>>> over > >>>> again? That's because they never received your ACK. If you can > >>>> turn on > >>>> sofia loglevel to 9 and then watch where you send the ACK, you > will > >>>> probably have your answer to why the other system did not receive > >>>> it. > >>>> If you're still not sure what's going on, post another pastebin > >>>> with > >>>> sofia loglevel set to 9. > >>>> > >>>> > >>>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold > > >>>> wrote: > >>>>> Oh, what happened to it? > >>>>> Anyway, here is a new pb: > >>>>> http://pastebin.freeswitch.org/10867 > >>>>> Ivan > >>>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins: > >>>>> > >>>>> > >>>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold > > >>>>> wrote: > >>>>>> > >>>>>> Here is a debug log from a call from an internal phone out to > an > >>>>>> external (my iPhone with nbr 91316356): > >>>>>> http://pastebin.freeswitch.org/108578 > >>>>>> > >>>>>> Ivan > >>>>>> > >>>>> Uh... you wanna try that PB number again? > >>>>> -MC > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > >>>>> freeswitch- > >>>>> users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > >>>>> freeswitch- > >>>>> users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>> -- > >>>> Eliot Gable > >>>> > >>>> "We do not inherit the Earth from our ancestors: we borrow it > from > >>>> our > >>>> children." ~David Brower > >>>> > >>>> "I decided the words were too conservative for me. We're not > >>>> borrowing > >>>> from our children, we're stealing from them--and it's not even > >>>> considered to be a crime." ~David Brower > >>>> > >>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat > to > >>>> live; not live to eat.) ~Marcus Tullius Cicero > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > freeswitch- > >>>> users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Eliot Gable > >> > >> "We do not inherit the Earth from our ancestors: we borrow it from > >> our > >> children." ~David Brower > >> > >> "I decided the words were too conservative for me. We're not > >> borrowing > >> from our children, we're stealing from them--and it's not even > >> considered to be a crime." ~David Brower > >> > >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > >> live; not live to eat.) ~Marcus Tullius Cicero > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/e9ab9ee5/attachment-0001.html From msc at freeswitch.org Mon Nov 2 13:04:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Nov 2009 13:04:12 -0800 Subject: [Freeswitch-users] SIP provider with extern rtp server In-Reply-To: <03E9A3AC-F501-4FAE-8199-A4AEC4D60891@myrvold.org> References: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com> <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org> <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org> <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org> <191c3a030911011220m22d7b515kda412e1fd9408f59@mail.gmail.com> <03E9A3AC-F501-4FAE-8199-A4AEC4D60891@myrvold.org> Message-ID: <87f2f3b90911021304yb972754gc4daa504d2304f92@mail.gmail.com> On Mon, Nov 2, 2009 at 12:58 PM, Ivan C Myrvold wrote: > That was it. My sip provider applied the patch to his Asterisk server that > was referenced in the link you was so kind to provide, and again everything > worked as it should. > > Thank you very much! > > This is why Tony's Asterisk karma is still so high! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/d8b1611d/attachment.html From hjqlopez at hotmail.com Mon Nov 2 14:30:38 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Mon, 2 Nov 2009 17:30:38 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media Message-ID: Hi Mike, I re-tried with trunk rev 15319 but I got almost the same behavior: There is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But still there is no reINVITE for A (with C's SDP) after the call from FS to C is established. Anyway, we decided for now to do a different implementation but if you want to explore more in this issue count me in ;-) Thank you very much! Humberto >Please re-try with latest svn trunk. > >Mike > >On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote: > >> >> Thanks for you answers guys, >> >> I test the parameters you suggested >> but still no audio due to the lack of reINVITE. By the way I'm using >> 1.0.4 but I also tried 1.0.5pre3. >> >> One particular condition is that there is no on-hold before the >> Blind Transfer. >> >> Regards, >> >> Humberto >> >>> >>> _________________________________________________________________ Lots of fantastic Windows 7 offers, in one convenient place. Get the perfect deal for you now. http://go.microsoft.com/?linkid=9691633 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/e4cb94ad/attachment.html From anthony.minessale at gmail.com Mon Nov 2 16:01:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Nov 2009 18:01:33 -0600 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <191c3a030911021601l660c294dp16179dbc5e23206e@mail.gmail.com> please try r15326 I think i have it working. I recommend for optimal results you set bypass_media_after_bridge=true either as a global or in your DP in place of bypass_media=true On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana wrote: > Hi Mike, > > I re-tried with trunk rev 15319 but I got almost the same behavior: There > is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But > still there is no reINVITE for A (with C's SDP) after the call from FS to C > is established. > > Anyway, we decided for now to do a different implementation but if you want > to explore more in this issue count me in ;-) > > > Thank you very much! > > Humberto > > > > > >Please re-try with latest svn trunk. > > > >Mike > > > >On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote: > > > >> > >> Thanks for you answers guys, > >> > >> I test the parameters you suggested > >> but still no audio due to the lack of reINVITE. By the way I'm using > >> 1.0.4 but I also tried 1.0.5pre3. > >> > >> One particular condition is that there is no on-hold before the > >> Blind Transfer. > >> > >> Regards, > >> > >> Humberto > >> > >>> > >>> > > ------------------------------ > Lots of fantastic offers on Windows 7, in one convenient place. Get a deal > on Windows 7 now > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/ec9fffdb/attachment.html From maciej.aniserowicz at gmail.com Mon Nov 2 21:48:45 2009 From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz) Date: Mon, 2 Nov 2009 21:48:45 -0800 (PST) Subject: [Freeswitch-users] "Can not record session. Media not enabled on channel." In-Reply-To: <1256746903296-3906568.post@n2.nabble.com> References: <1256034659005-3857858.post@n2.nabble.com> <87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com> <1256283391890-3877285.post@n2.nabble.com> <87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com> <1256541677550-3890610.post@n2.nabble.com> <87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com> <1256655334215-3899478.post@n2.nabble.com> <191c3a030910270818y79f436d9h8f6311c4671502c2@mail.gmail.com> <1256746903296-3906568.post@n2.nabble.com> Message-ID: <1257227325746-3936705.post@n2.nabble.com> Hi, Unfortunately getting the newest version did not solve the problem: "Can not record session. Media not enabled on channel." error still appears sometimes. MA Maciej Aniserowicz wrote: > > Correct - compiled but did not run. Works fine now. > > I'll see if the error shows up again and let you know if it does. > Thanks, > MA > > > > Anthony Minessale wrote: >> >> won't compile or won't run? >> maybe you should try rebuilding it. >> >> >> On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz < >> maciej.aniserowicz at gmail.com> wrote: >> >>> Sorry, trunk does not compile on win7, here are the details: >>> >>> >>> rev.15247 >>> >>> --------------------------- >>> Microsoft Visual C++ Debug Library >>> --------------------------- >>> Debug Assertion Failed! >>> >>> >>> >>> >>> >>> ----- Original Message ----- >>> *From:* [hidden >>> email] >>> *To:* [hidden >>> email] >>> *Sent:* Monday, October 26, 2009 10:32 PM >>> *Subject:* Re: [Freeswitch-users] "Can not record session. Media not >>> enabled on channel." >>> >>> >>> >>> On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden >>> email] >>> > wrote: >>> >>>> >>>> Yes, I can confirm - this exact error occurs each time when I start >>>> recording >>>> before the call is answered (just after sending ORIGINATE command) - >>>> but I >>>> think that's completely understandable that media is not ready for an >>>> unanswered call. >>>> But... is there any other event that guarantees media to be ready? >>>> >>>> Update to latest SVN and try again. >>> -MC >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> [hidden >>> email] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------ >>> View this message in context: Re: [Freeswitch-users] "Can not record >>> session. Media not enabled on >>> channel." >>> >>> Sent from the freeswitch-users mailing list >>> archiveat >>> Nabble.com. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3936705.html Sent from the freeswitch-users mailing list archive at Nabble.com. From lei.tlfly at gmail.com Tue Nov 3 01:05:24 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 3 Nov 2009 17:05:24 +0800 Subject: [Freeswitch-users] How to get digitals and stop play when speak tts? Just like session:playAndGetDigits do Message-ID: <50c41b4e0911030105k5585b8f2q3b355fbb488750ed@mail.gmail.com> Hi all, I'm writing lua ivr scirpt, Does some known how to get digitals and stop play when speak tts? Just like session:playAndGetDigits do. Thanks lots! Best Regards! -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/4ef9368f/attachment.html From brian.stafford at lattice-voice.com Tue Nov 3 01:28:18 2009 From: brian.stafford at lattice-voice.com (Brian Stafford) Date: Tue, 03 Nov 2009 09:28:18 +0000 Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg of call In-Reply-To: <4AEB1166.80002@lattice-voice.com> References: <4AEB0A9C.7010907@lattice-voice.com> <4AEB1166.80002@lattice-voice.com> Message-ID: <4AEFF7B2.3080607@lattice-voice.com> Brian Stafford wrote: > Brian West wrote: > >> You have to be doing it wrong then. >> >> Can you show us your dialplan you should have two extensions one for >> the lot range and one to attended transfer someone into the lot. >> >> /b >> >> > The relevant excerpt from the dialplan is > > > > > > > > > > > > > > > > x410-419 are the slots and 420 parks a call. Parking by picking one of > 410-419 works fine and subsequently dialling them from another works > fine, I added x420 for the auto feature. > > Regards > Brian > > _ Any clues what I'm doing wrong? Is more information needed? Brian From maciej.aniserowicz at gmail.com Tue Nov 3 02:28:13 2009 From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz) Date: Tue, 3 Nov 2009 02:28:13 -0800 (PST) Subject: [Freeswitch-users] Users hanged up for unknown reason Message-ID: <1257244093831-3937601.post@n2.nabble.com> Hi, I have a strange problem. I control FS with commands sent by tcp in response to events published via tcp. I do something like: 1) call 1st user 2) call 2nd user 3) 1st and 2nd talk 4) call another user 5) 1st and another talk etc... Sometimes (quite regularly) users are hanged up (with cause NORMAL_CLEARING) even if they do not hangup manually. I pasted one such scenario in pastebin (http://pastebin.freeswitch.org/10955), it includes logs from commands sent by me and events received from FS. Could someone take a look and see what am I doing wrong? The scenario includes 3 users 1st user (Unique-ID: f076261a-4537-40f2-b46d-933141320314) is supposed to be connected all the time but gets diconnected 2nd user (Unique-ID: ebdfb398-ec82-4760-9f79-81364e0f37b6) is supposed to talk for a few seconds and get killed 3rd user (Unique-ID: d5cd839e-793c-4b3c-adda-327841672a5f) is supposed to work like 2nd user All of them are simulated by dialplan extensions (using answer and playback tools), but the same thing happends for xlite or cisco phone. Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Users-hanged-up-for-unknown-reason-tp3937601p3937601.html Sent from the freeswitch-users mailing list archive at Nabble.com. From stevendt at primrosebank.net Tue Nov 3 03:28:23 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 11:28:23 -0000 Subject: [Freeswitch-users] Precompiled Windows Binaries Message-ID: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> Hi, I have read the Docs on the Wiki (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) but am still not sure of what the different Windows install files are. Currently, the Windows Installer directory contains :- LATEST_SVN_15106 - 6 Bytes freeswitch-1.0.4.exe - 42 Megabytes freeswitch.exe - 32 Megabytes I have installed the freeswitch-1.0.4.exe file which is dated 3rd September. The freeswitch.exe file is dated 7th October and think that it contains the minor updates since 3rd September ? Could someone who knows FreeSwitch under windows help me understand the two files please ? I chickened out of running the later exe in case it did something to the running install of FreeSwitch 1.0.4, is it safe to run the newer exe with the old one already installed ? What will it actually do ? regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/8518c608/attachment.html From jlenk at frontiernet.net Tue Nov 3 06:48:34 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 3 Nov 2009 06:48:34 -0800 (PST) Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> Message-ID: <1257259714704-3938887.post@n2.nabble.com> Hi Dave, These are supported by "Carlos Talbot" . They also include Freepbx v3 Just as you said freeswitch-1.0.4.exe is the tagged release and freeswitch.exe is a newer svn snapshot. There should be no problems installing the new version allthough best to just try and see! Not sure why the newest one is from October 7th. Jeff Dave Stevenson wrote: > > Hi, > > I have read the Docs on the Wiki > (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) > but am still not sure of what the different Windows install files are. > Currently, the Windows Installer directory contains :- > > LATEST_SVN_15106 - 6 Bytes > > freeswitch-1.0.4.exe - 42 Megabytes > > freeswitch.exe - 32 Megabytes > > I have installed the freeswitch-1.0.4.exe file which is dated 3rd > September. The freeswitch.exe file is dated 7th October and think that it > contains the minor updates since 3rd September ? > > Could someone who knows FreeSwitch under windows help me understand the > two files please ? > > I chickened out of running the later exe in case it did something to the > running install of FreeSwitch 1.0.4, is it safe to run the newer exe with > the old one already installed ? > What will it actually do ? > > regards > Dave > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html Sent from the freeswitch-users mailing list archive at Nabble.com. From stevendt at primrosebank.net Tue Nov 3 07:27:28 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 15:27:28 -0000 Subject: [Freeswitch-users] Precompiled Windows Binaries References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> Message-ID: Jeff, thanks a lot for the reply. I was a little confused by the fact that the "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried that I might lose something. As you say though, think that I'll cross my fingers and try the updated release. I am running FreeSwitch on a test machine at the moment until the target hardware arrives - hopefully tomorrow, so I can afford to have a little play. You mentioned FreePBX V3. I had been fumbling around trying to work out what this is and from what I've read, it seems to provide a GUI Front End for configuring FreeSwitch ? I am guessing that while it has been installed with FreeSwitch, I then need to run the FreePBX Installer to update the FreePBX/FreeSwitch configuration on my hardware ? When I start FreeSwitch, it does not automatically load the WAMPServer. When I start WAMPServer manually, and open up localhost (127.0.0.1) in a web browser, I can see the WampServer logo and various tools such as phpinfo() and phpmyadmin. FreePBX is there under Your Projects. When I opened this up the first time, it appeared to want to install FreePBX over FreeSwitch, I tried to abort this when it was going to overwrite some FreeSwitch conf files and I thought I'd better not go on until I had a better idea what was happening. I backed out of the FreePBX install and now I can't get the FreePBX or phpmyadmin pages up again (missing files) so it looks like I'm going to have to reinstall anyway. So, for next time,am I right in thinking that I should proceed with running the FreePBX install from the WAMPServer menu ? regards Dave ----- Original Message ----- From: "Jeff Lenk" To: Sent: Tuesday, November 03, 2009 2:48 PM Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > Hi Dave, > > These are supported by "Carlos Talbot" . They also include Freepbx v3 > > Just as you said freeswitch-1.0.4.exe is the tagged release and > freeswitch.exe is a newer svn snapshot. > > There should be no problems installing the new version allthough best to > just try and see! > > Not sure why the newest one is from October 7th. > > Jeff > > > Dave Stevenson wrote: >> >> Hi, >> >> I have read the Docs on the Wiki >> (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> but am still not sure of what the different Windows install files are. >> Currently, the Windows Installer directory contains :- >> >> LATEST_SVN_15106 - 6 Bytes >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> freeswitch.exe - 32 Megabytes >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> September. The freeswitch.exe file is dated 7th October and think that it >> contains the minor updates since 3rd September ? >> >> Could someone who knows FreeSwitch under windows help me understand the >> two files please ? >> >> I chickened out of running the later exe in case it did something to the >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe with >> the old one already installed ? >> What will it actually do ? >> >> regards >> Dave >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From odermann at googlemail.com Tue Nov 3 07:57:24 2009 From: odermann at googlemail.com (Dennis) Date: Tue, 3 Nov 2009 16:57:24 +0100 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com> <5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com> <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com> <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> Message-ID: <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> hi anthony, i believe, that there is no problem with the communication between fs and the cirpack (everything works to smooth as if this could be possible). if fs sends the 484, the cirpack sends more digits to fs (if there are some), so this works as it should. the problem is, that fs ends the session/socket after a 484, so that the cirpack sends the following digits into another socket. you wrote about a "1 line patch", which might not have been implemented - at least it seems so. is there a way to get someone of the sofia devs to fix this small problem, so that fs sends the 484 without ending the session/socket and waiting for an answer of the cirpack? we would take care of the rest. kind regards, dennis 2009/10/15 Anthony Minessale : > right you can reply 484 in your dp at any time > > > then it should try again. > > The bit i can't remember is if we committed a certain 1 line patch that > makes sofia parse the next invite to the same call properly, the patch was > to the sofia lib itself so test it and see. I may need to dig up the answer > again from the sofia dev. From msc at freeswitch.org Tue Nov 3 08:38:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Nov 2009 08:38:41 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.0.5 Scheduled For November 10; 1.0.5pre5 Now Available Message-ID: <87f2f3b90911030838h505a47b7qd5333fab525fc65b@mail.gmail.com> Greetings! The latest FreeSWITCH pre-release is now available: http://www.freeswitch.org/node/215 Please update and test as soon as possible. With the community's help we should be able to hit our target of releasing version 1.0.5 next Tuesday November 10. The FreeSWITCH developers appreciate all the hard work that the community does on behalf of the project. Like most open source projects, FreeSWITCH needs the community to "give back" a little and you all certainly do that. Please keep up the great work. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/da32af15/attachment.html From frank at carmickle.com Tue Nov 3 09:01:10 2009 From: frank at carmickle.com (Frank Carmickle) Date: Tue, 3 Nov 2009 12:01:10 -0500 Subject: [Freeswitch-users] portaudio error Message-ID: <20091103170110.GK10757@base.carmickle.com> Hello Debian lenny with svn15321 freeswitch at internal> load mod_portaudio -ERR [module load file routine returned an error] 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input devicefreeswitch at internal> 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 Cannot find an input device 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_portaudio.so **Module load routine returned an error** Any help would be appreciated. --FC From anthony.minessale at gmail.com Tue Nov 3 09:19:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 11:19:51 -0600 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com> <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com> <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> Message-ID: <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> The patch was it's ability to accept subsequent invites. Your problem is that in sip each new attempt to send an invite is another call. 484 is a final response so the call with too few digits is terminated. On Tue, Nov 3, 2009 at 9:57 AM, Dennis wrote: > hi anthony, > > i believe, that there is no problem with the communication between fs > and the cirpack (everything works to smooth as if this could be > possible). if fs sends the 484, the cirpack sends more digits to fs > (if there are some), so this works as it should. the problem is, that > fs ends the session/socket after a 484, so that the cirpack sends the > following digits into another socket. > > you wrote about a "1 line patch", which might not have been > implemented - at least it seems so. > > is there a way to get someone of the sofia devs to fix this small > problem, so that fs sends the 484 without ending the session/socket > and waiting for an answer of the cirpack? we would take care of the > rest. > > kind regards, > dennis > > > 2009/10/15 Anthony Minessale : > > right you can reply 484 in your dp at any time > > > > > > then it should try again. > > > > The bit i can't remember is if we committed a certain 1 line patch that > > makes sofia parse the next invite to the same call properly, the patch > was > > to the sofia lib itself so test it and see. I may need to dig up the > answer > > again from the sofia dev. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/21a896c6/attachment-0001.html From andrew at hijacked.us Tue Nov 3 09:24:38 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 3 Nov 2009 12:24:38 -0500 Subject: [Freeswitch-users] portaudio error In-Reply-To: <20091103170110.GK10757@base.carmickle.com> References: <20091103170110.GK10757@base.carmickle.com> Message-ID: <20091103172437.GA9418@hijacked.us> On Tue, Nov 03, 2009 at 12:01:10PM -0500, Frank Carmickle wrote: > Hello > > Debian lenny with svn15321 > > freeswitch at internal> load mod_portaudio > -ERR [module load file routine returned an error] > > 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input devicefreeswitch at internal> 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 Cannot find an input device > 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_portaudio.so > **Module load routine returned an error** > Try installing the alsa development headers, it's got some stupid name on debian like libasound2-devel or something. Then re-build the portaudio module and library (a couple well placed make cleans should do it). Andrew From stevendt at primrosebank.net Tue Nov 3 09:30:08 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 17:30:08 -0000 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch Message-ID: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension & password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/306a4add/attachment.html From mariano.dellano at gmail.com Tue Nov 3 09:47:05 2009 From: mariano.dellano at gmail.com (Mariano de Llano) Date: Tue, 3 Nov 2009 14:47:05 -0300 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <24282895.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> Message-ID: Hi, I'm having a problem with a Sipura, it is sending for the G729 the tag "G729a" witch is not correct due the RFC. Media Attribute (a): rtpmap:18 G729a/8000 FS is returning (200OK) Media Attribute (a): rtpmap:96 G729/8000 I think that the problem is that FS is not matching the codec, so it returns the first dynamic payload which is 96. I think that I've seen post with a similar issue, and the solution was to change the tag before it hit the switch, so, what I've done is to change the "switch_r_sdp" (I have the rest of the parameters correct due I also use it to dynamically change the codecs order) and it's changing the SDP, but when FS sends the 200OK it is returning to the endpoint: Media Attribute (a): rtpmap:96 G729/8000 Which is exactly the same problem that I have without the transformation of the SDP. Is it correct? Do I have another solution? Thanks From anthony.minessale at gmail.com Tue Nov 3 09:55:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 11:55:21 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: References: <24251951.post@talk.nabble.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> Message-ID: <191c3a030911030955k6406e81byf96f18250eb3b835@mail.gmail.com> I think you can edit the prefs in your sipura and change it to the correct string. On Tue, Nov 3, 2009 at 11:47 AM, Mariano de Llano wrote: > Hi, > > I'm having a problem with a Sipura, it is sending for the G729 the > tag "G729a" witch is not correct due the RFC. > > Media Attribute (a): rtpmap:18 G729a/8000 > > FS is returning (200OK) > > Media Attribute (a): rtpmap:96 G729/8000 > > I think that the problem is that FS is not matching the codec, so it > returns the first dynamic payload which is 96. > > I think that I've seen post with a similar issue, and the solution was > to change the tag before it hit the switch, so, what I've done is to > change the "switch_r_sdp" (I have the rest of the parameters correct > due I also use it to dynamically change the codecs order) and it's > changing the SDP, but when FS sends the 200OK it is returning to the > endpoint: > > Media Attribute (a): rtpmap:96 G729/8000 > > Which is exactly the same problem that I have without the > transformation of the SDP. > > Is it correct? Do I have another solution? > > Thanks > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/10f301eb/attachment.html From brian at freeswitch.org Tue Nov 3 09:58:48 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 11:58:48 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> Message-ID: <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> FIx your sipura to NOT include the a in the codec its in the admin section of the UI on the ATA. /b On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote: > Hi, > > I'm having a problem with a Sipura, it is sending for the G729 the > tag "G729a" witch is not correct due the RFC. > > Media Attribute (a): rtpmap:18 G729a/8000 > > FS is returning (200OK) > > Media Attribute (a): rtpmap:96 G729/8000 > > I think that the problem is that FS is not matching the codec, so it > returns the first dynamic payload which is 96. > > I think that I've seen post with a similar issue, and the solution was > to change the tag before it hit the switch, so, what I've done is to > change the "switch_r_sdp" (I have the rest of the parameters correct > due I also use it to dynamically change the codecs order) and it's > changing the SDP, but when FS sends the 200OK it is returning to the > endpoint: > > Media Attribute (a): rtpmap:96 G729/8000 > > Which is exactly the same problem that I have without the > transformation of the SDP. > > Is it correct? Do I have another solution? > > Thanks > From mariano.dellano at gmail.com Tue Nov 3 10:11:13 2009 From: mariano.dellano at gmail.com (Mariano de Llano) Date: Tue, 3 Nov 2009 15:11:13 -0300 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> Message-ID: <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Yes, that was my first option, but there many endpoints that I'm not able to configure. Basically it's a broadband solution where I have like 1000 endpoints that are out of my provisioning. Thanks, M On 03/11/2009, at 14:58, Brian West wrote: > FIx your sipura to NOT include the a in the codec its in the admin > section of the UI on the ATA. > > /b > > On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote: > >> Hi, >> >> I'm having a problem with a Sipura, it is sending for the G729 the >> tag "G729a" witch is not correct due the RFC. >> >> Media Attribute (a): rtpmap:18 G729a/8000 >> >> FS is returning (200OK) >> >> Media Attribute (a): rtpmap:96 G729/8000 >> >> I think that the problem is that FS is not matching the codec, so it >> returns the first dynamic payload which is 96. >> >> I think that I've seen post with a similar issue, and the solution >> was >> to change the tag before it hit the switch, so, what I've done is to >> change the "switch_r_sdp" (I have the rest of the parameters correct >> due I also use it to dynamically change the codecs order) and it's >> changing the SDP, but when FS sends the 200OK it is returning to the >> endpoint: >> >> Media Attribute (a): rtpmap:96 G729/8000 >> >> Which is exactly the same problem that I have without the >> transformation of the SDP. >> >> Is it correct? Do I have another solution? >> >> Thanks >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Nov 3 10:24:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 12:24:47 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> References: <24251951.post@talk.nabble.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Message-ID: <191c3a030911031024n1bf258abldbbb24f35f107f3b@mail.gmail.com> so imagine how much money all those sipuras cost. They get all the money *and* have a bug and we are free and are supposed to break the rules for them. On Tue, Nov 3, 2009 at 12:11 PM, Mariano de Llano wrote: > Yes, that was my first option, but there many endpoints that I'm not > able to configure. Basically it's a broadband solution where I have > like 1000 endpoints that are out of my provisioning. > > Thanks, > M > > On 03/11/2009, at 14:58, Brian West wrote: > > > FIx your sipura to NOT include the a in the codec its in the admin > > section of the UI on the ATA. > > > > /b > > > > On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote: > > > >> Hi, > >> > >> I'm having a problem with a Sipura, it is sending for the G729 the > >> tag "G729a" witch is not correct due the RFC. > >> > >> Media Attribute (a): rtpmap:18 G729a/8000 > >> > >> FS is returning (200OK) > >> > >> Media Attribute (a): rtpmap:96 G729/8000 > >> > >> I think that the problem is that FS is not matching the codec, so it > >> returns the first dynamic payload which is 96. > >> > >> I think that I've seen post with a similar issue, and the solution > >> was > >> to change the tag before it hit the switch, so, what I've done is to > >> change the "switch_r_sdp" (I have the rest of the parameters correct > >> due I also use it to dynamically change the codecs order) and it's > >> changing the SDP, but when FS sends the 200OK it is returning to the > >> endpoint: > >> > >> Media Attribute (a): rtpmap:96 G729/8000 > >> > >> Which is exactly the same problem that I have without the > >> transformation of the SDP. > >> > >> Is it correct? Do I have another solution? > >> > >> Thanks > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/def75bf7/attachment-0001.html From brian at freeswitch.org Tue Nov 3 10:27:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 12:27:39 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Message-ID: Sounds like bad planning. I would send out a memo to your users and have them fix it. I have raised a bug multiple times with Cisco g729a is NOT valid. /b On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: > Yes, that was my first option, but there many endpoints that I'm not > able to configure. Basically it's a broadband solution where I have > like 1000 endpoints that are out of my provisioning. > > Thanks, > M From anthony.minessale at gmail.com Tue Nov 3 10:45:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 12:45:53 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: References: <24251951.post@talk.nabble.com> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Message-ID: <191c3a030911031045ue8f9375ve0e76953d07d595d@mail.gmail.com> I am willing to support this with the note that its incorrect and will not support it by default but update to trunk and try: this should fix it for you, SIGH On Tue, Nov 3, 2009 at 12:27 PM, Brian West wrote: > Sounds like bad planning. I would send out a memo to your users and > have them fix it. I have raised a bug multiple times with Cisco g729a > is NOT valid. > > /b > > On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: > > > Yes, that was my first option, but there many endpoints that I'm not > > able to configure. Basically it's a broadband solution where I have > > like 1000 endpoints that are out of my provisioning. > > > > Thanks, > > M > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/751c5579/attachment.html From kristian.kielhofner at gmail.com Tue Nov 3 11:11:10 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 3 Nov 2009 14:11:10 -0500 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> References: <24251951.post@talk.nabble.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Message-ID: <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> It appears that Tony has already added an option (amazing) BUT you should really be setup for central provisioning with an installed base that large... You'll eventually have issues that *NO* amount of Tony/FreeSWITCH magic can fix. On Tue, Nov 3, 2009 at 1:11 PM, Mariano de Llano wrote: > Yes, that was my first option, but there many endpoints that I'm not > able to configure. Basically it's a broadband solution where I have > like 1000 endpoints that are out of my provisioning. > > Thanks, > M > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From achaloyan at yahoo.com Tue Nov 3 11:12:52 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Tue, 3 Nov 2009 11:12:52 -0800 (PST) Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Message-ID: <32896.60654.qm@web111308.mail.gq1.yahoo.com> Actually, there were a few more misinterpretations in earlier software of Cisco Gateways, which RFC implementers had to address in RFC3551, strange ... RTP Payload Type 19 remains reserved because "some implementations" wrongly interpreted 13 decimal as 13 hexadecimal value. Another issue is G726 bit packing. Again "some implementations" used wrong bit packing and RFC3551 tried to partially resolve this conflict introducing new payload format named AAL2-G726 ... ________________________________ From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Tue, November 3, 2009 10:27:39 PM Subject: Re: [Freeswitch-users] Sipura Codec Problem Sounds like bad planning. I would send out a memo to your users and have them fix it. I have raised a bug multiple times with Cisco g729a is NOT valid. /b On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: > Yes, that was my first option, but there many endpoints that I'm not > able to configure. Basically it's a broadband solution where I have > like 1000 endpoints that are out of my provisioning. > > Thanks, > M _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/a5ffdf9d/attachment.html From anthony.minessale at gmail.com Tue Nov 3 11:21:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 13:21:57 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <32896.60654.qm@web111308.mail.gq1.yahoo.com> References: <24251951.post@talk.nabble.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <32896.60654.qm@web111308.mail.gq1.yahoo.com> Message-ID: <191c3a030911031121n714daa3dj834b77e01d13638a@mail.gmail.com> Don't forget the one where there was a typo in the one for G722 so now we are all required to emulate that typo by running a 16khz codec with 8khz timestamps and sdp params. On Tue, Nov 3, 2009 at 1:12 PM, Arsen Chaloyan wrote: > Actually, there were a few more misinterpretations in earlier software of > Cisco Gateways, which RFC implementers had to address in RFC3551, strange > ... > > RTP Payload Type 19 remains reserved because "some implementations" wrongly > interpreted 13 decimal as 13 hexadecimal value. > Another issue is G726 bit packing. Again "some implementations" used wrong > bit packing and RFC3551 tried to partially resolve this conflict introducing > new payload format named AAL2-G726 ... > > ------------------------------ > *From:* Brian West > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Tue, November 3, 2009 10:27:39 PM > *Subject:* Re: [Freeswitch-users] Sipura Codec Problem > > Sounds like bad planning. I would send out a memo to your users and > have them fix it. I have raised a bug multiple times with Cisco g729a > is NOT valid. > > /b > > On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: > > > Yes, that was my first option, but there many endpoints that I'm not > > able to configure. Basically it's a broadband solution where I have > > like 1000 endpoints that are out of my provisioning. > > > > Thanks, > > M > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/20a59753/attachment.html From brian at freeswitch.org Tue Nov 3 11:22:50 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 13:22:50 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> References: <24251951.post@talk.nabble.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> Message-ID: At some point the paint will be rubbed off the magic lamp. /b On Nov 3, 2009, at 1:11 PM, Kristian Kielhofner wrote: > It appears that Tony has already added an option (amazing) BUT you > should really be setup for central provisioning with an installed base > that large... You'll eventually have issues that *NO* amount of > Tony/FreeSWITCH magic can fix. From brian at freeswitch.org Tue Nov 3 11:24:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 13:24:06 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <32896.60654.qm@web111308.mail.gq1.yahoo.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <32896.60654.qm@web111308.mail.gq1.yahoo.com> Message-ID: <2FEB08A4-FB22-4CC4-9BBC-E3732756B7DB@freeswitch.org> Yah this one is LLLAME!!!! :P We have some dyslexic engineers. /b On Nov 3, 2009, at 1:12 PM, Arsen Chaloyan wrote: > Another issue is G726 bit packing. Again "some implementations" used > wrong bit packing and RFC3551 tried to partially resolve this > conflict introducing new payload format named AAL2-G726 ... From brian at freeswitch.org Tue Nov 3 11:25:42 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 13:25:42 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <191c3a030911031121n714daa3dj834b77e01d13638a@mail.gmail.com> References: <24251951.post@talk.nabble.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <32896.60654.qm@web111308.mail.gq1.yahoo.com> <191c3a030911031121n714daa3dj834b77e01d13638a@mail.gmail.com> Message-ID: THE BLOODY MADNESS!!! I can only stop if people start saying 'NO'. :) /b On Nov 3, 2009, at 1:21 PM, Anthony Minessale wrote: > Don't forget the one where there was a typo in the one for G722 so > now we are all required to emulate that typo by running a 16khz > codec with 8khz timestamps and sdp params. From jlenk at frontiernet.net Tue Nov 3 11:26:16 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 3 Nov 2009 11:26:16 -0800 (PST) Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> Message-ID: <1257276376879-3940700.post@n2.nabble.com> Dave, Carlos can probably be a better help here too but yes Freepbx v3 is a web gui that is under heavy development - it probably is not yet ready for production but looks very promising! you can navigate to http://127.0.0.1/freepbx-v3/index.php/installer.html and restart the installer for freepbx if you want to experiment with it. The base FreeSWITCH installer does install and work well with windows and is quite easy to learn and configure. Their is a lot to learn though :) Regards, Jeff Dave Stevenson wrote: > > Jeff, > > thanks a lot for the reply. I was a little confused by the fact that the > "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried > that I might lose something. As you say though, think that I'll cross my > fingers and try the updated release. I am running FreeSwitch on a test > machine at the moment until the target hardware arrives - hopefully > tomorrow, so I can afford to have a little play. > > You mentioned FreePBX V3. I had been fumbling around trying to work out > what > this is and from what I've read, it seems to provide a GUI Front End for > configuring FreeSwitch ? > > I am guessing that while it has been installed with FreeSwitch, I then > need > to run the FreePBX Installer to update the FreePBX/FreeSwitch > configuration > on my hardware ? > > > When I start FreeSwitch, it does not automatically load the WAMPServer. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a > web > browser, I can see the WampServer logo and various tools such as phpinfo() > and phpmyadmin. FreePBX is there under Your Projects. > > When I opened this up the first time, it appeared to want to install > FreePBX > over FreeSwitch, I tried to abort this when it was going to overwrite some > FreeSwitch conf files and I thought I'd better not go on until I had a > better idea what was happening. I backed out of the FreePBX install and > now > I can't get the FreePBX or phpmyadmin pages up again (missing files) so it > looks like I'm going to have to reinstall anyway. > > So, for next time,am I right in thinking that I should proceed with > running > the FreePBX install from the WAMPServer menu ? > > regards > Dave > > > > ----- Original Message ----- > From: "Jeff Lenk" > To: > Sent: Tuesday, November 03, 2009 2:48 PM > Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > >> >> Hi Dave, >> >> These are supported by "Carlos Talbot" . They also include Freepbx v3 >> >> Just as you said freeswitch-1.0.4.exe is the tagged release and >> freeswitch.exe is a newer svn snapshot. >> >> There should be no problems installing the new version allthough best to >> just try and see! >> >> Not sure why the newest one is from October 7th. >> >> Jeff >> >> >> Dave Stevenson wrote: >>> >>> Hi, >>> >>> I have read the Docs on the Wiki >>> (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >>> but am still not sure of what the different Windows install files are. >>> Currently, the Windows Installer directory contains :- >>> >>> LATEST_SVN_15106 - 6 Bytes >>> >>> freeswitch-1.0.4.exe - 42 Megabytes >>> >>> freeswitch.exe - 32 Megabytes >>> >>> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >>> September. The freeswitch.exe file is dated 7th October and think that >>> it >>> contains the minor updates since 3rd September ? >>> >>> Could someone who knows FreeSwitch under windows help me understand the >>> two files please ? >>> >>> I chickened out of running the later exe in case it did something to the >>> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >>> with >>> the old one already installed ? >>> What will it actually do ? >>> >>> regards >>> Dave >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3940700.html Sent from the freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Tue Nov 3 11:53:26 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 3 Nov 2009 20:53:26 +0100 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch In-Reply-To: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> Message-ID: <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote: > Help please . . . . > > Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? > > I have got FreeSwitch up and running with the SoftPhone, but can't get a > 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP > Address from DHCP and it can see the FreeSwitch server but I can't find > anything in the phone to allow the extension & password to be configured. > Can FreeSwitch send this data to the phone (and if so, which configuration > files are involved) or must the phone be configured manually before it can > talk to FreeSwitch ? > > Any help would be really appreciated as I'm pulling my hair out here ! > > Regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/d428fa04/attachment.html From stevendt at primrosebank.net Tue Nov 3 12:03:13 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 20:03:13 -0000 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> Message-ID: Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. >From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a "special" version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave ----- Original Message ----- From: Tihomir Culjaga To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 03, 2009 7:53 PM Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension & password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/5388a9bd/attachment.html From chris.chen2004 at gmail.com Tue Nov 3 12:18:24 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Tue, 3 Nov 2009 15:18:24 -0500 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch In-Reply-To: References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> Message-ID: <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> I think you are most likely on the wrong track, 3COM phones are locked to either 3COM PBX or the special Asterisk edition locked-down by 3COM. You cannot make them work with either FreeSWITCH or any other open SIP server other than 3COM IP PBX systems. I learned this over one year ago by playing with 3COm 3102 phones myself. Chris On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson wrote: > Tihomir, > > thanks for the link, but actually, I had already found/downloaded/read and > almost understood that document ! > > However, the options to log into the phone and configure the extension > number etc. do not appear on my phone. > > From reading another post on the web, I don't think that the phone has the > SIP software loaded until it is downloaded from the Server - I think that > there is a "special" version of Asterix for 3Com that does this, maybe the > same functionality does not exist in FreeSwitch ? > > Maybe I should have been clearer in the post below, but I think that this > is the root of the problem. I think that the 3Com phone is looking for the > Switch to download the SIP firmware to it and FreeSwitch does not seem to do > that. > > Given that you have pointed me in the direction of that document, are you > using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, > but please let me know how you've made it work > > regards > Dave > > > > > ----- Original Message ----- > *From:* Tihomir Culjaga > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Tuesday, November 03, 2009 7:53 PM > *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with > FreeSwitch > > you might read this before you bigin :P > > http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf > > > T. > > > On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote: > >> Help please . . . . >> >> Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? >> >> I have got FreeSwitch up and running with the SoftPhone, but can't get a >> 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP >> Address from DHCP and it can see the FreeSwitch server but I can't find >> anything in the phone to allow the extension & password to be configured. >> Can FreeSwitch send this data to the phone (and if so, which configuration >> files are involved) or must the phone be configured manually before it can >> talk to FreeSwitch ? >> >> Any help would be really appreciated as I'm pulling my hair out here ! >> >> Regards >> Dave >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/b861e63f/attachment-0001.html From anthony.minessale at gmail.com Tue Nov 3 12:23:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 14:23:05 -0600 Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg of call In-Reply-To: <4AEFF7B2.3080607@lattice-voice.com> References: <4AEB0A9C.7010907@lattice-voice.com> <4AEB1166.80002@lattice-voice.com> <4AEFF7B2.3080607@lattice-voice.com> Message-ID: <191c3a030911031223p23835d6ev4c3c3ddd98193f50@mail.gmail.com> There are 2 ways to use the auto in one is to attended transfer the call into the extension with auto in the other is to bind_meta_app a call to valet_park + auto in blind transfer to auto in only has one leg so the guy you transferred is the only one who can hear it because when you press the blind xfer key you hangup the call on your side. On Tue, Nov 3, 2009 at 3:28 AM, Brian Stafford < brian.stafford at lattice-voice.com> wrote: > Brian Stafford wrote: > > Brian West wrote: > > > >> You have to be doing it wrong then. > >> > >> Can you show us your dialplan you should have two extensions one for > >> the lot range and one to attended transfer someone into the lot. > >> > >> /b > >> > >> > > The relevant excerpt from the dialplan is > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > x410-419 are the slots and 420 parks a call. Parking by picking one of > > 410-419 works fine and subsequently dialling them from another works > > fine, I added x420 for the auto feature. > > > > Regards > > Brian > > > > _ > > Any clues what I'm doing wrong? Is more information needed? > > Brian > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/53b92400/attachment.html From stevendt at primrosebank.net Tue Nov 3 12:25:20 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 20:25:20 -0000 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com><65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> Message-ID: <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com> Chris, thanks a lot for the response. It's not the answer that I wanted, but it is what I was coming round to thinking. As much as I'm disappointed (particularly as I've just got the phone), but at least it's a definitive answer and I can avoid wasting any more time with it, so thanks again. Oh well, off to try and find some open SIP phones that will actually work for me, regards Dave ----- Original Message ----- From: Chris Chen To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 03, 2009 8:18 PM Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch I think you are most likely on the wrong track, 3COM phones are locked to either 3COM PBX or the special Asterisk edition locked-down by 3COM. You cannot make them work with either FreeSWITCH or any other open SIP server other than 3COM IP PBX systems. I learned this over one year ago by playing with 3COm 3102 phones myself. Chris On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson wrote: Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a "special" version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave ----- Original Message ----- From: Tihomir Culjaga To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 03, 2009 7:53 PM Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension & password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/55003312/attachment.html From mastermind202 at gmail.com Tue Nov 3 09:52:09 2009 From: mastermind202 at gmail.com (mm_202) Date: Tue, 3 Nov 2009 12:52:09 -0500 Subject: [Freeswitch-users] FS and Skinny (SCCP) Message-ID: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> FS doesnt support SCCP (from what I gathered, just because no one has bothered coding it). Are there other users out there has use SCCP and FS? (with some middleware in between) If enough people would find a use for it, I'd be willing to actually code it (esp if someone offered a bounty). So, would anyone besides me want/use a SCCP endpoint in FS? -- mm_202. From tculjaga at gmail.com Tue Nov 3 12:53:42 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 3 Nov 2009 21:53:42 +0100 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch In-Reply-To: References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> Message-ID: <65d96fc80911031253j3ea587b3j48a71a5e2746ee23@mail.gmail.com> well, if it is a sip phone than you should be able to input your username&password somewhere. Usually, SIP phones downloads their configuration using dhcp/tftp|http method... the FW is downloaded just once if you need to upgrade the phone... I don't have any of these phones on my desk, just found the manual on the web. anyhow, freeswitch is expecting a SIP phone to register and thats it :P ... there is no specific phone provisioning from FS side. T. On Tue, Nov 3, 2009 at 9:03 PM, Dave Stevenson wrote: > Tihomir, > > thanks for the link, but actually, I had already found/downloaded/read and > almost understood that document ! > > However, the options to log into the phone and configure the extension > number etc. do not appear on my phone. > > From reading another post on the web, I don't think that the phone has the > SIP software loaded until it is downloaded from the Server - I think that > there is a "special" version of Asterix for 3Com that does this, maybe the > same functionality does not exist in FreeSwitch ? > > Maybe I should have been clearer in the post below, but I think that this > is the root of the problem. I think that the 3Com phone is looking for the > Switch to download the SIP firmware to it and FreeSwitch does not seem to do > that. > > Given that you have pointed me in the direction of that document, are you > using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, > but please let me know how you've made it work > > regards > Dave > > > > > ----- Original Message ----- > *From:* Tihomir Culjaga > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Tuesday, November 03, 2009 7:53 PM > *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with > FreeSwitch > > you might read this before you bigin :P > > http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf > > > T. > > > On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote: > >> Help please . . . . >> >> Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? >> >> I have got FreeSwitch up and running with the SoftPhone, but can't get a >> 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP >> Address from DHCP and it can see the FreeSwitch server but I can't find >> anything in the phone to allow the extension & password to be configured. >> Can FreeSwitch send this data to the phone (and if so, which configuration >> files are involved) or must the phone be configured manually before it can >> talk to FreeSwitch ? >> >> Any help would be really appreciated as I'm pulling my hair out here ! >> >> Regards >> Dave >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/f9f31ef1/attachment-0001.html From tculjaga at gmail.com Tue Nov 3 12:55:28 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 3 Nov 2009 21:55:28 +0100 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch In-Reply-To: <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> Message-ID: <65d96fc80911031255o3e7b7790ta61f3cb29107d786@mail.gmail.com> pity,the phone looks quite nice... On Tue, Nov 3, 2009 at 9:18 PM, Chris Chen wrote: > I think you are most likely on the wrong track, 3COM phones are locked to > either 3COM PBX or the special Asterisk edition locked-down by 3COM. You > cannot make them work with either FreeSWITCH or any other open SIP server > other than 3COM IP PBX systems. > I learned this over one year ago by playing with 3COm 3102 phones myself. > > Chris > > > > On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson wrote: > >> Tihomir, >> >> thanks for the link, but actually, I had already found/downloaded/read and >> almost understood that document ! >> >> However, the options to log into the phone and configure the extension >> number etc. do not appear on my phone. >> >> From reading another post on the web, I don't think that the phone has the >> SIP software loaded until it is downloaded from the Server - I think that >> there is a "special" version of Asterix for 3Com that does this, maybe the >> same functionality does not exist in FreeSwitch ? >> >> Maybe I should have been clearer in the post below, but I think that this >> is the root of the problem. I think that the 3Com phone is looking for >> the Switch to download the SIP firmware to it and FreeSwitch does not seem >> to do that. >> >> Given that you have pointed me in the direction of that document, are you >> using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, >> but please let me know how you've made it work >> >> regards >> Dave >> >> >> >> >> ----- Original Message ----- >> *From:* Tihomir Culjaga >> *To:* freeswitch-users at lists.freeswitch.org >> *Sent:* Tuesday, November 03, 2009 7:53 PM >> *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with >> FreeSwitch >> >> you might read this before you bigin :P >> >> http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf >> >> >> T. >> >> >> On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson > > wrote: >> >>> Help please . . . . >>> >>> Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? >>> >>> I have got FreeSwitch up and running with the SoftPhone, but can't get a >>> 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP >>> Address from DHCP and it can see the FreeSwitch server but I can't find >>> anything in the phone to allow the extension & password to be configured. >>> Can FreeSwitch send this data to the phone (and if so, which configuration >>> files are involved) or must the phone be configured manually before it can >>> talk to FreeSwitch ? >>> >>> Any help would be really appreciated as I'm pulling my hair out here ! >>> >>> Regards >>> Dave >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/789a3427/attachment.html From stevendt at primrosebank.net Tue Nov 3 12:57:12 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 20:57:12 -0000 Subject: [Freeswitch-users] IP Phones with FreeSwitch References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com><65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com><507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com> Message-ID: <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> Hi again, sorry to be here again ! OK, now that I know that 3Com phones and FreeSwitch don't mix, my next question is about Cisco ! I see that the FreeSwitch Interoperability list includes Cisco phones such as the 7940 and 7960. I believe that these phones need user licenses to work with Cisco Call Manager. What I'd like to confirm is that I would not need any Cisco licenses or anything else to get a Cisco IP phone working with FreeSwitch. Again, I'd really appreciate feedback from anyone using either of these (or other) Cisco phones with FreeSwitch on whether any additional licenses or software are required to work with an "out of the box" FreeSwitch installation ? regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/33e4d45b/attachment.html From sicfslist at gmail.com Tue Nov 3 13:09:09 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Tue, 03 Nov 2009 15:09:09 -0600 Subject: [Freeswitch-users] IP Phones with FreeSwitch In-Reply-To: <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com><65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com><507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com> <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> Message-ID: <4AF09BF5.4000802@gmail.com> Any of the Cisco phones with a SIP image should work fine ... no license required. SDR Dave Stevenson wrote: > Hi again, > > sorry to be here again ! > > OK, now that I know that 3Com phones and FreeSwitch don't mix, my next > question is about Cisco ! > > I see that the FreeSwitch Interoperability list includes Cisco phones > such as the 7940 and 7960. > > I believe that these phones need user licenses to work with Cisco Call > Manager. > > What I'd like to confirm is that I would not need any Cisco licenses > or anything else to get a Cisco IP phone working with FreeSwitch. > > Again, I'd really appreciate feedback from anyone using either of > these (or other) Cisco phones with FreeSwitch on whether any > additional licenses or software are required to work with an "out of > the box" FreeSwitch installation ? > > regards > Dave > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From william.suffill at gmail.com Tue Nov 3 13:10:19 2009 From: william.suffill at gmail.com (William Suffill) Date: Tue, 3 Nov 2009 16:10:19 -0500 Subject: [Freeswitch-users] IP Phones with FreeSwitch In-Reply-To: <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com> <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> Message-ID: <6b65470d0911031310g7f487ff9rfb61368280831471@mail.gmail.com> Cisco 7960 and the like that they push on the enterprise level for call manager also can be flashed with sip based firmware. I've only used the 7960 with the sip firmware. SPA942 and the like that used to be under Linksys/Sipura before that are targeted more toward smaller businesses and run SIP out of the box without any license complications. -- W From rupa at rupa.com Tue Nov 3 13:19:56 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 3 Nov 2009 15:19:56 -0600 Subject: [Freeswitch-users] IP Phones with FreeSwitch In-Reply-To: <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com> <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> Message-ID: These phones work with FS, come by irc and you can talk to sekil about his use of them. In general, if you haven't invested in a bunch of phones, I'd recommend: Polycom 330,450,550 -- pick your price point snom - again, pick your price point These are generally well supported over the rest. On Tue, Nov 3, 2009 at 2:57 PM, Dave Stevenson wrote: > Hi again, > > sorry to be here again ! > > OK, now that I know that 3Com phones and FreeSwitch don't mix, my next > question is about Cisco ! > > I see that the FreeSwitch Interoperability list includes Cisco phones such > as the 7940 and 7960. > > I believe that these phones need user licenses to work with Cisco Call > Manager. > > What I'd like to confirm is that I would not need any Cisco licenses or > anything else to get a Cisco IP phone working with FreeSwitch. > > Again, I'd really appreciate feedback from anyone using either of these (or > other) Cisco phones with FreeSwitch on whether any additional licenses or > software are required to work with an "out of the box" FreeSwitch > installation ? > > regards > Dave > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From jerry.richards at teotech.com Tue Nov 3 13:22:11 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 3 Nov 2009 13:22:11 -0800 Subject: [Freeswitch-users] Dial Plan Question Message-ID: <0A46BCC1ED4C452CAD31DF64A734C492@greyhawk.tonecommander.com> My understanding of DialPlan/CallRouting is that it can be accomplished via static XML tags, or alternatively, via a DialPlan Application that interfaces with the dptools module. Question: If my above assumption is true, how does one select one approach over the other? What is the criteria/considerations that would govern the decision? Best Regards, Jerry From jerry.richards at teotech.com Tue Nov 3 13:25:58 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 3 Nov 2009 13:25:58 -0800 Subject: [Freeswitch-users] Error checking for PMP [general error] Message-ID: When I start Freeswitch, I see an "Error checking for PMP [general error]" as shown below. Does anyone know what could cause this? [root at TeoProxy bin]# ./freeswitch Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch -waste. auto-adjusting stack size for optimal performance.... 2009-11-02 10:12:27.17579 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-02 10:12:27.18373 [DEBUG] switch_event.c:553 Create event dispatch thread 0 2009-11-02 10:12:27.428749 [INFO] switch_nat.c:392 Scanning for NAT 2009-11-02 10:12:27.428885 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 2009-11-02 10:12:27.678480 [DEBUG] switch_nat.c:152 Checking for PMP 2/5 2009-11-02 10:12:27.679449 [DEBUG] switch_nat.c:152 Checking for PMP 3/5 2009-11-02 10:12:28.179388 [DEBUG] switch_nat.c:152 Checking for PMP 4/5 2009-11-02 10:12:29.179217 [DEBUG] switch_nat.c:152 Checking for PMP 5/5 2009-11-02 10:12:31.178879 [ERR] switch_nat.c:183 Error checking for PMP [general error] 2009-11-02 10:12:31.178902 [DEBUG] switch_nat.c:397 Checking for UPnP 2009-11-02 10:12:43.176881 [INFO] switch_nat.c:411 No PMP or UPnP NAT detected! 2009-11-02 10:12:43.210145 [INFO] switch_core_sqldb.c:538 Opening DB 2009-11-02 10:12:43.919804 [NOTICE] switch_scheduler.c:166 Starting task thread 2009-11-02 10:12:43.937881 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1257185563 2009-11-02 10:12:43.937980 [CONSOLE] switch_core.c:1449 Bringing up environment. 2009-11-02 10:12:43.937994 [CONSOLE] switch_core.c:1450 Loading Modules. 2009-11-02 10:12:43.938319 [INFO] switch_time.c:661 Timezone loaded 530 definitions 2009-11-02 10:12:43.938336 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_SOFTTIMER_MODULE] 2009-11-02 10:12:43.938351 [NOTICE] switch_loadable_module.c:228 Adding Timer 'soft' 2009-11-02 10:12:43.938413 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_PCM_MODULE] Best Regards, Jerry From stevendt at primrosebank.net Tue Nov 3 13:26:26 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 21:26:26 -0000 Subject: [Freeswitch-users] IP Phones with FreeSwitch References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com><65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com><507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com><127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com><1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> Message-ID: Rupa, thanks a lot for the pointers - I'm just about to try to pick up some phones, so the tips are timely. Actually, I have been trying the IRC thing today, but keep getting "connection refused", it's been a few years since I used IRC, but I think I have a Firewall problem that I'm working on. Hopefully, I'll be there soon, regards Dave ----- Original Message ----- From: "Rupa Schomaker" To: Sent: Tuesday, November 03, 2009 9:19 PM Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch > These phones work with FS, come by irc and you can talk to sekil about > his use of them. > > In general, if you haven't invested in a bunch of phones, I'd recommend: > > Polycom 330,450,550 -- pick your price point > snom - again, pick your price point > > These are generally well supported over the rest. > > On Tue, Nov 3, 2009 at 2:57 PM, Dave Stevenson > wrote: >> Hi again, >> >> sorry to be here again ! >> >> OK, now that I know that 3Com phones and FreeSwitch don't mix, my next >> question is about Cisco ! >> >> I see that the FreeSwitch Interoperability list includes Cisco phones >> such >> as the 7940 and 7960. >> >> I believe that these phones need user licenses to work with Cisco Call >> Manager. >> >> What I'd like to confirm is that I would not need any Cisco licenses or >> anything else to get a Cisco IP phone working with FreeSwitch. >> >> Again, I'd really appreciate feedback from anyone using either of these >> (or >> other) Cisco phones with FreeSwitch on whether any additional licenses or >> software are required to work with an "out of the box" FreeSwitch >> installation ? >> >> regards >> Dave >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From stevendt at primrosebank.net Tue Nov 3 13:28:41 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 21:28:41 -0000 Subject: [Freeswitch-users] IP Phones with FreeSwitch References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com><1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> <6b65470d0911031310g7f487ff9rfb61368280831471@mail.gmail.com> Message-ID: Thanks a lot - to both William and Shelby, that makes me more confident about trying out at least one Cisco and Rupa has just given me a few more options, so, hopefully, I won't make the 3Com mistake again ! regards Dave ----- Original Message ----- From: "William Suffill" To: Sent: Tuesday, November 03, 2009 9:10 PM Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch > Cisco 7960 and the like that they push on the enterprise level for > call manager also can be flashed with sip based firmware. I've only > used the 7960 with the sip firmware. > > > SPA942 and the like that used to be under Linksys/Sipura before that > are targeted more toward smaller businesses and run SIP out of the box > without any license complications. > > -- W > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jerry.richards at teotech.com Tue Nov 3 13:35:05 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 3 Nov 2009 13:35:05 -0800 Subject: [Freeswitch-users] WARNING On Inbound Call Question Message-ID: I have my Freeswitch server with an installed Sangoma A101D card. Most everything works okay, however, when I get an inbound call from the PSTN, I see the following warning show up in the log. Additionally, the caller (on the PSTN) does not hear ringback, and if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. Here are the two warnings: [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA Here is the log of the warning upon an inbound call: freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> 2009-11-02 09:06:01.664835 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0 Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176] 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on 1:1 from DOWN to RING 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig [START] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound channel OpenZAP/1:1/5384 2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384) State Change CS_NEW -> CS_INIT 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_INIT 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/5384) State INIT 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384) State Change CS_INIT -> CS_ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/5384) State INIT going to sleep 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/5384) State ROUTING 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384 CHANNEL ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78 OpenZAP/1:1/5384 Standard ROUTING 2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing 4253813176->5384 in context default Dialplan: OpenZAP/1:1/5384 parsing [default->unloop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->tod_example] continue=true Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example] Dialplan: OpenZAP/1:1/5384 Action set(open=true) Dialplan: OpenZAP/1:1/5384 parsing [default->SangomaPRI] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI] destination_number(5384) =~ /^9(\d+)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->global-intercept] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept] destination_number(5384) =~ /^(5380)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->group-intercept] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept] destination_number(5384) =~ /^\*8$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->intercept-ext] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext] destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->redial] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384) =~ /^870$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->global] continue=true Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: OpenZAP/1:1/5384 Absolute Condition [global] Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-2] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-2] destination_number(5384) =~ /^9001$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-1] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-1] destination_number(5384) =~ /^9000$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] destination_number(5384) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] destination_number(5384) =~ /^779$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call_return] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call_return] destination_number(5384) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->del-group] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [del-group] destination_number(5384) =~ /^80(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->add-group] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [add-group] destination_number(5384) =~ /^81(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-simo] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-simo] destination_number(5384) =~ /^82(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-order] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-order] destination_number(5384) =~ /^83(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->extension-intercom] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [extension-intercom] destination_number(5384) =~ /^8(5[34][8901][0-9])$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->Local_Extension] continue=false Dialplan: OpenZAP/1:1/5384 Regex (PASS) [Local_Extension] destination_number(5384) =~ /^(5[34][8901][0-9])$/ break=on-false Dialplan: OpenZAP/1:1/5384 Action set(dialed_extension=5384) Dialplan: OpenZAP/1:1/5384 Action export(dialed_extension=5384) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: OpenZAP/1:1/5384 Action set(ringback=${us-ring}) Dialplan: OpenZAP/1:1/5384 Action set(transfer_ringback=local_stream://moh) Dialplan: OpenZAP/1:1/5384 Action set(call_timeout=30) Dialplan: OpenZAP/1:1/5384 Action set(hangup_after_bridge=true) Dialplan: OpenZAP/1:1/5384 Action set(continue_on_fail=true) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: OpenZAP/1:1/5384 Action answer() Dialplan: OpenZAP/1:1/5384 Action sleep(1000) Dialplan: OpenZAP/1:1/5384 Action voicemail(default ${domain_name} ${dialed_extension}) 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:114 (OpenZAP/1:1/5384) State Change CS_ROUTING -> CS_EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/5384) State ROUTING going to sleep 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:491 (OpenZAP/1:1/5384) State EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] mod_openzap.c:408 OpenZAP/1:1/5384 CHANNEL EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:151 OpenZAP/1:1/5384 Standard EXECUTE EXECUTE OpenZAP/1:1/5384 set(open=true) 2009-11-02 09:06:01.666685 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [open]=[true] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-spymap/4253813176/b678f311-ab74-4cc1-afac-b83d89a 53132) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/4253813176/5384) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/global/b678f311-ab74-4cc1-afac-b83d89a5 3132) EXECUTE OpenZAP/1:1/5384 set(dialed_extension=5384) 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [dialed_extension]=[5384] EXECUTE OpenZAP/1:1/5384 export(dialed_extension=5384) 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:886 EXPORT [dialed_extension]=[5384] EXECUTE OpenZAP/1:1/5384 bind_meta_app(1 b s execute_extension::dx XML features) 2009-11-02 09:06:01.667682 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 execute_extension::dx XML features EXECUTE OpenZAP/1:1/5384 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 -01.wav) 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 -01.wav EXECUTE OpenZAP/1:1/5384 bind_meta_app(3 b s execute_extension::cf XML features) 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 execute_extension::cf XML features EXECUTE OpenZAP/1:1/5384 set(ringback=%(2000,4000,440.0,480.0)) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE OpenZAP/1:1/5384 set(transfer_ringback=local_stream://moh) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [transfer_ringback]=[local_stream://moh] EXECUTE OpenZAP/1:1/5384 set(call_timeout=30) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [call_timeout]=[30] EXECUTE OpenZAP/1:1/5384 set(hangup_after_bridge=true) 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [hangup_after_bridge]=[true] EXECUTE OpenZAP/1:1/5384 set(continue_on_fail=true) 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [continue_on_fail]=[true] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-call_return/5384/4253813176) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial_ext/5384/b678f311-ab74-4cc1-afac-b83d89 a53132) EXECUTE OpenZAP/1:1/5384 set(called_party_callgroup=techsupport) 2009-11-02 09:06:01.670679 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [called_party_callgroup]=[techsupport] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/techsupport/b678f311-ab74-4cc1-afac-b83 d89a53132) EXECUTE OpenZAP/1:1/5384 bridge(user/5384 at 192.168.72.141) 2009-11-02 09:06:01.671683 [DEBUG] switch_ivr_originate.c:1027 variable string 0 = [presence_id=5384 at 192.168.72.141] 2009-11-02 09:06:01.671683 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:5384 at 192.168.72.163:5060 [9e7b8fae-6194-430c-951b-948ebd2c2a3b] 2009-11-02 09:06:01.671683 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_NEW -> CS_INIT 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_INIT 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:83 sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA INIT 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_INIT -> CS_ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT going to sleep 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:130 sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING going to sleep 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:5384 at 192.168.72.163:5060 entering state [calling][0] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:5384 at 192.168.72.163:5060) State CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:1701 OpenZAP/1:1/5384 receive message [PROGRESS] 2009-11-02 09:06:01.673742 [DEBUG] mod_openzap.c:759 Changing state on 1:1 from RING to PROGRESS 2009-11-02 09:06:01.674787 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [PROGRESS] 2009-11-02 09:06:01.675844 [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 2009-11-02 09:06:01.684776 [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA 2009-11-02 09:06:01.684776 [DEBUG] switch_core_session.c:630 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.684776 [NOTICE] switch_ivr_originate.c:1701 Pre-Answer OpenZAP/1:1/5384! 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1777 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-02 09:06:01.693835 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:5384 at 192.168.72.163:5060 entering state [proceeding][180] 2009-11-02 09:06:01.693835 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:5384 at 192.168.72.163:5060! 2009-11-02 09:06:01.705777 [DEBUG] switch_core_io.c:649 OpenZAP/1:1/5384 receive message [TRANSCODING_NECESSARY] freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> Best Regards, Jerry From sicfslist at gmail.com Tue Nov 3 13:44:55 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Tue, 03 Nov 2009 15:44:55 -0600 Subject: [Freeswitch-users] Dial Plan Question In-Reply-To: <0A46BCC1ED4C452CAD31DF64A734C492@greyhawk.tonecommander.com> References: <0A46BCC1ED4C452CAD31DF64A734C492@greyhawk.tonecommander.com> Message-ID: <4AF0A457.5080702@gmail.com> I think the real question is what are you trying to do ... for some things it's very easy to just whip up a static XML file and be done with it. For others you probably want some sort of interaction with a DB. The options here are pretty endless: -- XML curl -- handing off the call to a script call from a static dial plan (use lua if there is going to be any load) -- event_socket -- mod_lcr But ultimately I think it's what you're trying to accomplish that matters. For a PBX install I'd say static files is probably about as easy as it is going to get. For delivering a service you'd probably want interaction with a DB. I've use XML curl a lot and have even starting using direct DB queries from static dialplans using mod_memcache and memcachedb (not memcache ... persistent storage). SDR Jerry Richards wrote: > My understanding of DialPlan/CallRouting is that it can be accomplished via > static XML tags, or alternatively, via a DialPlan Application that > interfaces with the dptools module. > > Question: If my above assumption is true, how does one select one approach > over the other? What is the criteria/considerations that would govern the > decision? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From frank at carmickle.com Tue Nov 3 14:00:22 2009 From: frank at carmickle.com (Frank Carmickle) Date: Tue, 3 Nov 2009 17:00:22 -0500 Subject: [Freeswitch-users] portaudio error In-Reply-To: <20091103172437.GA9418@hijacked.us> References: <20091103170110.GK10757@base.carmickle.com> <20091103172437.GA9418@hijacked.us> Message-ID: <20091103220022.GL10757@base.carmickle.com> On Tue, Nov 03, Andrew Thompson wrote: > On Tue, Nov 03, 2009 at 12:01:10PM -0500, Frank Carmickle wrote: > > Hello > > > > Debian lenny with svn15321 > > > > freeswitch at internal> load mod_portaudio > > -ERR [module load file routine returned an error] > > > > 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input devicefreeswitch at internal> 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 Cannot find an input device > > 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_portaudio.so > > **Module load routine returned an error** > > > Try installing the alsa development headers, it's got some stupid name > on debian like libasound2-devel or something. Then re-build the > portaudio module and library (a couple well placed make cleans should do > it). Hi Libasound2-dev is still installed. I have had PA working in the passed. I think it was as of svn 14000 or so. Thanks for the help. --FC From rupa at rupa.com Tue Nov 3 14:18:35 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 3 Nov 2009 16:18:35 -0600 Subject: [Freeswitch-users] Error checking for PMP [general error] In-Reply-To: References: Message-ID: If you don't have a router with NAT-PMP enabled then it is expected. Same if you don't have upnp. If you are behind a NAT, it is in your best interest to enable one or the other in your router. It will save you a bunch of headaches... On Tue, Nov 3, 2009 at 3:25 PM, Jerry Richards wrote: > > When I start Freeswitch, I see an "Error checking for PMP [general error]" > as shown below. ?Does anyone know what could cause this? > > > [root at TeoProxy bin]# ./freeswitch > Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch > -waste. > auto-adjusting stack size for optimal performance.... > 2009-11-02 10:12:27.17579 [INFO] switch_event.c:565 Activate Eventing > Engine. > 2009-11-02 10:12:27.18373 [DEBUG] switch_event.c:553 Create event dispatch > thread 0 > 2009-11-02 10:12:27.428749 [INFO] switch_nat.c:392 Scanning for NAT > 2009-11-02 10:12:27.428885 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 > 2009-11-02 10:12:27.678480 [DEBUG] switch_nat.c:152 Checking for PMP 2/5 > 2009-11-02 10:12:27.679449 [DEBUG] switch_nat.c:152 Checking for PMP 3/5 > 2009-11-02 10:12:28.179388 [DEBUG] switch_nat.c:152 Checking for PMP 4/5 > 2009-11-02 10:12:29.179217 [DEBUG] switch_nat.c:152 Checking for PMP 5/5 > 2009-11-02 10:12:31.178879 [ERR] switch_nat.c:183 Error checking for PMP > [general error] > 2009-11-02 10:12:31.178902 [DEBUG] switch_nat.c:397 Checking for UPnP > 2009-11-02 10:12:43.176881 [INFO] switch_nat.c:411 No PMP or UPnP NAT > detected! > 2009-11-02 10:12:43.210145 [INFO] switch_core_sqldb.c:538 Opening DB > 2009-11-02 10:12:43.919804 [NOTICE] switch_scheduler.c:166 Starting task > thread > 2009-11-02 10:12:43.937881 [DEBUG] switch_scheduler.c:214 Added task 1 > heartbeat (core) to run at 1257185563 > 2009-11-02 10:12:43.937980 [CONSOLE] switch_core.c:1449 Bringing up > environment. > 2009-11-02 10:12:43.937994 [CONSOLE] switch_core.c:1450 Loading Modules. > 2009-11-02 10:12:43.938319 [INFO] switch_time.c:661 Timezone loaded 530 > definitions > 2009-11-02 10:12:43.938336 [CONSOLE] switch_loadable_module.c:889 > Successfully Loaded [CORE_SOFTTIMER_MODULE] > 2009-11-02 10:12:43.938351 [NOTICE] switch_loadable_module.c:228 Adding > Timer 'soft' > 2009-11-02 10:12:43.938413 [CONSOLE] switch_loadable_module.c:889 > Successfully Loaded [CORE_PCM_MODULE] > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From anthony.minessale at gmail.com Tue Nov 3 14:23:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 16:23:09 -0600 Subject: [Freeswitch-users] WARNING On Inbound Call Question In-Reply-To: References: Message-ID: <191c3a030911031423y58905d44mc00a7d949573ff86@mail.gmail.com> can you try the same thing with the latest trunk or pre-release tarball. On Tue, Nov 3, 2009 at 3:35 PM, Jerry Richards wrote: > > I have my Freeswitch server with an installed Sangoma A101D card. Most > everything works okay, however, when I get an inbound call from the PSTN, I > see the following warning show up in the log. Additionally, the caller (on > the PSTN) does not hear ringback, and if the call is not answered within > about 12 seconds, the call ends (so it doesn't go to voice mail). If I > make > a call from one internal phone to another, then it will go to voice mail > after 30 seconds. > > > Here are the two warnings: > > [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] > Rc=0 CSid=0 Seq=11 > [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to > PROGRESS_MEDIA > > > Here is the log of the warning upon an inbound call: > > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> 2009-11-02 09:06:01.664835 > [WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0 > Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176] > 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on > 1:1 from DOWN to RING > 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING] > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig > [START] > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound > channel OpenZAP/1:1/5384 > 2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel > OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132] > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384) > State Change CS_NEW -> CS_INIT > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal > OpenZAP/1:1/5384 [BREAK] > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 > (OpenZAP/1:1/5384) Running State Change CS_INIT > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 > (OpenZAP/1:1/5384) State INIT > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384) > State Change CS_INIT -> CS_ROUTING > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal > OpenZAP/1:1/5384 [BREAK] > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 > (OpenZAP/1:1/5384) State INIT going to sleep > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 > (OpenZAP/1:1/5384) Running State Change CS_ROUTING > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484 > (OpenZAP/1:1/5384) State ROUTING > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384 > CHANNEL ROUTING > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78 > OpenZAP/1:1/5384 Standard ROUTING > 2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing > 4253813176->5384 in context default > Dialplan: OpenZAP/1:1/5384 parsing [default->unloop] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~ > /^true$/ break=on-false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~ > /^true$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->tod_example] continue=true > Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example] > Dialplan: OpenZAP/1:1/5384 Action set(open=true) > Dialplan: OpenZAP/1:1/5384 parsing [default->SangomaPRI] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI] > destination_number(5384) =~ /^9(\d+)$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->global-intercept] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept] > destination_number(5384) =~ /^(5380)$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->group-intercept] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept] > destination_number(5384) =~ /^\*8$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->intercept-ext] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext] > destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->redial] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384) > =~ > /^870$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->global] continue=true > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~ > /^true$/ break=never > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > Dialplan: OpenZAP/1:1/5384 Absolute Condition [global] > Dialplan: OpenZAP/1:1/5384 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: OpenZAP/1:1/5384 Action > > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe > r}) > Dialplan: OpenZAP/1:1/5384 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-2] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-2] > destination_number(5384) =~ /^9001$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-1] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-1] > destination_number(5384) =~ /^9000$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] > destination_number(5384) > =~ /^88(.*)$|^\*0(.*)$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] > destination_number(5384) > =~ /^779$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->call_return] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call_return] > destination_number(5384) =~ /^\*69$|^869$|^lcr$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->del-group] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [del-group] > destination_number(5384) > =~ /^80(\d{2})$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->add-group] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [add-group] > destination_number(5384) > =~ /^81(\d{2})$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-simo] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-simo] > destination_number(5384) =~ /^82(\d{2})$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-order] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-order] > destination_number(5384) =~ /^83(\d{2})$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->extension-intercom] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [extension-intercom] > destination_number(5384) =~ /^8(5[34][8901][0-9])$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->Local_Extension] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (PASS) [Local_Extension] > destination_number(5384) =~ /^(5[34][8901][0-9])$/ break=on-false > Dialplan: OpenZAP/1:1/5384 Action set(dialed_extension=5384) > Dialplan: OpenZAP/1:1/5384 Action export(dialed_extension=5384) > Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(1 b s execute_extension::dx > XML features) > Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(2 b s > > record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft > ime(%Y-%m-%d-%H-%M-%S)}.wav) > Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(3 b s execute_extension::cf > XML features) > Dialplan: OpenZAP/1:1/5384 Action set(ringback=${us-ring}) > Dialplan: OpenZAP/1:1/5384 Action set(transfer_ringback=local_stream://moh) > Dialplan: OpenZAP/1:1/5384 Action set(call_timeout=30) > Dialplan: OpenZAP/1:1/5384 Action set(hangup_after_bridge=true) > Dialplan: OpenZAP/1:1/5384 Action set(continue_on_fail=true) > Dialplan: OpenZAP/1:1/5384 Action > > hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe > r}) > Dialplan: OpenZAP/1:1/5384 Action > hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) > Dialplan: OpenZAP/1:1/5384 Action > set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}) > Dialplan: OpenZAP/1:1/5384 Action > hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) > Dialplan: OpenZAP/1:1/5384 Action > bridge(user/${dialed_extension}@${domain_name}) > Dialplan: OpenZAP/1:1/5384 Action answer() > Dialplan: OpenZAP/1:1/5384 Action sleep(1000) > Dialplan: OpenZAP/1:1/5384 Action voicemail(default ${domain_name} > ${dialed_extension}) > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:114 > (OpenZAP/1:1/5384) State Change CS_ROUTING -> CS_EXECUTE > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_session.c:932 Send signal > OpenZAP/1:1/5384 [BREAK] > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:484 > (OpenZAP/1:1/5384) State ROUTING going to sleep > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:398 > (OpenZAP/1:1/5384) Running State Change CS_EXECUTE > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:491 > (OpenZAP/1:1/5384) State EXECUTE > 2009-11-02 09:06:01.666685 [DEBUG] mod_openzap.c:408 OpenZAP/1:1/5384 > CHANNEL EXECUTE > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:151 > OpenZAP/1:1/5384 Standard EXECUTE > EXECUTE OpenZAP/1:1/5384 set(open=true) > 2009-11-02 09:06:01.666685 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [open]=[true] > EXECUTE OpenZAP/1:1/5384 > > hash(insert/192.168.72.141-spymap/4253813176/b678f311-ab74-4cc1-afac-b83d89a > 53132) > EXECUTE OpenZAP/1:1/5384 > hash(insert/192.168.72.141-last_dial/4253813176/5384) > EXECUTE OpenZAP/1:1/5384 > > hash(insert/192.168.72.141-last_dial/global/b678f311-ab74-4cc1-afac-b83d89a5 > 3132) > EXECUTE OpenZAP/1:1/5384 set(dialed_extension=5384) > 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [dialed_extension]=[5384] > EXECUTE OpenZAP/1:1/5384 export(dialed_extension=5384) > 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:886 EXPORT > [dialed_extension]=[5384] > EXECUTE OpenZAP/1:1/5384 bind_meta_app(1 b s execute_extension::dx XML > features) > 2009-11-02 09:06:01.667682 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 > execute_extension::dx XML features > EXECUTE OpenZAP/1:1/5384 bind_meta_app(2 b s > > record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 > -01.wav) > 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 > > record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 > -01.wav > EXECUTE OpenZAP/1:1/5384 bind_meta_app(3 b s execute_extension::cf XML > features) > 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 > execute_extension::cf XML features > EXECUTE OpenZAP/1:1/5384 set(ringback=%(2000,4000,440.0,480.0)) > 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [ringback]=[%(2000,4000,440.0,480.0)] > EXECUTE OpenZAP/1:1/5384 set(transfer_ringback=local_stream://moh) > 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [transfer_ringback]=[local_stream://moh] > EXECUTE OpenZAP/1:1/5384 set(call_timeout=30) > 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [call_timeout]=[30] > EXECUTE OpenZAP/1:1/5384 set(hangup_after_bridge=true) > 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [hangup_after_bridge]=[true] > EXECUTE OpenZAP/1:1/5384 set(continue_on_fail=true) > 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [continue_on_fail]=[true] > EXECUTE OpenZAP/1:1/5384 > hash(insert/192.168.72.141-call_return/5384/4253813176) > EXECUTE OpenZAP/1:1/5384 > > hash(insert/192.168.72.141-last_dial_ext/5384/b678f311-ab74-4cc1-afac-b83d89 > a53132) > EXECUTE OpenZAP/1:1/5384 set(called_party_callgroup=techsupport) > 2009-11-02 09:06:01.670679 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [called_party_callgroup]=[techsupport] > EXECUTE OpenZAP/1:1/5384 > > hash(insert/192.168.72.141-last_dial/techsupport/b678f311-ab74-4cc1-afac-b83 > d89a53132) > EXECUTE OpenZAP/1:1/5384 bridge(user/5384 at 192.168.72.141) > 2009-11-02 09:06:01.671683 [DEBUG] switch_ivr_originate.c:1027 variable > string 0 = [presence_id=5384 at 192.168.72.141] > 2009-11-02 09:06:01.671683 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/sip:5384 at 192.168.72.163:5060 > [9e7b8fae-6194-430c-951b-948ebd2c2a3b] > 2009-11-02 09:06:01.671683 [DEBUG] mod_sofia.c:2811 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_NEW -> > CS_INIT > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_INIT > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT > 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:83 > sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA INIT > 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:111 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_INIT -> > CS_ROUTING > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT going to sleep > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change > CS_ROUTING > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING > 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:130 > sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA ROUTING > 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:63 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING going to sleep > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change > CS_CONSUME_MEDIA > 2009-11-02 09:06:01.672688 [DEBUG] sofia.c:3289 Channel > sofia/internal/sip:5384 at 192.168.72.163:5060 entering state [calling][0] > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State CONSUME_MEDIA > 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:1701 > OpenZAP/1:1/5384 receive message [PROGRESS] > 2009-11-02 09:06:01.673742 [DEBUG] mod_openzap.c:759 Changing state on 1:1 > from RING to PROGRESS > 2009-11-02 09:06:01.674787 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE > [PROGRESS] > 2009-11-02 09:06:01.675844 [WARNING] ss7_boost_client.c:218 TX EVENT (N): > CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 > 2009-11-02 09:06:01.684776 [WARNING] mod_openzap.c:761 VETO Changing state > on 1:1 from PROGRESS to PROGRESS_MEDIA > 2009-11-02 09:06:01.684776 [DEBUG] switch_core_session.c:630 Send signal > OpenZAP/1:1/5384 [BREAK] > 2009-11-02 09:06:01.684776 [NOTICE] switch_ivr_originate.c:1701 Pre-Answer > OpenZAP/1:1/5384! > 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1718 Raw Codec > Activation Success L16 at 8000hz 1 channel 20ms > 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1777 Play > Ringback > Tone [%(2000,4000,440.0,480.0)] > 2009-11-02 09:06:01.693835 [DEBUG] sofia.c:3289 Channel > sofia/internal/sip:5384 at 192.168.72.163:5060 entering state > [proceeding][180] > 2009-11-02 09:06:01.693835 [NOTICE] sofia.c:3353 Ring-Ready > sofia/internal/sip:5384 at 192.168.72.163:5060! > 2009-11-02 09:06:01.705777 [DEBUG] switch_core_io.c:649 OpenZAP/1:1/5384 > receive message [TRANSCODING_NECESSARY] > > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/c9d835e4/attachment-0001.html From mkitchin.public at gmail.com Tue Nov 3 14:19:04 2009 From: mkitchin.public at gmail.com (mkitchin.public at gmail.com) Date: Tue, 03 Nov 2009 16:19:04 -0600 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Message-ID: <4AF0AC58.3010506@gmail.com> I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: 7940-Config.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/cea1d113/attachment-0001.txt -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: FS-Log.log Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/cea1d113/attachment-0001.pl From hjqlopez at hotmail.com Tue Nov 3 16:05:53 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Tue, 3 Nov 2009 19:05:53 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media Message-ID: Hi, I tried r15332 and set in the sofia profile: a) bypass_media_after_bridge=true only b) bypass_media_after_bridge=true, param name="media-option" value="resume-media-on-hold"/> In both cases FS is hanging up the initial call (A to FS) after accepting the REFER to C: A <- reINVITE with FS' SDP <- FS A -> 200 -> FS A <- ACK <- FS A <- BYE <- FS The call to C is not even tried. I found this line is the logs that could give some idea: 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] after sending the ACK for the reINVITE Regards, Humberto >please try r15326 >I think i have it working. > >I recommend for optimal results you set bypass_media_after_bridge=true >either as a global or in your DP in place of bypass_media=true > > >On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana wrote: > >> Hi Mike, >> >> I re-tried with trunk rev 15319 but I got almost the same behavior: There >> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But >> still there is no reINVITE for A (with C's SDP) after the call from FS to C >> is established. >> >> Anyway, we decided for now to do a different implementation but if you want >> to explore more in this issue count me in ;-) >> >> >> Thank you very much! >> >> Humberto _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://go.microsoft.com/?linkid=9691817 From msc at freeswitch.org Tue Nov 3 16:16:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Nov 2009 16:16:10 -0800 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> References: <24251951.post@talk.nabble.com> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> Message-ID: <87f2f3b90911031616t2c731372i192f514d302522e9@mail.gmail.com> On Tue, Nov 3, 2009 at 11:11 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > It appears that Tony has already added an option (amazing) BUT you > should really be setup for central provisioning with an installed base > that large... You'll eventually have issues that *NO* amount of > Tony/FreeSWITCH magic can fix. > > Kristian is correct. Listen to him because he's familiar with having lots and lots of units out in the field. The bandage Tony applied will eventually wear off. The long-term solution is to treat the malady and not the symptom. I'm certain that members of the FS community could point you toward some resources to assist with central provisioning. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/31fc9d54/attachment.html From msc at freeswitch.org Tue Nov 3 16:27:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Nov 2009 16:27:30 -0800 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <4AF0AC58.3010506@gmail.com> References: <4AF0AC58.3010506@gmail.com> Message-ID: <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public at gmail.com < mkitchin.public at gmail.com> wrote: > I'm working on an alternative to a $120,000 Cisco phone system that my > > company is looking at. I got Freeswitch installed on CentOS last week > using the Quick and Dirty instructions. That part was painless. We had a > few 7940s laying around. After some wrestling with it, I got the latest > SIP firmware installed and what I hoped was a functional config > (attached). X-Lite phones can call each other no problem. 7940s can call > X-Lite no problem. Anytime I try and call a 7940, it goes straight to > voicemail. I attached a log file that shows the activity when trying to > call a7940 from X-Lite. > X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is > nshplpbx1.unix/10.85.0.53. Everything is on the same LAN. Different > subnets, but no firewalls. > I didn't see anything that said posting attachments was frowned upon. I > apologize if it isn't appropriate. I'm guessing this is something simple > and I'm just clueless on how to diagnose the issue. > I'm not tied to using this model for good, but it is what we had laying > around. Any help would be greatly appreciated. Next step is configuring > it to talk to Verizon VOIP over a DS3. > > Thanks, > Matthew Kitchin > > Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now. Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd say first thing to do is capture the SIP traffic to see if there are any clues. A "normal temporary failure" doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org) You can also join the IRC channel #freeswitch on irc.freenode.net and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/c583af89/attachment.html From brian at freeswitch.org Tue Nov 3 16:31:46 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 18:31:46 -0600 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <338A8EA0-956E-4B49-8234-AC534244FDFE@freeswitch.org> Do you have ANY nat involved? /b On Nov 3, 2009, at 6:05 PM, Humberto Quintana wrote: > 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/ > external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] > after sending the ACK for the reINVITE From anthony.minessale at gmail.com Tue Nov 3 16:38:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 18:38:20 -0600 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <191c3a030911031638h340bc54dm83ac9c3a888e5ba0@mail.gmail.com> I don't know what you are talking about anymore. The scenario I had tested is when a call is bridged in bypass_media=true bridge and you blind transfer that call back to the dialplan as soon as it hits the routing state it will resume media. it has been confirmed to not work and confirmed to have been fixed several time and if you are still having a problem you must have something blocking some of your packets or something . You have to understand that sip is a protocol and your description is completely non-standard. Perhaps you should get a console trace and attach it to a jira. The trace probably makes more sense to me. sofia profile internal siptrace on console loglevel debug reproduce and attach the whole capture. On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > > Hi, > > I tried r15332 and set in the sofia profile: > > a) bypass_media_after_bridge=true only > b) bypass_media_after_bridge=true, param name="media-option" > value="resume-media-on-hold"/> > > > In both cases FS is hanging up the initial call (A to FS) after accepting > the REFER to C: > > A <- reINVITE with FS' SDP <- FS > A -> 200 -> FS > A <- ACK <- FS > A <- BYE <- FS > > The call to C is not even tried. > > I found this line is the logs that could give some idea: > > 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup > sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] > after sending the ACK for the reINVITE > > > Regards, > > > Humberto > > >please try r15326 > >I think i have it working. > > > >I recommend for optimal results you set bypass_media_after_bridge=true > >either as a global or in your DP in place of bypass_media=true > > > > > >On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana hotmail.com>wrote: > > > >> Hi Mike, > >> > >> I re-tried with trunk rev 15319 but I got almost the same behavior: > There > >> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. > But > >> still there is no reINVITE for A (with C's SDP) after the call from FS > to C > >> is established. > >> > >> Anyway, we decided for now to do a different implementation but if you > want > >> to explore more in this issue count me in ;-) > >> > >> > >> Thank you very much! > >> > >> Humberto > > > _________________________________________________________________ > Windows Live: Friends get your Flickr, Yelp, and Digg updates when they > e-mail you. > http://go.microsoft.com/?linkid=9691817 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/29ef0f9b/attachment.html From peder at networkoblivion.com Tue Nov 3 18:12:54 2009 From: peder at networkoblivion.com (Peder) Date: Tue, 3 Nov 2009 20:12:54 -0600 Subject: [Freeswitch-users] IP Phones with FreeSwitch In-Reply-To: References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com><1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> <6b65470d0911031310g7f487ff9rfb61368280831471@mail.gmail.com> Message-ID: <037301ca5cf4$5193d960$f4bb8c20$@com> FYI, you can't do "presence" with the Cisco phones, so you can't see if someone is on the phone. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dave Stevenson Sent: Tuesday, November 03, 2009 3:29 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch Thanks a lot - to both William and Shelby, that makes me more confident about trying out at least one Cisco and Rupa has just given me a few more options, so, hopefully, I won't make the 3Com mistake again ! regards Dave ----- Original Message ----- From: "William Suffill" To: Sent: Tuesday, November 03, 2009 9:10 PM Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch > Cisco 7960 and the like that they push on the enterprise level for > call manager also can be flashed with sip based firmware. I've only > used the 7960 with the sip firmware. > > > SPA942 and the like that used to be under Linksys/Sipura before that > are targeted more toward smaller businesses and run SIP out of the box > without any license complications. > > -- W > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ujjval at simplesignal.com Tue Nov 3 18:27:59 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Tue, 3 Nov 2009 18:27:59 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> , <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7170307767D@EXMBXCLUS01.citservers.local> Was that sarcasm or you really mean it? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, November 02, 2009 9:08 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator you know I have heard this before... It seems to ONLY be AT&T /b On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote: > Yes, I think I did. However here is what furthur testing revelas. If > I dial in from AT&T cell phone, I do not see any DTMF using Don's > IVR.xml.conf to call my conf app. But when I dial the same number > using a Verizon Cell, it works. > > When I dial a number that is provisioned to call the Conf App > directly from the public.xml dialplan...it works even with the same > AT&T cell phone... > > Strange behaviour _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gromovd at gmail.com Tue Nov 3 19:16:43 2009 From: gromovd at gmail.com (Dmitry Gromov) Date: Tue, 3 Nov 2009 22:16:43 -0500 Subject: [Freeswitch-users] Wiki typo Message-ID: Hi! Was just reading wiki here: http://wiki.freeswitch.org/wiki/Home_PBX_Example It lists sample sofia.conf.xml which has this parameter: I think it should read inbound-*bypass*-media and not inbound-*no*-media... I know, it says "outdated" but still, can be confusing. Anyone here who can edit wiki and correct? Thanks, Dmitry -- DG NJ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/e889dcd2/attachment.html From brian at freeswitch.org Tue Nov 3 19:34:24 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 21:34:24 -0600 Subject: [Freeswitch-users] Wiki typo In-Reply-To: References: Message-ID: <26432E7B-B9E7-4F4E-921F-E4B6C9AD4F6C@freeswitch.org> Yes you can login and edit the wiki yourself. Thanks, /b On Nov 3, 2009, at 9:16 PM, Dmitry Gromov wrote: > Hi! > > Was just reading wiki here: http://wiki.freeswitch.org/wiki/Home_PBX_Example > It lists sample sofia.conf.xml which has this parameter: > > > I think it should read inbound-bypass-media and not inbound-no- > media... > > I know, it says "outdated" but still, can be confusing. > > Anyone here who can edit wiki and correct? > > Thanks, > Dmitry > > -- > DG > NJ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/2f14b1f4/attachment.html From gromovd at gmail.com Tue Nov 3 19:45:22 2009 From: gromovd at gmail.com (Dmitry Gromov) Date: Tue, 3 Nov 2009 22:45:22 -0500 Subject: [Freeswitch-users] Wiki typo In-Reply-To: <26432E7B-B9E7-4F4E-921F-E4B6C9AD4F6C@freeswitch.org> References: <26432E7B-B9E7-4F4E-921F-E4B6C9AD4F6C@freeswitch.org> Message-ID: Thanks, done - page has been corrected! On Tue, Nov 3, 2009 at 22:34, Brian West wrote: > Yes you can login and edit the wiki yourself. > > > You know... I actually spent some time looking for login/create account link when I noticed this typo. No idea why I did not see it then :) -- DG NJ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/2d7dbdd5/attachment.html From mkitchin.public at gmail.com Tue Nov 3 20:10:14 2009 From: mkitchin.public at gmail.com (mkitchin.public at gmail.com) Date: Tue, 03 Nov 2009 22:10:14 -0600 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> Message-ID: <4AF0FEA6.7070308@gmail.com> Michael Collins wrote: > > > On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public at gmail.com > > wrote: > > I'm working on an alternative to a $120,000 Cisco phone system that my > > company is looking at. I got Freeswitch installed on CentOS last week > using the Quick and Dirty instructions. That part was painless. We > had a > few 7940s laying around. After some wrestling with it, I got the > latest > SIP firmware installed and what I hoped was a functional config > (attached). X-Lite phones can call each other no problem. 7940s > can call > X-Lite no problem. Anytime I try and call a 7940, it goes straight to > voicemail. I attached a log file that shows the activity when > trying to > call a7940 from X-Lite. > X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is > nshplpbx1.unix/10.85.0.53 . Everything is on > the same LAN. Different > subnets, but no firewalls. > I didn't see anything that said posting attachments was frowned > upon. I > apologize if it isn't appropriate. I'm guessing this is something > simple > and I'm just clueless on how to diagnose the issue. > I'm not tied to using this model for good, but it is what we had > laying > around. Any help would be greatly appreciated. Next step is > configuring > it to talk to Verizon VOIP over a DS3. > > Thanks, > Matthew Kitchin > > > Matthew, > Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We > think you'll find FS is as powerful as any software out there right now. > > Here's a handy wiki page that will help you get the diagnosing skills > you need: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > I'd say first thing to do is capture the SIP traffic to see if there > are any clues. A "normal temporary failure" doesn't give you a lot of > detail. :) If you're new to SIP debugging then the best thing to do is > to capture the SIP trace and put it in the pastebin. > (http://pastebin.freeswitch.org) > > You can also join the IRC channel #freeswitch on irc.freenode.net > and get some real-time help. There are some > sharp folks in there, not the least of which are the three main > FreeSWITCH developers. > > -MC Thank you. I think I did what you are looking for. I stopped FS and launched this command. TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch and captured all output to http://pastebin.freeswitch.org/10965 Does this tell you anything? I'm definitely new to SIP and phone system admin in general. I have plenty of network and Linux experience. With that in mind, someone on this mailing list emailed me directly and said SipX would be a better fit for me. Is that blasphemy for me to even mention? I went through the documentation and the provisioning aspect and web interface do look tempting to a novice. I apologize if this is like trying to buy a chevy at a ford dealership. I'm looking to deploy about 150 handsets at a corporate office and then 10 to 12 handsets at 120 remote locations. We are moving from an old key system, so our current features are very limited. We just need a few ACD groups, call history, and the other general basics. I first found Asterisk and read about some of the shortcomings. FS looks like the most robust solution. I have no idea where SipX would fit in. The people here are obviously a very knowledgeable group and I would gladly accept any thoughts, comments, etc. From msc at freeswitch.org Tue Nov 3 20:39:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Nov 2009 20:39:14 -0800 Subject: [Freeswitch-users] Wiki typo In-Reply-To: References: <26432E7B-B9E7-4F4E-921F-E4B6C9AD4F6C@freeswitch.org> Message-ID: <87f2f3b90911032039j3a21c1f6p9bc068fa5391766d@mail.gmail.com> On Tue, Nov 3, 2009 at 7:45 PM, Dmitry Gromov wrote: > Thanks, done - page has been corrected! > > On Tue, Nov 3, 2009 at 22:34, Brian West wrote: > >> Yes you can login and edit the wiki yourself. >> >> >> > You know... I actually spent some time looking for login/create account > link when I noticed this typo. No idea why I did not see it then :) > > > Thank you for not giving up! :) We appreciate it when the community helps out. Nicely done. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/4478c6ef/attachment.html From peter at cindyandpeter.com Tue Nov 3 21:37:59 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Wed, 4 Nov 2009 00:37:59 -0500 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <4AF0FEA6.7070308@gmail.com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> Message-ID: <025101ca5d10$f81228c0$e8367a40$@com> Matthew, I'm about in the same boat as you are, just on a smaller scale. We have a ton of Nortel telephony gear, but it's time to move out of the 90's and enter this millennium. My Cisco quote was in the same ballpark as yours. The Cisco stuff is mature, rock solid, meshes very well with their network gear and is actually relatively easy to set up and maintain if you know your way around IOS. I just refuse to pay that kind of money for yet another semi-proprietary solution. After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The developers and the community are great and available, but just starting out with SIP and voip in general, this may not be the best platform. So let the blasphemy begin :) SipX was a breeze to install (insert CD, boot, next next next...) and looks pretty solid. I believe they actually use FreeSWITCH for their voicemail and conferencing, internally. I just couldn't get my head around their GUI, ACD was too basic and had all kinds of issues getting stuff to "just work". 3CX (Windows Only) was completely painless. It just worked. But I'm still not convinced that I want to run all my voice on a single windows box. Plus it's not free/open/etc and I don't want to lock myself in again. Although it's an asterisk based solution, I found trixbox to be very easy. Setup is automatic and everything "just worked". The GUI is simple and logical enough that I can let somebody else handle the day-to-day phone setup and basic admin. I have my doubts about it scaling to 250 users, though. This may be a completely flawed strategy and I may very well be shooting myself in the foot by doing this, but I plan on piloting a trixbox install with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH box next to it for the more advanced stuff. Once I get more comfortable with the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH, I have a feeling that that trixbox is going to get phased out... Peter -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mkitchin.public at gmail.com Sent: Tuesday, November 03, 2009 11:10 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Michael Collins wrote: > > > On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public at gmail.com > > wrote: > > I'm working on an alternative to a $120,000 Cisco phone system that my > > company is looking at. I got Freeswitch installed on CentOS last week > using the Quick and Dirty instructions. That part was painless. We > had a > few 7940s laying around. After some wrestling with it, I got the > latest > SIP firmware installed and what I hoped was a functional config > (attached). X-Lite phones can call each other no problem. 7940s > can call > X-Lite no problem. Anytime I try and call a 7940, it goes straight to > voicemail. I attached a log file that shows the activity when > trying to > call a7940 from X-Lite. > X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is > nshplpbx1.unix/10.85.0.53 . Everything is on > the same LAN. Different > subnets, but no firewalls. > I didn't see anything that said posting attachments was frowned > upon. I > apologize if it isn't appropriate. I'm guessing this is something > simple > and I'm just clueless on how to diagnose the issue. > I'm not tied to using this model for good, but it is what we had > laying > around. Any help would be greatly appreciated. Next step is > configuring > it to talk to Verizon VOIP over a DS3. > > Thanks, > Matthew Kitchin > > > Matthew, > Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We > think you'll find FS is as powerful as any software out there right now. > > Here's a handy wiki page that will help you get the diagnosing skills > you need: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > I'd say first thing to do is capture the SIP traffic to see if there > are any clues. A "normal temporary failure" doesn't give you a lot of > detail. :) If you're new to SIP debugging then the best thing to do is > to capture the SIP trace and put it in the pastebin. > (http://pastebin.freeswitch.org) > > You can also join the IRC channel #freeswitch on irc.freenode.net > and get some real-time help. There are some > sharp folks in there, not the least of which are the three main > FreeSWITCH developers. > > -MC Thank you. I think I did what you are looking for. I stopped FS and launched this command. TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch and captured all output to http://pastebin.freeswitch.org/10965 Does this tell you anything? I'm definitely new to SIP and phone system admin in general. I have plenty of network and Linux experience. With that in mind, someone on this mailing list emailed me directly and said SipX would be a better fit for me. Is that blasphemy for me to even mention? I went through the documentation and the provisioning aspect and web interface do look tempting to a novice. I apologize if this is like trying to buy a chevy at a ford dealership. I'm looking to deploy about 150 handsets at a corporate office and then 10 to 12 handsets at 120 remote locations. We are moving from an old key system, so our current features are very limited. We just need a few ACD groups, call history, and the other general basics. I first found Asterisk and read about some of the shortcomings. FS looks like the most robust solution. I have no idea where SipX would fit in. The people here are obviously a very knowledgeable group and I would gladly accept any thoughts, comments, etc. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jason at jasonjgw.net Tue Nov 3 22:42:01 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 4 Nov 2009 17:42:01 +1100 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <025101ca5d10$f81228c0$e8367a40$@com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> <025101ca5d10$f81228c0$e8367a40$@com> Message-ID: <20091104064201.GA15804@jdc.jasonjgw.net> Peter J. Zandvoort wrote: > After looking at various asterisk distributions, SipX, 3CX and > what-have-you, I've come to the conclusion that FreeSWITCH is by far the > most advanced platform out there. Its architecture and performance is > literally light years ahead of the rest and I have yet to come up with > something that it can't do. But all that comes at a price: The learning > curve is like scaling a brick wall. The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki. From tculjaga at gmail.com Tue Nov 3 23:15:58 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 4 Nov 2009 08:15:58 +0100 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <87f2f3b90911031616t2c731372i192f514d302522e9@mail.gmail.com> References: <24251951.post@talk.nabble.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> <87f2f3b90911031616t2c731372i192f514d302522e9@mail.gmail.com> Message-ID: <65d96fc80911032315n4e5c5474kf68a60964a73320d@mail.gmail.com> just an off-topic question but it concenns mass provissioning ... does anyone know if there is an open TR069 platform we can work on? T. On Wed, Nov 4, 2009 at 1:16 AM, Michael Collins wrote: > > > On Tue, Nov 3, 2009 at 11:11 AM, Kristian Kielhofner < > kristian.kielhofner at gmail.com> wrote: > >> It appears that Tony has already added an option (amazing) BUT you >> should really be setup for central provisioning with an installed base >> that large... You'll eventually have issues that *NO* amount of >> Tony/FreeSWITCH magic can fix. >> >> Kristian is correct. Listen to him because he's familiar with having lots > and lots of units out in the field. The bandage Tony applied will eventually > wear off. The long-term solution is to treat the malady and not the symptom. > I'm certain that members of the FS community could point you toward some > resources to assist with central provisioning. > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/e0d0270e/attachment.html From odermann at googlemail.com Wed Nov 4 00:23:01 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 4 Nov 2009 09:23:01 +0100 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com> <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> Message-ID: <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive? so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits. 2009/11/3 Anthony Minessale : > The patch was it's ability to accept subsequent invites. > Your problem is that in sip each new attempt to send an invite is another > call. > > 484 is a final response so the call with too few digits is terminated. From brian.stafford at lattice-voice.com Wed Nov 4 01:44:43 2009 From: brian.stafford at lattice-voice.com (Brian Stafford) Date: Wed, 04 Nov 2009 09:44:43 +0000 Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg of call In-Reply-To: <191c3a030911031223p23835d6ev4c3c3ddd98193f50@mail.gmail.com> References: <4AEB0A9C.7010907@lattice-voice.com> <4AEB1166.80002@lattice-voice.com> <4AEFF7B2.3080607@lattice-voice.com> <191c3a030911031223p23835d6ev4c3c3ddd98193f50@mail.gmail.com> Message-ID: <4AF14D0B.2060704@lattice-voice.com> Anthony Minessale wrote: > There are 2 ways to use the auto in > > one is to attended transfer the call into the extension with auto in > the other is to bind_meta_app a call to valet_park + auto in > > blind transfer to auto in only has one leg so the guy you transferred > is the only one who can hear it because when you press the blind xfer > key you hangup the call on your side. The penny drops - pretty obvious in hindsight. I've set it up with bind_meta_app and it's working very nicely now. Many thanks. Brian From sicfslist at gmail.com Wed Nov 4 04:08:06 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 04 Nov 2009 06:08:06 -0600 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <025101ca5d10$f81228c0$e8367a40$@com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> <025101ca5d10$f81228c0$e8367a40$@com> Message-ID: <4AF16EA6.7040708@gmail.com> Peter, Did you look at ? Probably just what you are looking for. GUI goodness based on FS. SDR Peter J. Zandvoort wrote: > Matthew, > > I'm about in the same boat as you are, just on a smaller scale. We have a > ton of Nortel telephony gear, but it's time to move out of the 90's and > enter this millennium. My Cisco quote was in the same ballpark as yours. > > The Cisco stuff is mature, rock solid, meshes very well with their network > gear and is actually relatively easy to set up and maintain if you know your > way around IOS. I just refuse to pay that kind of money for yet another > semi-proprietary solution. > > After looking at various asterisk distributions, SipX, 3CX and > what-have-you, I've come to the conclusion that FreeSWITCH is by far the > most advanced platform out there. Its architecture and performance is > literally light years ahead of the rest and I have yet to come up with > something that it can't do. But all that comes at a price: The learning > curve is like scaling a brick wall. The developers and the community are > great and available, but just starting out with SIP and voip in general, > this may not be the best platform. So let the blasphemy begin :) > > SipX was a breeze to install (insert CD, boot, next next next...) and looks > pretty solid. I believe they actually use FreeSWITCH for their voicemail and > conferencing, internally. I just couldn't get my head around their GUI, ACD > was too basic and had all kinds of issues getting stuff to "just work". > > 3CX (Windows Only) was completely painless. It just worked. But I'm still > not convinced that I want to run all my voice on a single windows box. Plus > it's not free/open/etc and I don't want to lock myself in again. > > Although it's an asterisk based solution, I found trixbox to be very easy. > Setup is automatic and everything "just worked". The GUI is simple and > logical enough that I can let somebody else handle the day-to-day phone > setup and basic admin. I have my doubts about it scaling to 250 users, > though. > > This may be a completely flawed strategy and I may very well be shooting > myself in the foot by doing this, but I plan on piloting a trixbox install > with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH > box next to it for the more advanced stuff. Once I get more comfortable with > the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH, > I have a feeling that that trixbox is going to get phased out... > > Peter > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > mkitchin.public at gmail.com > Sent: Tuesday, November 03, 2009 11:10 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones > > Michael Collins wrote: > >> On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public at gmail.com >> > > wrote: >> >> I'm working on an alternative to a $120,000 Cisco phone system that my >> >> company is looking at. I got Freeswitch installed on CentOS last week >> using the Quick and Dirty instructions. That part was painless. We >> had a >> few 7940s laying around. After some wrestling with it, I got the >> latest >> SIP firmware installed and what I hoped was a functional config >> (attached). X-Lite phones can call each other no problem. 7940s >> can call >> X-Lite no problem. Anytime I try and call a 7940, it goes straight to >> voicemail. I attached a log file that shows the activity when >> trying to >> call a7940 from X-Lite. >> X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is >> nshplpbx1.unix/10.85.0.53 . Everything is on >> the same LAN. Different >> subnets, but no firewalls. >> I didn't see anything that said posting attachments was frowned >> upon. I >> apologize if it isn't appropriate. I'm guessing this is something >> simple >> and I'm just clueless on how to diagnose the issue. >> I'm not tied to using this model for good, but it is what we had >> laying >> around. Any help would be greatly appreciated. Next step is >> configuring >> it to talk to Verizon VOIP over a DS3. >> >> Thanks, >> Matthew Kitchin >> >> >> Matthew, >> Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We >> think you'll find FS is as powerful as any software out there right now. >> >> Here's a handy wiki page that will help you get the diagnosing skills >> you need: >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> I'd say first thing to do is capture the SIP traffic to see if there >> are any clues. A "normal temporary failure" doesn't give you a lot of >> detail. :) If you're new to SIP debugging then the best thing to do is >> to capture the SIP trace and put it in the pastebin. >> (http://pastebin.freeswitch.org) >> >> You can also join the IRC channel #freeswitch on irc.freenode.net >> and get some real-time help. There are some >> sharp folks in there, not the least of which are the three main >> FreeSWITCH developers. >> >> -MC >> > Thank you. I think I did what you are looking for. I stopped FS and > launched this command. > TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch > and captured all output to http://pastebin.freeswitch.org/10965 > Does this tell you anything? > I'm definitely new to SIP and phone system admin in general. I have > plenty of network and Linux experience. With that in mind, someone on > this mailing list emailed me directly and said SipX would be a better > fit for me. Is that blasphemy for me to even mention? I went through the > documentation and the provisioning aspect and web interface do look > tempting to a novice. I apologize if this is like trying to buy a chevy > at a ford dealership. I'm looking to deploy about 150 handsets at a > corporate office and then 10 to 12 handsets at 120 remote locations. We > are moving from an old key system, so our current features are very > limited. We just need a few ACD groups, call history, and the other > general basics. I first found Asterisk and read about some of the > shortcomings. FS looks like the most robust solution. I have no idea > where SipX would fit in. The people here are obviously a very > knowledgeable group and I would gladly accept any thoughts, comments, etc. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Wed Nov 4 06:03:25 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Nov 2009 08:03:25 -0600 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com> <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> Message-ID: <9948F343-9230-42B8-AB83-C6DE0DA9886D@freeswitch.org> I'm going to say No! /b On Nov 4, 2009, at 2:23 AM, Dennis wrote: > is there a way to send something like 484 (or something else), which > does not make it a final answer and keep the call/socket alive? > > so we can ask the cirpack for further digits and decide what to do, if > the cirpack does not send any digits. From roy at net-vantage.com Tue Nov 3 21:41:05 2009 From: roy at net-vantage.com (RA Cohen) Date: Wed, 04 Nov 2009 00:41:05 -0500 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Message-ID: <4AF113F1.3090300@net-vantage.com> Here's what's in switch.conf.xml: Yet this message: SIP/2.0 503 Maximum Calls In Progress This is a small medical practice, 5-6 extensions, 3000 outbound minutes per month and at least the same inbound. We did fsctl shutdown restart and it flushed the sessions. What is going on? Thank you for your help! -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations" From carlos.talbot at gmail.com Wed Nov 4 06:51:37 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 4 Nov 2009 10:51:37 -0400 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> Message-ID: <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson wrote: > Jeff, > > thanks a lot for the reply. I was a little confused by the fact that the > "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried > that I might lose something. As you say though, think that I'll cross my > fingers and try the updated release. I am running FreeSwitch on a test > machine at the moment until the target hardware arrives - hopefully > tomorrow, so I can afford to have a little play. > I usually try to update the svn file at least once a month. I have a new version ready that was compiled last night but am ironing out login issues with the FS dudes for upload access. Also, the SVN snapshot now includes binaries for 32 and 64 bit. It no longer includes flite though as the install file was approaching 80MB in size. I will revisit this later if others feel it important to include flite. > > You mentioned FreePBX V3. I had been fumbling around trying to work out > what > this is and from what I've read, it seems to provide a GUI Front End for > configuring FreeSwitch ? > Yes, it's still in development phase and as such not ready for production use. > > I am guessing that while it has been installed with FreeSwitch, I then need > to run the FreePBX Installer to update the FreePBX/FreeSwitch configuration > on my hardware ? > > > When I start FreeSwitch, it does not automatically load the WAMPServer. > > Freeswitch and WAMPServer are independant of each other. WAMPServer is bundled in this install for the purpose of FreePBX as MySQL, Apache and PHP are all required components of FreePBX. When I start WAMPServer manually, and open up localhost (127.0.0.1) in a web > browser, I can see the WampServer logo and various tools such as phpinfo() > and phpmyadmin. FreePBX is there under Your Projects. > > If you want to configure FreePBX you need to click on the FreePBX.url shortcut that gets created on your desktop. > When I opened this up the first time, it appeared to want to install > FreePBX > over FreeSwitch, I tried to abort this when it was going to overwrite some > FreeSwitch conf files and I thought I'd better not go on until I had a > better idea what was happening. I backed out of the FreePBX install and now > I can't get the FreePBX or phpmyadmin pages up again (missing files) so it > looks like I'm going to have to reinstall anyway. > > So, for next time,am I right in thinking that I should proceed with running > the FreePBX install from the WAMPServer menu ? > No, launch it from the shortcut as stated above. Unfortunately, at this time there is very little user documentation on configuring FreePBX. Here is the link to the developer's info: http://www.freepbx.org/v3 regards, Carlos > > > ----- Original Message ----- > From: "Jeff Lenk" > To: > Sent: Tuesday, November 03, 2009 2:48 PM > Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > > > > > Hi Dave, > > > > These are supported by "Carlos Talbot" . They also include Freepbx v3 > > > > Just as you said freeswitch-1.0.4.exe is the tagged release and > > freeswitch.exe is a newer svn snapshot. > > > > There should be no problems installing the new version allthough best to > > just try and see! > > > > Not sure why the newest one is from October 7th. > > > > Jeff > > > > > > Dave Stevenson wrote: > >> > >> Hi, > >> > >> I have read the Docs on the Wiki > >> ( > http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) > >> but am still not sure of what the different Windows install files are. > >> Currently, the Windows Installer directory contains :- > >> > >> LATEST_SVN_15106 - 6 Bytes > >> > >> freeswitch-1.0.4.exe - 42 Megabytes > >> > >> freeswitch.exe - 32 Megabytes > >> > >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd > >> September. The freeswitch.exe file is dated 7th October and think that > it > >> contains the minor updates since 3rd September ? > >> > >> Could someone who knows FreeSwitch under windows help me understand the > >> two files please ? > >> > >> I chickened out of running the later exe in case it did something to the > >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe > with > >> the old one already installed ? > >> What will it actually do ? > >> > >> regards > >> Dave > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > -- > > View this message in context: > > http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/52173352/attachment-0001.html From diego.viola at gmail.com Wed Nov 4 07:25:55 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 4 Nov 2009 15:25:55 +0000 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress In-Reply-To: <4AF113F1.3090300@net-vantage.com> References: <4AF113F1.3090300@net-vantage.com> Message-ID: <86a32abc0911040725t621c287ax53b992fbcefd8691@mail.gmail.com> Hello, I tried to help Roy with this issue yesterday, I saw that calls couldn't go through and then I made a sofia profile internal siptrace on. Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and saw he had like 800 sessions. I thought it was an ACL issue but it wasn't, it seems like he reached a session limit, when I restarted his FS the problem went away. Best Regards, Diego On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen wrote: > Here's what's in switch.conf.xml: > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > per month and at least the same inbound. We did fsctl shutdown restart > and it flushed the sessions. What is going on? > > Thank you for your help! > > -- > Roy A Cohen > Network Advantage LLC > www.net-vantage.com > 413.223.9007 option 1 > -------------------------------------------------- > "Bringing Cost-Saving, State-of-the-Art Technology > Solutions to Small and Mid-Size Organizations" > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/a737b2ef/attachment.html From anthony.minessale at gmail.com Wed Nov 4 07:39:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Nov 2009 09:39:21 -0600 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> Message-ID: <191c3a030911040739k5e24a2ferdb56a3419196b581@mail.gmail.com> You cannot. This is how the sip spec works. Every new invite is a new call and a new trip to the dialplan. You will probably need to design your code to send the appropriate 484 and be prepared to exit and be called again with the new digits. On Wed, Nov 4, 2009 at 2:23 AM, Dennis wrote: > is there a way to send something like 484 (or something else), which > does not make it a final answer and keep the call/socket alive? > > so we can ask the cirpack for further digits and decide what to do, if > the cirpack does not send any digits. > > > > 2009/11/3 Anthony Minessale : > > The patch was it's ability to accept subsequent invites. > > Your problem is that in sip each new attempt to send an invite is another > > call. > > > > 484 is a final response so the call with too few digits is terminated. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/3c245f8f/attachment.html From mike at jerris.com Wed Nov 4 07:43:10 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 4 Nov 2009 10:43:10 -0500 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress In-Reply-To: <86a32abc0911040725t621c287ax53b992fbcefd8691@mail.gmail.com> References: <4AF113F1.3090300@net-vantage.com> <86a32abc0911040725t621c287ax53b992fbcefd8691@mail.gmail.com> Message-ID: Call loop? On Nov 4, 2009, at 10:25 AM, Diego Viola wrote: > Hello, > > I tried to help Roy with this issue yesterday, I saw that calls > couldn't go through and then I made a sofia profile internal > siptrace on. > > Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" > and saw he had like 800 sessions. > > I thought it was an ACL issue but it wasn't, it seems like he > reached a session limit, when I restarted his FS the problem went > away. > > Best Regards, > > Diego From jlenk at frontiernet.net Wed Nov 4 07:43:54 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Wed, 4 Nov 2009 07:43:54 -0800 (PST) Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> Message-ID: <1257349434463-3946039.post@n2.nabble.com> Hi Carlos, very cool that the x64 version is included now! Hopefully this will get more people using the x64 version under Windows! Regards, Jeff Carlos Talbot wrote: > > On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson > wrote: > >> Jeff, >> >> thanks a lot for the reply. I was a little confused by the fact that the >> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >> that I might lose something. As you say though, think that I'll cross my >> fingers and try the updated release. I am running FreeSwitch on a test >> machine at the moment until the target hardware arrives - hopefully >> tomorrow, so I can afford to have a little play. >> > > I usually try to update the svn file at least once a month. I have a new > version ready that was compiled last night but am ironing out login issues > with the FS dudes for upload access. Also, the SVN snapshot now includes > binaries for 32 and 64 bit. It no longer includes flite though as the > install file was approaching 80MB in size. I will revisit this later if > others feel it important to include flite. > >> >> You mentioned FreePBX V3. I had been fumbling around trying to work out >> what >> this is and from what I've read, it seems to provide a GUI Front End for >> configuring FreeSwitch ? >> > Yes, it's still in development phase and as such not ready for production > use. > >> >> I am guessing that while it has been installed with FreeSwitch, I then >> need >> to run the FreePBX Installer to update the FreePBX/FreeSwitch >> configuration >> on my hardware ? >> >> >> When I start FreeSwitch, it does not automatically load the WAMPServer. >> >> Freeswitch and WAMPServer are independant of each other. WAMPServer is > bundled in this install for the purpose of FreePBX as MySQL, Apache and > PHP > are all required components of FreePBX. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a > web >> browser, I can see the WampServer logo and various tools such as >> phpinfo() >> and phpmyadmin. FreePBX is there under Your Projects. >> >> If you want to configure FreePBX you need to click on the FreePBX.url > shortcut that gets created on your desktop. > > >> When I opened this up the first time, it appeared to want to install >> FreePBX >> over FreeSwitch, I tried to abort this when it was going to overwrite >> some >> FreeSwitch conf files and I thought I'd better not go on until I had a >> better idea what was happening. I backed out of the FreePBX install and >> now >> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >> it >> looks like I'm going to have to reinstall anyway. >> >> So, for next time,am I right in thinking that I should proceed with >> running >> the FreePBX install from the WAMPServer menu ? >> > > No, launch it from the shortcut as stated above. Unfortunately, at this > time > there is very little user documentation on configuring FreePBX. Here is > the > link to the developer's info: http://www.freepbx.org/v3 > > regards, > > Carlos > >> >> >> ----- Original Message ----- >> From: "Jeff Lenk" >> To: >> Sent: Tuesday, November 03, 2009 2:48 PM >> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >> >> >> > >> > Hi Dave, >> > >> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >> > >> > Just as you said freeswitch-1.0.4.exe is the tagged release and >> > freeswitch.exe is a newer svn snapshot. >> > >> > There should be no problems installing the new version allthough best >> to >> > just try and see! >> > >> > Not sure why the newest one is from October 7th. >> > >> > Jeff >> > >> > >> > Dave Stevenson wrote: >> >> >> >> Hi, >> >> >> >> I have read the Docs on the Wiki >> >> ( >> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> >> but am still not sure of what the different Windows install files are. >> >> Currently, the Windows Installer directory contains :- >> >> >> >> LATEST_SVN_15106 - 6 Bytes >> >> >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> >> >> freeswitch.exe - 32 Megabytes >> >> >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> >> September. The freeswitch.exe file is dated 7th October and think that >> it >> >> contains the minor updates since 3rd September ? >> >> >> >> Could someone who knows FreeSwitch under windows help me understand >> the >> >> two files please ? >> >> >> >> I chickened out of running the later exe in case it did something to >> the >> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >> with >> >> the old one already installed ? >> >> What will it actually do ? >> >> >> >> regards >> >> Dave >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > -- >> > View this message in context: >> > >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html Sent from the freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Wed Nov 4 07:44:08 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 4 Nov 2009 16:44:08 +0100 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <9948F343-9230-42B8-AB83-C6DE0DA9886D@freeswitch.org> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> <9948F343-9230-42B8-AB83-C6DE0DA9886D@freeswitch.org> Message-ID: <65d96fc80911040744v3bc8c604w557ebbf68ebaddf4@mail.gmail.com> Brian is right, pls, lets stop with exceptions and get stick to RFCs... otherwise it will be a big mess ... T. On Wed, Nov 4, 2009 at 3:03 PM, Brian West wrote: > I'm going to say No! > > /b > > On Nov 4, 2009, at 2:23 AM, Dennis wrote: > > > is there a way to send something like 484 (or something else), which > > does not make it a final answer and keep the call/socket alive? > > > > so we can ask the cirpack for further digits and decide what to do, if > > the cirpack does not send any digits. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/6109c3c0/attachment.html From anthony.minessale at gmail.com Wed Nov 4 07:46:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Nov 2009 09:46:11 -0600 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress In-Reply-To: <4AF113F1.3090300@net-vantage.com> References: <4AF113F1.3090300@net-vantage.com> Message-ID: <191c3a030911040746g516f8579u5f641c30c7a54b8a@mail.gmail.com> Which revision of FreeSWITCH are you using? On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen wrote: > Here's what's in switch.conf.xml: > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > per month and at least the same inbound. We did fsctl shutdown restart > and it flushed the sessions. What is going on? > > Thank you for your help! > > -- > Roy A Cohen > Network Advantage LLC > www.net-vantage.com > 413.223.9007 option 1 > -------------------------------------------------- > "Bringing Cost-Saving, State-of-the-Art Technology > Solutions to Small and Mid-Size Organizations" > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/dfb1a0e7/attachment-0001.html From roy at net-vantage.com Wed Nov 4 08:00:37 2009 From: roy at net-vantage.com (RA Cohen) Date: Wed, 04 Nov 2009 11:00:37 -0500 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Message-ID: <4AF1A525.5090909@net-vantage.com> FreeSWITCH Version 1.0.trunk (15321) -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations" From peter at cindyandpeter.com Wed Nov 4 08:33:57 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Wed, 4 Nov 2009 11:33:57 -0500 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <20091104064201.GA15804@jdc.jasonjgw.net> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> <025101ca5d10$f81228c0$e8367a40$@com> <20091104064201.GA15804@jdc.jasonjgw.net> Message-ID: <027f01ca5d6c$9c1603a0$d4420ae0$@com> Absolutely agreed. To use Matthew's original car metaphor: When you just got your learner's permit, the old Chevy may be a better choice than the Ferrari. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, November 04, 2009 1:42 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Peter J. Zandvoort wrote: > After looking at various asterisk distributions, SipX, 3CX and > what-have-you, I've come to the conclusion that FreeSWITCH is by far the > most advanced platform out there. Its architecture and performance is > literally light years ahead of the rest and I have yet to come up with > something that it can't do. But all that comes at a price: The learning > curve is like scaling a brick wall. The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jbarou at sqli.com Wed Nov 4 08:56:29 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Wed, 4 Nov 2009 17:56:29 +0100 Subject: [Freeswitch-users] Question about jingle_profiles Message-ID: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> Hi everybody, I actually working on mod_dingaling (gtalk). I can make call from FS to Gtalk, and from Gtalk to FS. But I have a problem, in jingle_profile I have a file like this : here when I put an user account like john or bob its doesn't work whereas I put something like 1000 or 8400 it works. When I tried to put a real phone number It doesn't work too (I have a gateway with my PBX). Somebody know, why it doesn't work with name and work with number ? Thanks. -- Jonathan BAROU SQLI LYON - CRCI 0472405368 jbarou at sqli.com lyon.crci at sqli.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/6d60edb4/attachment.html From qinglan_zeng at hotmail.com Wed Nov 4 08:54:09 2009 From: qinglan_zeng at hotmail.com (=?gb2312?B?tPPE4MjL?=) Date: Wed, 4 Nov 2009 16:54:09 +0000 Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: Hello All, Newbie to FS and I installed FS using Windows precompiled binaries. I want to set up some skype trunks with FS and so I followed the instructions while get some errors: (1). Launch Skype by clicking the skype.exe. (2). Launch FS (3) In FS I enter the cmd as: load mod_skypiax and then come to error: module load file routine retured an error. I had saved a screenshot for your referrence. Any idea on this? Thanks Daniel Zeng From: freeswitch-users-request at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 41, Issue 27 To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 07:46:45 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --??????-- From: diego.viola at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 15:25:55 +0000 Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Hello, I tried to help Roy with this issue yesterday, I saw that calls couldn't go through and then I made a sofia profile internal siptrace on. Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and saw he had like 800 sessions. I thought it was an ACL issue but it wasn't, it seems like he reached a session limit, when I restarted his FS the problem went away. Best Regards, Diego On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen wrote: Here's what's in switch.conf.xml: Yet this message: SIP/2.0 503 Maximum Calls In Progress This is a small medical practice, 5-6 extensions, 3000 outbound minutes per month and at least the same inbound. We did fsctl shutdown restart and it flushed the sessions. What is going on? Thank you for your help! -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations" _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --??????-- From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 09:39:21 -0600 Subject: Re: [Freeswitch-users] SIP Overlap support? You cannot. This is how the sip spec works. Every new invite is a new call and a new trip to the dialplan. You will probably need to design your code to send the appropriate 484 and be prepared to exit and be called again with the new digits. On Wed, Nov 4, 2009 at 2:23 AM, Dennis wrote: is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive? so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits. 2009/11/3 Anthony Minessale : > The patch was it's ability to accept subsequent invites. > Your problem is that in sip each new attempt to send an invite is another > call. > > 484 is a final response so the call with too few digits is terminated. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 --??????-- From: mike at jerris.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 10:43:10 -0500 Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Call loop? On Nov 4, 2009, at 10:25 AM, Diego Viola wrote: > Hello, > > I tried to help Roy with this issue yesterday, I saw that calls > couldn't go through and then I made a sofia profile internal > siptrace on. > > Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" > and saw he had like 800 sessions. > > I thought it was an ACL issue but it wasn't, it seems like he > reached a session limit, when I restarted his FS the problem went > away. > > Best Regards, > > Diego --??????-- From: jlenk at frontiernet.net To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 07:43:54 -0800 Subject: Re: [Freeswitch-users] Precompiled Windows Binaries Hi Carlos, very cool that the x64 version is included now! Hopefully this will get more people using the x64 version under Windows! Regards, Jeff Carlos Talbot wrote: > > On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson > wrote: > >> Jeff, >> >> thanks a lot for the reply. I was a little confused by the fact that the >> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >> that I might lose something. As you say though, think that I'll cross my >> fingers and try the updated release. I am running FreeSwitch on a test >> machine at the moment until the target hardware arrives - hopefully >> tomorrow, so I can afford to have a little play. >> > > I usually try to update the svn file at least once a month. I have a new > version ready that was compiled last night but am ironing out login issues > with the FS dudes for upload access. Also, the SVN snapshot now includes > binaries for 32 and 64 bit. It no longer includes flite though as the > install file was approaching 80MB in size. I will revisit this later if > others feel it important to include flite. > >> >> You mentioned FreePBX V3. I had been fumbling around trying to work out >> what >> this is and from what I've read, it seems to provide a GUI Front End for >> configuring FreeSwitch ? >> > Yes, it's still in development phase and as such not ready for production > use. > >> >> I am guessing that while it has been installed with FreeSwitch, I then >> need >> to run the FreePBX Installer to update the FreePBX/FreeSwitch >> configuration >> on my hardware ? >> >> >> When I start FreeSwitch, it does not automatically load the WAMPServer. >> >> Freeswitch and WAMPServer are independant of each other. WAMPServer is > bundled in this install for the purpose of FreePBX as MySQL, Apache and > PHP > are all required components of FreePBX. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a > web >> browser, I can see the WampServer logo and various tools such as >> phpinfo() >> and phpmyadmin. FreePBX is there under Your Projects. >> >> If you want to configure FreePBX you need to click on the FreePBX.url > shortcut that gets created on your desktop. > > >> When I opened this up the first time, it appeared to want to install >> FreePBX >> over FreeSwitch, I tried to abort this when it was going to overwrite >> some >> FreeSwitch conf files and I thought I'd better not go on until I had a >> better idea what was happening. I backed out of the FreePBX install and >> now >> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >> it >> looks like I'm going to have to reinstall anyway. >> >> So, for next time,am I right in thinking that I should proceed with >> running >> the FreePBX install from the WAMPServer menu ? >> > > No, launch it from the shortcut as stated above. Unfortunately, at this > time > there is very little user documentation on configuring FreePBX. Here is > the > link to the developer's info: http://www.freepbx.org/v3 > > regards, > > Carlos > >> >> >> ----- Original Message ----- >> From: "Jeff Lenk" >> To: >> Sent: Tuesday, November 03, 2009 2:48 PM >> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >> >> >> > >> > Hi Dave, >> > >> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >> > >> > Just as you said freeswitch-1.0.4.exe is the tagged release and >> > freeswitch.exe is a newer svn snapshot. >> > >> > There should be no problems installing the new version allthough best >> to >> > just try and see! >> > >> > Not sure why the newest one is from October 7th. >> > >> > Jeff >> > >> > >> > Dave Stevenson wrote: >> >> >> >> Hi, >> >> >> >> I have read the Docs on the Wiki >> >> ( >> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> >> but am still not sure of what the different Windows install files are. >> >> Currently, the Windows Installer directory contains :- >> >> >> >> LATEST_SVN_15106 - 6 Bytes >> >> >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> >> >> freeswitch.exe - 32 Megabytes >> >> >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> >> September. The freeswitch.exe file is dated 7th October and think that >> it >> >> contains the minor updates since 3rd September ? >> >> >> >> Could someone who knows FreeSwitch under windows help me understand >> the >> >> two files please ? >> >> >> >> I chickened out of running the later exe in case it did something to >> the >> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >> with >> >> the old one already installed ? >> >> What will it actually do ? >> >> >> >> regards >> >> Dave >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > -- >> > View this message in context: >> > >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html Sent from the freeswitch-users mailing list archive at Nabble.com. --??????-- From: tculjaga at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 16:44:08 +0100 Subject: Re: [Freeswitch-users] SIP Overlap support? Brian is right, pls, lets stop with exceptions and get stick to RFCs... otherwise it will be a big mess ... T. On Wed, Nov 4, 2009 at 3:03 PM, Brian West wrote: I'm going to say No! /b On Nov 4, 2009, at 2:23 AM, Dennis wrote: > is there a way to send something like 484 (or something else), which > does not make it a final answer and keep the call/socket alive? > > so we can ask the cirpack for further digits and decide what to do, if > the cirpack does not send any digits. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --??????-- From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 09:46:11 -0600 Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Which revision of FreeSWITCH are you using? On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen wrote: Here's what's in switch.conf.xml: Yet this message: SIP/2.0 503 Maximum Calls In Progress This is a small medical practice, 5-6 extensions, 3000 outbound minutes per month and at least the same inbound. We did fsctl shutdown restart and it flushed the sessions. What is going on? Thank you for your help! -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations" _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _________________________________________________________________ ?????????????????msn????? http://ditu.live.com/?form=TL&swm=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/4556646d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: skypiax load error.doc Type: application/msword Size: 113152 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/4556646d/attachment-0001.doc From gmaruzz at celliax.org Wed Nov 4 09:18:47 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 4 Nov 2009 18:18:47 +0100 Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: <7b197bef0911040918u6cfd7f46s157e90fceacd5ddd@mail.gmail.com> 2009/11/4 ??? : > Newbie to FS and I installed FS using Windows precompiled binaries. I want > to set up some skype trunks with FS and so I followed the instructions while > get some errors: > (1). Launch Skype by clicking the skype.exe. > (2). Launch FS > (3) In FS I enter the cmd as: load mod_skypiax and then come to error: > module load file routine retured an error. I had saved a screenshot for your > referrence. > Please Daniel, do not send mail both to me personally and to the mailing list. Send only to the mailing list. As you can see in the screenshot you attach, mod_skypiax cannot find its configuration file. For what I can understand, you have not the skills needed for administering FS, so it would be better if you find someone (a friend, etc) that can help you. -giovanni > Any idea on this? > > Thanks > Daniel Zeng > > From: freeswitch-users-request at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 41, Issue 27 > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 07:46:45 -0800 > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > --??????-- > From: diego.viola at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date : Wed, 4 Nov 2009 15:25:55 +0000 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Hello, > > I tried to help Roy with this issue yesterday, I saw that calls couldn't go > through and then I made a sofia profile internal siptrace on. > > Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and saw > he had like 800 sessions. > > > I thought it was an ACL issue but it wasn't, it seems like he reached a > session limit, when I restarted his FS the problem went away. > > Best Regards, > > Diego > > On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen wrote: > > Here's what's in switch.conf.xml: > > > > > > > > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > > per month and at least the same inbound. We did fsctl shutdown restart > > and it flushed the sessions. What is going on? > > > > Thank you for your help! > > > > -- > > Roy A Cohen > > Network Advantage LLC > > www.net-vantage.com > > 413.223.9007 option 1 > > -------------------------------------------------- > > "Bringing Cost-Saving, State-of-the-Art Technology > > Solutions to Small and M id-Size Organizations" > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > --??????-- > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 09:39:21 -0600 > Subject: Re: [Freeswitch-users] SIP Overlap support? > > You cannot. > This is how the sip spec works. > Every new invite is a new call and a new trip to the dialplan. > > You will probabl y need to design your code to send the appropriate 484 and > be prepared to exit and be called again with the new digits. > > > > On Wed, Nov 4, 2009 at 2:23 AM, Dennis wrote: > > is there a way to send something like 484 (or something else), which > > does not make it a final answer and keep the call/socket alive? > > > > so we can ask the cirpack for further digits and decide what to do, if > > the cirpack does not send any digits. > > > > > > > > 2009/11/3 Anthony Minessale : > >> The patch was it's ability to accept subsequent invites. > >> Your problem is that in sip each new attempt to send an invite is another > >> call. > >> > >> 484 is a final response so the call with too few digits is terminated. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > --??????-- > From: mike at jerris.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 10:43:10 -0500 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Call loop? > > On Nov 4, 2009, at 10:25 AM, Diego Viola wrote: > >> Hello, >> >> I tried to help Roy with this issue yesterday, I saw that ca > lls >> couldn't go through and then I made a sofia profile internal >> siptrace on. >> >> Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" >> and saw he had like 800 sessions. >> >> I thought it was an ACL issue but it wasn't, it seems like he >> reached a session limit, when I restarted his FS the problem went >> away. >> >> Best Regards, >> >> Diego > > > > > > --??????-- > From: jlenk at frontiernet.net > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 07:43:54 -0800 > Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > > Hi Carlos, > > very cool that the x64 version is included now! Hopefully this will get more > people using the x64 version under Windows! > > Regards, > Jeff > > > Carlos Talbot wrote: >> >> On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson >> ebank.net>wrote: >> >>> Jeff, >>> >>> thanks a lot for the reply. I was a little confused by the fact that the >>> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >>> that I might lose something. As you say though, think that I'll cross my >>> fingers and try the updated release. I am running FreeSwitch on a test >>> machine at the moment until the target hardware arrives - hopefully >>> tomorrow, so I can afford to have a little play. >>> >> >> I usually try to update the svn file at least once a month. I have a new >> version ready that was compiled last night but am ironing out login issues >> with the FS dudes for upload access. Also, the SVN snapshot now includes >> binaries for 32 and 64 bit. It no longer includes flite though as the >> install file was approaching 80MB in size. I will revisit this later if >> others feel it impo > rtant to include flite. >> >>> >>> You mentioned FreePBX V3. I had been fumbling around trying to work out >>> what >>> this is and from what I've read, it seems to provide a GUI Front End for >>> configuring FreeSwitch ? >>> >> Yes, it's still in development phase and as such not ready for production >> use. >> >>> >>> I am guessing that while it has been installed with FreeSwitch, I then >>> need >>> to run the FreePBX Installer to update the FreePBX/FreeSwitch >>> configuration >>> on my hardware ? >>> >>> >>> When I start FreeSwitch, it does not automatically load the WAMPServer. >>> >>> Freeswitch and WAMPServer are independant of each other. WAMPServer is >> bundled in this install for the purpose of FreePBX as MySQL, Apache and >> PHP >> are all required components of FreePBX. >> >> When > I start WAMPServer manually, and open up localhost (127.0.0.1) in a >> web >>> browser, I can see the WampServer logo and various tools such as >>> phpinfo() >>> and phpmyadmin. FreePBX is there under Your Projects. >>> >>> If you want to configure FreePBX you need to click on the FreePBX.url >> shortcut that gets created on your desktop. >> >> >>> When I opened this up the first time, it appeared to want to install >>> FreePBX >>> over FreeSwitch, I tried to abort this when it was going to overwrite >>> some >>> FreeSwitch conf files and I thought I'd better not go on until I had a >>> better idea what was happening. I backed out of the FreePBX install and >>> now >>> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >>> it >>> looks like I'm going to have to reinstall anyway. >>> >>> So, for nex > t time,am I right in thinking that I should proceed with >>> running >>> the FreePBX install from the WAMPServer menu ? >>> >> >> No, launch it from the shortcut as stated above. Unfortunately, at this >> time >> there is very little user documentation on configuring FreePBX. Here is >> the >> link to the developer's info: http://www.freepbx.org/v3 >> >> regards, >> >> Carlos >> >>> >>> >>> ----- Original Message ----- >>> From: "Jeff Lenk" >>> To: >>> Sent: Tuesday, November 03, 2009 2:48 PM >>> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >>> >>> >>> > >>> > Hi Dave, >>> > >>> > These are supported by "Carlos Talbot" . They also include Freepbx v3< > BR>>> > >>> > Just as you said freeswitch-1.0.4.exe is the tagged release and >>> > freeswitch.exe is a newer svn snapshot. >>> > >>> > There should be no problems installing the new version allthough best >>> to >>> > just try and see! >>> > >>> > Not sure why the newest one is from October 7th. >>> > >>> > Jeff >>> > >>> > >>> > Dave Stevenson wrote: >>> >> >>> >> Hi, >>> >> >>> >> I have read the Docs on the Wiki >>> >> ( >>> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >>> >> but am still not sure of what the different Windows install files are. >>> >> Currently, the Windows Installer directory contains :- >>> >> >>> >> LATEST_SVN_15106 - 6 Bytes >>> >> >>> >> freeswitch-1.0.4.exe - 42 Megabytes >>> >> >>> >> freeswitch.exe - 32 Megabytes >>> >> >>> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >>> >> September. The freeswitch.exe file is dated 7th October and think that >>> it >>> >> contains the minor updates since 3rd September ? >>> >> >>> >> Could someone who knows FreeSwitch under windows help me understand >>> the >>> >> two files please ? >>> >> >>> >> I chickened out of running the later exe in case it did something to >>> the >>> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >>> with >>> >> the old one already installed ? >>> >> What will it > actually do ? >>> >> >>> >> regards >>> >> Dave >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> > >>> > -- >>> > View this message in context: >>> > >>> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3 > 937943p3938887.html >>> > Sent from the freeswitch-users mailing list archive at Nabble.com. >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http > ://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http > ://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > --??????-- > From: tculjaga at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 16:44:08 +0100 > Subject: Re: [Freeswitch-users] SIP Overlap support? > > Brian is right, > > pls, lets stop with exceptions and get stick to RFCs... otherwise it will be > a big mess ... > > T. > > On Wed, Nov 4, 2009 at 3:03 PM, Brian West wrote: > > I'm going to say No! > > > > /b > > > > On Nov 4, 2009, at 2:23 AM, Dennis wrote: > > > >> is there a way to send something like 484 (or something else), which > >> does not make it a final answer and keep the call/socket alive? > >> > >> so we can ask the cirpack for further digits and decide what to do, if > >> the cirpack does not send any digits. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > --??????-- > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 09:46:11 -0600 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Which revision of FreeSWITCH are you using? > > > On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen wrote: > > Here's what's in switch.conf.xml: > > > > > > > > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > > per month and at least the same inbound. We did fsctl shutdown restart > > and it flushed the sessions. What is going on? > > > > Thank you for your help! > > > > -- > > Roy A Cohen > > Network Advantage LLC > > www.net-vantage.com > > 413.223.9007 option 1 > > -------------------------------------------------- > > "Bringing Cost-Saving, State-of-the-Art Technology > > Solutions to Small and M id-Size Organizations" > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > ________________________________ > ??????????MSN??? ????? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at celliax.org Wed Nov 4 09:22:29 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 4 Nov 2009 18:22:29 +0100 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <1257349434463-3946039.post@n2.nabble.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <1257349434463-3946039.post@n2.nabble.com> Message-ID: <7b197bef0911040922y21aa10c0l53909b9f3ea07c5@mail.gmail.com> On Wed, Nov 4, 2009 at 4:43 PM, Jeff Lenk wrote: > > very cool that the x64 version is included now! Hopefully this will get more > people using the x64 version under Windows! ...and will get more people using the x64 version of Windows! ;) -gm > > Regards, > Jeff > > > Carlos Talbot wrote: >> >> On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson >> wrote: >> >>> Jeff, >>> >>> thanks a lot for the reply. I was a little confused by the fact that the >>> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >>> that I might lose something. As you say though, think that I'll cross my >>> fingers and try the updated release. I am running FreeSwitch on a test >>> machine at the moment until the target hardware arrives - hopefully >>> tomorrow, so I can afford to have a little play. >>> >> >> I usually try to update the svn file at least once a month. I have a new >> version ready that was compiled last night but am ironing out login issues >> with the FS dudes for upload access. Also, the SVN snapshot now includes >> binaries for 32 and 64 bit. It no longer includes flite though as the >> install file was approaching 80MB in size. I will revisit this later if >> others feel it important to include flite. >> >>> >>> You mentioned FreePBX V3. I had been fumbling around trying to work out >>> what >>> this is and from what I've read, it seems to provide a GUI Front End for >>> configuring FreeSwitch ? >>> >> Yes, it's still in development phase and as such not ready for production >> use. >> >>> >>> I am guessing that while it has been installed with FreeSwitch, I then >>> need >>> to run the FreePBX Installer to update the FreePBX/FreeSwitch >>> configuration >>> on my hardware ? >>> >>> >>> When I start FreeSwitch, it does not automatically load the WAMPServer. >>> >>> Freeswitch and WAMPServer are independant of each other. WAMPServer is >> bundled in this install for the purpose of FreePBX as MySQL, Apache and >> PHP >> are all required components of FreePBX. >> >> When I start WAMPServer manually, and open up localhost (127.0.0.1) in a >> web >>> browser, I can see the WampServer logo and various tools such as >>> phpinfo() >>> and phpmyadmin. FreePBX is there under Your Projects. >>> >>> If you want to configure FreePBX you need to click on the FreePBX.url >> shortcut that gets created on your desktop. >> >> >>> When I opened this up the first time, it appeared to want to install >>> FreePBX >>> over FreeSwitch, I tried to abort this when it was going to overwrite >>> some >>> FreeSwitch conf files and I thought I'd better not go on until I had a >>> better idea what was happening. I backed out of the FreePBX install and >>> now >>> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >>> it >>> looks like I'm going to have to reinstall anyway. >>> >>> So, for next time,am I right in thinking that I should proceed with >>> running >>> the FreePBX install from the WAMPServer menu ? >>> >> >> No, launch it from the shortcut as stated above. Unfortunately, at this >> time >> there is very little user documentation on configuring FreePBX. Here is >> the >> link to the developer's info: http://www.freepbx.org/v3 >> >> regards, >> >> Carlos >> >>> >>> >>> ----- Original Message ----- >>> From: "Jeff Lenk" >>> To: >>> Sent: Tuesday, November 03, 2009 2:48 PM >>> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >>> >>> >>> > >>> > Hi Dave, >>> > >>> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >>> > >>> > Just as you said freeswitch-1.0.4.exe is the tagged release and >>> > freeswitch.exe is a newer svn snapshot. >>> > >>> > There should be no problems installing the new version allthough best >>> to >>> > just try and see! >>> > >>> > Not sure why the newest one is from October 7th. >>> > >>> > Jeff >>> > >>> > >>> > Dave Stevenson wrote: >>> >> >>> >> Hi, >>> >> >>> >> I have read the Docs on the Wiki >>> >> ( >>> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >>> >> but am still not sure of what the different Windows install files are. >>> >> Currently, the Windows Installer directory contains :- >>> >> >>> >> LATEST_SVN_15106 - 6 Bytes >>> >> >>> >> freeswitch-1.0.4.exe - 42 Megabytes >>> >> >>> >> freeswitch.exe - 32 Megabytes >>> >> >>> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >>> >> September. The freeswitch.exe file is dated 7th October and think that >>> it >>> >> contains the minor updates since 3rd September ? >>> >> >>> >> Could someone who knows FreeSwitch under windows help me understand >>> the >>> >> two files please ? >>> >> >>> >> I chickened out of running the later exe in case it did something to >>> the >>> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >>> with >>> >> the old one already installed ? >>> >> What will it actually do ? >>> >> >>> >> regards >>> >> Dave >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> > >>> > -- >>> > View this message in context: >>> > >>> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >>> > Sent from the freeswitch-users mailing list archive at Nabble.com. >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From jlenk at frontiernet.net Wed Nov 4 09:28:07 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Wed, 4 Nov 2009 09:28:07 -0800 (PST) Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: <1257355687424-3946759.post@n2.nabble.com> follow these intructions -> http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances ??? wrote: > > > Hello All, > > > > Newbie to FS and I installed FS using Windows precompiled binaries. I want > to set up some skype trunks with FS and so I followed the instructions > while get some errors: > > (1). Launch Skype by clicking the skype.exe. > > (2). Launch FS > > (3) In FS I enter the cmd as: load mod_skypiax and then come to error: > module load file routine retured an error. I had saved a screenshot for > your referrence. > > > > Any idea on this? > > > > Thanks > > Daniel Zeng > > From: freeswitch-users-request at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 41, Issue 27 > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 07:46:45 -0800 > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > --??????-- > From: diego.viola at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 15:25:55 +0000 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Hello, > > I tried to help Roy with this issue yesterday, I saw that calls couldn't > go through and then I made a sofia profile internal siptrace on. > > Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and > saw he had like 800 sessions. > > > I thought it was an ACL issue but it wasn't, it seems like he reached a > session limit, when I restarted his FS the problem went away. > > Best Regards, > > Diego > > > On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen wrote: > > > Here's what's in switch.conf.xml: > > > > > > > > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > > per month and at least the same inbound. We did fsctl shutdown restart > > and it flushed the sessions. What is going on? > > > > Thank you for your help! > > > > -- > > Roy A Cohen > > Network Advantage LLC > > www.net-vantage.com > > 413.223.9007 option 1 > > -------------------------------------------------- > > "Bringing Cost-Saving, State-of-the-Art Technology > > Solutions to Small and Mid-Size Organizations" > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > --??????-- > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 09:39:21 -0600 > Subject: Re: [Freeswitch-users] SIP Overlap support? > > You cannot. > This is how the sip spec works. > Every new invite is a new call and a new trip to the dialplan. > > You will probably need to design your code to send the appropriate 484 and > be prepared to exit and be called again with the new digits. > > > > > On Wed, Nov 4, 2009 at 2:23 AM, Dennis wrote: > > > is there a way to send something like 484 (or something else), which > > does not make it a final answer and keep the call/socket alive? > > > > so we can ask the cirpack for further digits and decide what to do, if > > the cirpack does not send any digits. > > > > > > > > 2009/11/3 Anthony Minessale : > > >> The patch was it's ability to accept subsequent invites. > >> Your problem is that in sip each new attempt to send an invite is another > >> call. > >> > >> 484 is a final response so the call with too few digits is terminated. > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > --??????-- > From: mike at jerris.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 10:43:10 -0500 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Call loop? > > On Nov 4, 2009, at 10:25 AM, Diego Viola wrote: > >> Hello, >> >> I tried to help Roy with this issue yesterday, I saw that calls >> couldn't go through and then I made a sofia profile internal >> siptrace on. >> >> Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" >> and saw he had like 800 sessions. >> >> I thought it was an ACL issue but it wasn't, it seems like he >> reached a session limit, when I restarted his FS the problem went >> away. >> >> Best Regards, >> >> Diego > > > > > > --??????-- > From: jlenk at frontiernet.net > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 07:43:54 -0800 > Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > > Hi Carlos, > > very cool that the x64 version is included now! Hopefully this will get > more > people using the x64 version under Windows! > > Regards, > Jeff > > > Carlos Talbot wrote: >> >> On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson >> wrote: >> >>> Jeff, >>> >>> thanks a lot for the reply. I was a little confused by the fact that the >>> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >>> that I might lose something. As you say though, think that I'll cross my >>> fingers and try the updated release. I am running FreeSwitch on a test >>> machine at the moment until the target hardware arrives - hopefully >>> tomorrow, so I can afford to have a little play. >>> >> >> I usually try to update the svn file at least once a month. I have a new >> version ready that was compiled last night but am ironing out login >> issues >> with the FS dudes for upload access. Also, the SVN snapshot now includes >> binaries for 32 and 64 bit. It no longer includes flite though as the >> install file was approaching 80MB in size. I will revisit this later if >> others feel it important to include flite. >> >>> >>> You mentioned FreePBX V3. I had been fumbling around trying to work out >>> what >>> this is and from what I've read, it seems to provide a GUI Front End for >>> configuring FreeSwitch ? >>> >> Yes, it's still in development phase and as such not ready for production >> use. >> >>> >>> I am guessing that while it has been installed with FreeSwitch, I then >>> need >>> to run the FreePBX Installer to update the FreePBX/FreeSwitch >>> configuration >>> on my hardware ? >>> >>> >>> When I start FreeSwitch, it does not automatically load the WAMPServer. >>> >>> Freeswitch and WAMPServer are independant of each other. WAMPServer is >> bundled in this install for the purpose of FreePBX as MySQL, Apache and >> PHP >> are all required components of FreePBX. >> >> When I start WAMPServer manually, and open up localhost (127.0.0.1) in a >> web >>> browser, I can see the WampServer logo and various tools such as >>> phpinfo() >>> and phpmyadmin. FreePBX is there under Your Projects. >>> >>> If you want to configure FreePBX you need to click on the FreePBX.url >> shortcut that gets created on your desktop. >> >> >>> When I opened this up the first time, it appeared to want to install >>> FreePBX >>> over FreeSwitch, I tried to abort this when it was going to overwrite >>> some >>> FreeSwitch conf files and I thought I'd better not go on until I had a >>> better idea what was happening. I backed out of the FreePBX install and >>> now >>> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >>> it >>> looks like I'm going to have to reinstall anyway. >>> >>> So, for next time,am I right in thinking that I should proceed with >>> running >>> the FreePBX install from the WAMPServer menu ? >>> >> >> No, launch it from the shortcut as stated above. Unfortunately, at this >> time >> there is very little user documentation on configuring FreePBX. Here is >> the >> link to the developer's info: http://www.freepbx.org/v3 >> >> regards, >> >> Carlos >> >>> >>> >>> ----- Original Message ----- >>> From: "Jeff Lenk" >>> To: >>> Sent: Tuesday, November 03, 2009 2:48 PM >>> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >>> >>> >>> > >>> > Hi Dave, >>> > >>> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >>> > >>> > Just as you said freeswitch-1.0.4.exe is the tagged release and >>> > freeswitch.exe is a newer svn snapshot. >>> > >>> > There should be no problems installing the new version allthough best >>> to >>> > just try and see! >>> > >>> > Not sure why the newest one is from October 7th. >>> > >>> > Jeff >>> > >>> > >>> > Dave Stevenson wrote: >>> >> >>> >> Hi, >>> >> >>> >> I have read the Docs on the Wiki >>> >> ( >>> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >>> >> but am still not sure of what the different Windows install files >>> are. >>> >> Currently, the Windows Installer directory contains :- >>> >> >>> >> LATEST_SVN_15106 - 6 Bytes >>> >> >>> >> freeswitch-1.0.4.exe - 42 Megabytes >>> >> >>> >> freeswitch.exe - 32 Megabytes >>> >> >>> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >>> >> September. The freeswitch.exe file is dated 7th October and think >>> that >>> it >>> >> contains the minor updates since 3rd September ? >>> >> >>> >> Could someone who knows FreeSwitch under windows help me understand >>> the >>> >> two files please ? >>> >> >>> >> I chickened out of running the later exe in case it did something to >>> the >>> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >>> with >>> >> the old one already installed ? >>> >> What will it actually do ? >>> >> >>> >> regards >>> >> Dave >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> > >>> > -- >>> > View this message in context: >>> > >>> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >>> > Sent from the freeswitch-users mailing list archive at Nabble.com. >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > --??????-- > From: tculjaga at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 16:44:08 +0100 > Subject: Re: [Freeswitch-users] SIP Overlap support? > > Brian is right, > > pls, lets stop with exceptions and get stick to RFCs... otherwise it will > be a big mess ... > > T. > > > On Wed, Nov 4, 2009 at 3:03 PM, Brian West wrote: > > > I'm going to say No! > > > > /b > > > > > On Nov 4, 2009, at 2:23 AM, Dennis wrote: > > > >> is there a way to send something like 484 (or something else), which > >> does not make it a final answer and keep the call/socket alive? > >> > >> so we can ask the cirpack for further digits and decide what to do, if > >> the cirpack does not send any digits. > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > --??????-- > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 09:46:11 -0600 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Which revision of FreeSWITCH are you using? > > > > On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen wrote: > > > Here's what's in switch.conf.xml: > > > > > > > > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > > per month and at least the same inbound. We did fsctl shutdown restart > > and it flushed the sessions. What is going on? > > > > Thank you for your help! > > > > -- > > Roy A Cohen > > Network Advantage LLC > > www.net-vantage.com > > 413.223.9007 option 1 > > -------------------------------------------------- > > "Bringing Cost-Saving, State-of-the-Art Technology > > Solutions to Small and Mid-Size Organizations" > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _________________________________________________________________ > ?????????????????msn????? > http://ditu.live.com/?form=TL&swm=1 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Skypiax-load-error-tp3946656p3946759.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Nov 4 09:33:15 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Nov 2009 11:33:15 -0600 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <7b197bef0911040922y21aa10c0l53909b9f3ea07c5@mail.gmail.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <1257349434463-3946039.post@n2.nabble.com> <7b197bef0911040922y21aa10c0l53909b9f3ea07c5@mail.gmail.com> Message-ID: I looked out my window... but I didn't see pigs flying... did I miss something! :P /b On Nov 4, 2009, at 11:22 AM, Giovanni Maruzzelli wrote: > ...and will get more people using the x64 version of Windows! ;) > > -gm From msc at freeswitch.org Wed Nov 4 09:36:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Nov 2009 09:36:38 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7170307767D@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> <3C04B27FC880044F8FCD735D0D952FF7170307767D@EXMBXCLUS01.citservers.local> Message-ID: <87f2f3b90911040936j12094470ld8e4d5328cc109f3@mail.gmail.com> On Tue, Nov 3, 2009 at 6:27 PM, Ujjval Karihaloo wrote: > Was that sarcasm or you really mean it? > No, he's serious. There are some issues that are quite endemic to AT&T. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/8ce0d1b3/attachment.html From stevendt at primrosebank.net Wed Nov 4 09:49:42 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 17:49:42 -0000 Subject: [Freeswitch-users] Precompiled Windows Binaries References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> Message-ID: <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> Hi Carlos, thanks a lot for the reply on the Windows stuff - and for doing the pre-compile ! It saves me from having to do that myself while I'm busy just trying to learn FreeSwitch - not an easy task in itself, well, not for me anyway !. As much as a GUI would make working with FS easier, I think I'll keep away from FreePBX at the moment, I'm liable just to get myself more confused and break what I have working. I'll wait for a "Production Ready" version before I mess with this again I think. Just one clarification then, am I right in thinking that, after I have installed a release version of FS, if I then install one of the SVNs over it, it will keep all configuration etc. in tact, i.e., I won't lose anything that I've changed in the conf files etc? Regards Dave ----- Original Message ----- From: Carlos Talbot To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, November 04, 2009 2:51 PM Subject: Re: [Freeswitch-users] Precompiled Windows Binaries On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson wrote: Jeff, thanks a lot for the reply. I was a little confused by the fact that the "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried that I might lose something. As you say though, think that I'll cross my fingers and try the updated release. I am running FreeSwitch on a test machine at the moment until the target hardware arrives - hopefully tomorrow, so I can afford to have a little play. I usually try to update the svn file at least once a month. I have a new version ready that was compiled last night but am ironing out login issues with the FS dudes for upload access. Also, the SVN snapshot now includes binaries for 32 and 64 bit. It no longer includes flite though as the install file was approaching 80MB in size. I will revisit this later if others feel it important to include flite. You mentioned FreePBX V3. I had been fumbling around trying to work out what this is and from what I've read, it seems to provide a GUI Front End for configuring FreeSwitch ? Yes, it's still in development phase and as such not ready for production use. I am guessing that while it has been installed with FreeSwitch, I then need to run the FreePBX Installer to update the FreePBX/FreeSwitch configuration on my hardware ? When I start FreeSwitch, it does not automatically load the WAMPServer. Freeswitch and WAMPServer are independant of each other. WAMPServer is bundled in this install for the purpose of FreePBX as MySQL, Apache and PHP are all required components of FreePBX. When I start WAMPServer manually, and open up localhost (127.0.0.1) in a web browser, I can see the WampServer logo and various tools such as phpinfo() and phpmyadmin. FreePBX is there under Your Projects. If you want to configure FreePBX you need to click on the FreePBX.url shortcut that gets created on your desktop. When I opened this up the first time, it appeared to want to install FreePBX over FreeSwitch, I tried to abort this when it was going to overwrite some FreeSwitch conf files and I thought I'd better not go on until I had a better idea what was happening. I backed out of the FreePBX install and now I can't get the FreePBX or phpmyadmin pages up again (missing files) so it looks like I'm going to have to reinstall anyway. So, for next time,am I right in thinking that I should proceed with running the FreePBX install from the WAMPServer menu ? No, launch it from the shortcut as stated above. Unfortunately, at this time there is very little user documentation on configuring FreePBX. Here is the link to the developer's info: http://www.freepbx.org/v3 regards, Carlos ----- Original Message ----- From: "Jeff Lenk" To: Sent: Tuesday, November 03, 2009 2:48 PM Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > Hi Dave, > > These are supported by "Carlos Talbot" . They also include Freepbx v3 > > Just as you said freeswitch-1.0.4.exe is the tagged release and > freeswitch.exe is a newer svn snapshot. > > There should be no problems installing the new version allthough best to > just try and see! > > Not sure why the newest one is from October 7th. > > Jeff > > > Dave Stevenson wrote: >> >> Hi, >> >> I have read the Docs on the Wiki >> (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> but am still not sure of what the different Windows install files are. >> Currently, the Windows Installer directory contains :- >> >> LATEST_SVN_15106 - 6 Bytes >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> freeswitch.exe - 32 Megabytes >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> September. The freeswitch.exe file is dated 7th October and think that it >> contains the minor updates since 3rd September ? >> >> Could someone who knows FreeSwitch under windows help me understand the >> two files please ? >> >> I chickened out of running the later exe in case it did something to the >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe with >> the old one already installed ? >> What will it actually do ? >> >> regards >> Dave >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/5eeb37ef/attachment.html From jmillican at sentinelcommunications.com Wed Nov 4 09:52:18 2009 From: jmillican at sentinelcommunications.com (John Millican) Date: Wed, 04 Nov 2009 12:52:18 -0500 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <1257349434463-3946039.post@n2.nabble.com> <7b197bef0911040922y21aa10c0l53909b9f3ea07c5@mail.gmail.com> Message-ID: <4AF1BF52.7010000@sentinelcommunications.com> Brian West wrote: > I looked out my window... but I didn't see pigs flying... did I miss > something! :P > > /b > > On Nov 4, 2009, at 11:22 AM, Giovanni Maruzzelli wrote: > >> ...and will get more people using the x64 version of Windows! ;) >> >> -gm When their own commercials say that there old software is prone to crashing and hangs, why should I trust there "new" software? -- JohnM From stevendt at primrosebank.net Wed Nov 4 10:03:26 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 18:03:26 -0000 Subject: [Freeswitch-users] Gateway Error References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> Message-ID: Hi, I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway and am seeing the following error :- "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sigura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who know what will happen.." Having just bought the Gateway specifically for FS, that was a bit of a "rude awakening" ! Does anyone know of a fix in the pipeline, or am I sc***ed already ? Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/df99f265/attachment.html From ujjval at simplesignal.com Wed Nov 4 10:11:43 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Wed, 4 Nov 2009 10:11:43 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <87f2f3b90911040936j12094470ld8e4d5328cc109f3@mail.gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> <3C04B27FC880044F8FCD735D0D952FF7170307767D@EXMBXCLUS01.citservers.local> <87f2f3b90911040936j12094470ld8e4d5328cc109f3@mail.gmail.com> Message-ID: <3C04B27FC880044F8FCD735D0D952FF717030777DE@EXMBXCLUS01.citservers.local> Interesting...Thx for the heads up. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, November 04, 2009 10:37 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator On Tue, Nov 3, 2009 at 6:27 PM, Ujjval Karihaloo > wrote: Was that sarcasm or you really mean it? No, he's serious. There are some issues that are quite endemic to AT&T. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/62658188/attachment.html From msc at freeswitch.org Wed Nov 4 10:16:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Nov 2009 10:16:43 -0800 Subject: [Freeswitch-users] Users hanged up for unknown reason In-Reply-To: <1257244093831-3937601.post@n2.nabble.com> References: <1257244093831-3937601.post@n2.nabble.com> Message-ID: <87f2f3b90911041016u620ca88bk4f0d6a4ceb339b4b@mail.gmail.com> On Tue, Nov 3, 2009 at 2:28 AM, Maciej Aniserowicz < maciej.aniserowicz at gmail.com> wrote: > > Hi, > I have a strange problem. I control FS with commands sent by tcp in > response > to events published via tcp. I do something like: > 1) call 1st user > 2) call 2nd user > 3) 1st and 2nd talk > 4) call another user > 5) 1st and another talk > etc... > > Sometimes (quite regularly) users are hanged up (with cause > NORMAL_CLEARING) > even if they do not hangup manually. > > I pasted one such scenario in pastebin > (http://pastebin.freeswitch.org/10955), it includes logs from commands > sent > by me and events received from FS. Could someone take a look and see what > am > I doing wrong? > Seeing only the events it is difficult to see what triggered them. Can you repeat these tests and capture the debug output from the CLI? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/afc91d6c/attachment.html From msc at freeswitch.org Wed Nov 4 10:23:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Nov 2009 10:23:01 -0800 Subject: [Freeswitch-users] Gateway Error In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> Message-ID: <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> On Wed, Nov 4, 2009 at 10:03 AM, Dave Stevenson wrote: > Hi, > > I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway > and am seeing the following error :- > > "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they meant > to say was 20 > This issue has so far been identified to happen on the following broken > platforms/devices: > Linksys/Sigura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so broken > who know what will happen.." > > Having just bought the Gateway specifically for FS, that was a bit of a > "rude awakening" ! > > Does anyone know of a fix in the pipeline, or am I sc***ed already ? > The cynical among us will say that you were hosed the moment you paid for a Linksys device. :) It's very sad but the FS devs find this kind of thing all the time. They've literally got all sorts of checks in the code to make sure that devices aren't saying one thing and doing something else. Cisco is not the only one to do stupid things like this. In any case, just be aware of it. If you want suggestions then list to the others here who can offer their experiences with various devices they have in production. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/ced0a94c/attachment.html From stevendt at primrosebank.net Wed Nov 4 10:48:28 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 18:48:28 -0000 Subject: [Freeswitch-users] Gateway Error References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com><6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> Message-ID: <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> Michael et al - and specifically, the FS Developers, this is all the more annoying given the fact that the SPA-3102 was bought specifically to run with FreeSwitch following a recommendation here in the UK. It was just unwrapped this afternoon :-( (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). I am setting up a VOIP system at home, and this device sounded like the ideal gateway to the PSTN. What does the error message actually mean - is this device a non-starter or are there work-arounds or fixes to the code in progress ? Surely the device can't be as "broken" as the message - or am I just being too hopeful ? Regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, November 04, 2009 6:23 PM Subject: Re: [Freeswitch-users] Gateway Error On Wed, Nov 4, 2009 at 10:03 AM, Dave Stevenson wrote: Hi, I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway and am seeing the following error :- "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sigura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who know what will happen.." Having just bought the Gateway specifically for FS, that was a bit of a "rude awakening" ! Does anyone know of a fix in the pipeline, or am I sc***ed already ? The cynical among us will say that you were hosed the moment you paid for a Linksys device. :) It's very sad but the FS devs find this kind of thing all the time. They've literally got all sorts of checks in the code to make sure that devices aren't saying one thing and doing something else. Cisco is not the only one to do stupid things like this. In any case, just be aware of it. If you want suggestions then list to the others here who can offer their experiences with various devices they have in production. -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/4535e099/attachment-0001.html From mastermind202 at gmail.com Wed Nov 4 10:57:29 2009 From: mastermind202 at gmail.com (mm_202) Date: Wed, 4 Nov 2009 13:57:29 -0500 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <027f01ca5d6c$9c1603a0$d4420ae0$@com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> <025101ca5d10$f81228c0$e8367a40$@com> <20091104064201.GA15804@jdc.jasonjgw.net> <027f01ca5d6c$9c1603a0$d4420ae0$@com> Message-ID: <63de75710911041057x472d44aj9bc52bb460a8c8cd@mail.gmail.com> I had the exact same problem with the Cisco phones not being able to receive calls. I fixed it by messing around with the NAT settings in the internal sofia profile. From what I remember, I just removed the line and everything worked fine. -- mm_202. On Wed, Nov 4, 2009 at 11:33 AM, Peter J. Zandvoort wrote: > Absolutely agreed. To use Matthew's original car metaphor: When you just got > your learner's permit, the old Chevy may be a better choice than the > Ferrari. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason > White > Sent: Wednesday, November 04, 2009 1:42 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones > > Peter J. Zandvoort wrote: >> After looking at various asterisk distributions, SipX, 3CX and >> what-have-you, I've come to the conclusion that FreeSWITCH is by far the >> most advanced platform out there. Its architecture and performance is >> literally light years ahead of the rest and I have yet to come up with >> something that it can't do. But all that comes at a price: The learning >> curve is like scaling a brick wall. > > The most flexible and sophisticated tools tend to have this characteristic, > the best solution to which is a supportive community and good documentation. > FreeSWITCH has the community; the documentation is improving thanks to > ongoing > efforts to extend, clarify and enhance the wiki. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From testeador01 at gmail.com Wed Nov 4 11:34:09 2009 From: testeador01 at gmail.com (Milena) Date: Wed, 4 Nov 2009 14:34:09 -0500 Subject: [Freeswitch-users] Question about jingle_profiles In-Reply-To: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> References: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> Message-ID: Hello, A question to clarify: do you have an extension on your dialplan that matches "john" or "bob" and bridges the call? If you don't, you need to create it, if you do have it but it doesn't work, post the cli output for when you try to call so we can see what is going on. 2009/11/4 Jonathan Barou > Hi everybody, > > I actually working on mod_dingaling (gtalk). I can make call from FS to > Gtalk, and from Gtalk to FS. > But I have a problem, in jingle_profile I have a file like this : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > here when I put an user account like > john or bob its doesn't work whereas I put something like 1000 or 8400 it > works. > > When I tried to put a real phone number It doesn't work too (I have a > gateway with my PBX). > > Somebody know, why it doesn't work with name and work with number ? > > Thanks. > > > -- > Jonathan BAROU > SQLI LYON - CRCI > 0472405368 > jbarou at sqli.com > lyon.crci at sqli.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/c83f8580/attachment.html From mike at jerris.com Wed Nov 4 11:36:49 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 4 Nov 2009 14:36:49 -0500 Subject: [Freeswitch-users] Gateway Error In-Reply-To: <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com><6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> Message-ID: <33167DE5-D670-46F0-BECA-4802B917E206@jerris.com> It means you need to go change the setting from the broken defaults, thats all. Mike On Nov 4, 2009, at 1:48 PM, Dave Stevenson wrote: > Michael et al - and specifically, the FS Developers, > > this is all the more annoying given the fact that the SPA-3102 was > bought specifically to run with FreeSwitch following a > recommendation here in the UK. It was just unwrapped this > afternoon :-( > > (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). > > I am setting up a VOIP system at home, and this device sounded like > the ideal gateway to the PSTN. > > What does the error message actually mean - is this device a non- > starter or are there work-arounds or fixes to the code in progress ? > > Surely the device can't be as "broken" as the message - or am I just > being too hopeful ? > > Regards > Dave > > > ----- Original Message ----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, November 04, 2009 6:23 PM > Subject: Re: [Freeswitch-users] Gateway Error > > > > On Wed, Nov 4, 2009 at 10:03 AM, Dave Stevenson > wrote: > Hi, > > I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice > Gateway and am seeing the following error :- > > "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what > they meant to say was 20 > This issue has so far been identified to happen on the following > broken platforms/devices: > Linksys/Sigura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so > broken who know what will happen.." > > Having just bought the Gateway specifically for FS, that was a bit > of a "rude awakening" ! > > Does anyone know of a fix in the pipeline, or am I sc***ed already ? > > The cynical among us will say that you were hosed the moment you > paid for a Linksys device. :) It's very sad but the FS devs find > this kind of thing all the time. They've literally got all sorts of > checks in the code to make sure that devices aren't saying one thing > and doing something else. Cisco is not the only one to do stupid > things like this. In any case, just be aware of it. > > If you want suggestions then list to the others here who can offer > their experiences with various devices they have in production. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/7a7ecbe7/attachment.html From chris.chen2004 at gmail.com Wed Nov 4 11:45:02 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Wed, 4 Nov 2009 14:45:02 -0500 Subject: [Freeswitch-users] Question about jingle_profiles In-Reply-To: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> References: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> Message-ID: <507898380911041145u431865f8uc8877fce3c2e3778@mail.gmail.com> you have to define the extension "john" or "bob" or whatever number you want in the dialplan for the context "public". Just follow your jingle profile you define. Simple, no other tricks. Thanks, Chris On Wed, Nov 4, 2009 at 11:56 AM, Jonathan Barou wrote: > Hi everybody, > > I actually working on mod_dingaling (gtalk). I can make call from FS to > Gtalk, and from Gtalk to FS. > But I have a problem, in jingle_profile I have a file like this : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > here when I put an user account like > john or bob its doesn't work whereas I put something like 1000 or 8400 it > works. > > When I tried to put a real phone number It doesn't work too (I have a > gateway with my PBX). > > Somebody know, why it doesn't work with name and work with number ? > > Thanks. > > > -- > Jonathan BAROU > SQLI LYON - CRCI > 0472405368 > jbarou at sqli.com > lyon.crci at sqli.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/46d81714/attachment-0001.html From stevendt at primrosebank.net Wed Nov 4 11:49:11 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 19:49:11 -0000 Subject: [Freeswitch-users] Gateway Error References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com><6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> <33167DE5-D670-46F0-BECA-4802B917E206@jerris.com> Message-ID: <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> Phew ! Thanks Mike, I was very worried there. Now, if I just knew which were the "broken defaults", I'd know where to go next :-) Regards Dave ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, November 04, 2009 7:36 PM Subject: Re: [Freeswitch-users] Gateway Error It means you need to go change the setting from the broken defaults, thats all. Mike On Nov 4, 2009, at 1:48 PM, Dave Stevenson wrote: Michael et al - and specifically, the FS Developers, this is all the more annoying given the fact that the SPA-3102 was bought specifically to run with FreeSwitch following a recommendation here in the UK. It was just unwrapped this afternoon :-( (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). I am setting up a VOIP system at home, and this device sounded like the ideal gateway to the PSTN. What does the error message actually mean - is this device a non-starter or are there work-arounds or fixes to the code in progress ? Surely the device can't be as "broken" as the message - or am I just being too hopeful ? Regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, November 04, 2009 6:23 PM Subject: Re: [Freeswitch-users] Gateway Error On Wed, Nov 4, 2009 at 10:03 AM, Dave Stevenson wrote: Hi, I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway and am seeing the following error :- "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sigura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who know what will happen.." Having just bought the Gateway specifically for FS, that was a bit of a "rude awakening" ! Does anyone know of a fix in the pipeline, or am I sc***ed already ? The cynical among us will say that you were hosed the moment you paid for a Linksys device. :) It's very sad but the FS devs find this kind of thing all the time. They've literally got all sorts of checks in the code to make sure that devices aren't saying one thing and doing something else. Cisco is not the only one to do stupid things like this. In any case, just be aware of it. If you want suggestions then list to the others here who can offer their experiences with various devices they have in production. -MC -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/359364bb/attachment.html From info at daccii.it Wed Nov 4 11:55:39 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Wed, 04 Nov 2009 20:55:39 +0100 Subject: [Freeswitch-users] Question about set/export applications Message-ID: <4AF1DC3B.6070607@daccii.it> Hi to all, i'm trying to setup a simple after hours ivr, without using lua/javascript, but only xml. What i do is to catch weekdays, set some vars, catch working hours based on weekdays, and, in the end, catch if it is working time or not. If not, just set another var. Actually the code is really bad, i'll reorganize it later, the biggest problem is that i can't read setted variables! This code --- --- should set variable working_day_a to true, but if, in the following extension, i check ${working_day_a} field i get it empty. From logs --- Dialplan: OpenZAP/1:1/03 Action set(working_day_a=true) . . . Dialplan: OpenZAP/1:1/03 Regex (FAIL) [working_day_a_hours] ${working_day_a}() =~ /^true$/ break=on-false --- Here extensions http://pastebin.freeswitch.org/10977 while here relevant parts of log http://pastebin.freeswitch.org/10978 Thank for your help! Best Regards, Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/7f0712ee/attachment.vcf From info at daccii.it Wed Nov 4 12:04:36 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Wed, 04 Nov 2009 21:04:36 +0100 Subject: [Freeswitch-users] Question about set/export applications In-Reply-To: <4AF1DC3B.6070607@daccii.it> References: <4AF1DC3B.6070607@daccii.it> Message-ID: <4AF1DE54.4050300@daccii.it> Ehm, sorry, i just fixed it using inline attribute applying it on action/application/set tags: i missed it on dialplan wiki page. Sorry for the mail Albano Daniele Salvatore - Lavoro ha scritto: > Hi to all, > > i'm trying to setup a simple after hours ivr, without using > lua/javascript, but only xml. > > What i do is to catch weekdays, set some vars, catch working hours based > on weekdays, and, in the end, catch if it is working time or not. If > not, just set another var. > > Actually the code is really bad, i'll reorganize it later, the biggest > problem is that i can't read setted variables! > > . > . > . -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/a750e8bb/attachment.vcf From larclap at yahoo.com Wed Nov 4 12:20:29 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 4 Nov 2009 12:20:29 -0800 Subject: [Freeswitch-users] Copy voicemail greeting Message-ID: <011501ca5d8c$415fc340$c41f49c0$@com> Is it possible to copy an existing wav greeting from one extension to another? I think something has to be added to db/voicemail_default.db, but it's not a text file. Is it just easier to re-record the message from the 2nd extension? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/9c68c2ea/attachment.html From brian at freeswitch.org Wed Nov 4 12:24:32 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Nov 2009 14:24:32 -0600 Subject: [Freeswitch-users] Copy voicemail greeting In-Reply-To: <011501ca5d8c$415fc340$c41f49c0$@com> References: <011501ca5d8c$415fc340$c41f49c0$@com> Message-ID: copy the wav file and insert the record. /b On Nov 4, 2009, at 2:20 PM, Lars Zeb wrote: > Is it possible to copy an existing wav greeting from one extension > to another? I think something has to be added to db/ > voicemail_default.db, but it?s not a text file. > > Is it just easier to re-record the message from the 2nd extension? > > Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/6eb72a1f/attachment-0001.html From kristian.kielhofner at gmail.com Wed Nov 4 12:57:05 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 4 Nov 2009 15:57:05 -0500 Subject: [Freeswitch-users] Gateway Error In-Reply-To: <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> <33167DE5-D670-46F0-BECA-4802B917E206@jerris.com> <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> Message-ID: <2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo On Wed, Nov 4, 2009 at 2:49 PM, Dave Stevenson wrote: > Phew ! > > Thanks Mike, I was very worried there. > > Now, if I just knew which were the "broken defaults", I'd know where to go > next :-) > > Regards > Dave > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, November 04, 2009 7:36 PM > Subject: Re: [Freeswitch-users] Gateway Error > It means you need to go change the setting from the broken defaults, thats > all. > Mike > On Nov 4, 2009, at 1:48 PM, Dave Stevenson wrote: > > Michael et al - and specifically, the FS Developers, > > this is all the more annoying given the fact that the SPA-3102 was bought > specifically to run with FreeSwitch following a recommendation here in the > UK. It?was just unwrapped this afternoon :-( > > (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). > > I am setting up a VOIP system at home, and this device sounded like the > ideal gateway to the PSTN. > > What does the error message actually mean - is this device a non-starter or > are there work-arounds or?fixes to the code in progress ? > > Surely the device can't be as "broken" as the message - or am I just being > too hopeful ? > > Regards > Dave > > > > ----- Original Message ----- > From:?Michael Collins > To:?freeswitch-users at lists.freeswitch.org > Sent:?Wednesday, November 04, 2009 6:23 PM > Subject:?Re: [Freeswitch-users] Gateway Error > > > On Wed, Nov 4, 2009 at 10:03 AM, Dave > Stevenson??wrote: >> >> Hi, >> >> I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway >> and am seeing the following error :- >> >> "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they >> meant to say was 20 >> This issue has so far been identified to happen on the following broken >> platforms/devices: >> Linksys/Sigura aka Cisco >> ShoreTel >> Sonus/L3 >> We will try to fix it but some of the devices on this list are so broken >> who know what will happen.." >> >> Having just bought the Gateway specifically for FS, that was a bit of a >> "rude awakening" ! >> >> Does anyone know of a fix in the pipeline, or am I sc***ed already ? > > The cynical among us will say that you were hosed the moment you paid for a > Linksys device. :) It's very sad but the FS devs find this kind of thing all > the time. They've literally got all sorts of checks in the code to make sure > that devices aren't saying one thing and doing something else. Cisco is not > the only one to do stupid things like this. In any case, just be aware of > it. > > If you want suggestions then list to the others here who can offer their > experiences with various devices they have in production. > -MC > > > ________________________________ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ________________________________ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From carlos.talbot at gmail.com Wed Nov 4 12:58:20 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 4 Nov 2009 16:58:20 -0400 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> Message-ID: <5800526b0911041258w262f9277o9eba45b05ebbfc8c@mail.gmail.com> On Wed, Nov 4, 2009 at 1:49 PM, Dave Stevenson wrote: > Hi Carlos, > > Just one clarification then, am I right in thinking that, after I have > installed a release version of FS, if I then install one of the SVNs over > it, it will keep all configuration etc. in tact, i.e., I won't lose anything > that I've changed in the conf files etc? > > Regards > Dave > Hi Dave, Your best bet would be to keep 1.0.4 and SVN in separate installation directories since the SVN version will install default copies of the conf folder and overwrite your existing config. This is no different than your typical Windows App install file. You could just install SVN in a temp location and copy out all but the conf folder to your current 1.0.4 location. FYI, I just uploaded SVN 15355. regards, Carlos > > > > ----- Original Message ----- > *From:* Carlos Talbot > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Wednesday, November 04, 2009 2:51 PM > *Subject:* Re: [Freeswitch-users] Precompiled Windows Binaries > > > > On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson > wrote: > >> Jeff, >> >> thanks a lot for the reply. I was a little confused by the fact that the >> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >> that I might lose something. As you say though, think that I'll cross my >> fingers and try the updated release. I am running FreeSwitch on a test >> machine at the moment until the target hardware arrives - hopefully >> tomorrow, so I can afford to have a little play. >> > > I usually try to update the svn file at least once a month. I have a new > version ready that was compiled last night but am ironing out login issues > with the FS dudes for upload access. Also, the SVN snapshot now includes > binaries for 32 and 64 bit. It no longer includes flite though as the > install file was approaching 80MB in size. I will revisit this later if > others feel it important to include flite. > >> >> You mentioned FreePBX V3. I had been fumbling around trying to work out >> what >> this is and from what I've read, it seems to provide a GUI Front End for >> configuring FreeSwitch ? >> > Yes, it's still in development phase and as such not ready for production > use. > >> >> I am guessing that while it has been installed with FreeSwitch, I then >> need >> to run the FreePBX Installer to update the FreePBX/FreeSwitch >> configuration >> on my hardware ? >> >> >> When I start FreeSwitch, it does not automatically load the WAMPServer. >> >> Freeswitch and WAMPServer are independant of each other. WAMPServer is > bundled in this install for the purpose of FreePBX as MySQL, Apache and PHP > are all required components of FreePBX. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a >> web >> browser, I can see the WampServer logo and various tools such as phpinfo() >> and phpmyadmin. FreePBX is there under Your Projects. >> >> If you want to configure FreePBX you need to click on the FreePBX.url > shortcut that gets created on your desktop. > > >> When I opened this up the first time, it appeared to want to install >> FreePBX >> over FreeSwitch, I tried to abort this when it was going to overwrite some >> FreeSwitch conf files and I thought I'd better not go on until I had a >> better idea what was happening. I backed out of the FreePBX install and >> now >> I can't get the FreePBX or phpmyadmin pages up again (missing files) so it >> looks like I'm going to have to reinstall anyway. >> >> So, for next time,am I right in thinking that I should proceed with >> running >> the FreePBX install from the WAMPServer menu ? >> > > No, launch it from the shortcut as stated above. Unfortunately, at this > time there is very little user documentation on configuring FreePBX. Here is > the link to the developer's info: http://www.freepbx.org/v3 > > regards, > > Carlos > >> >> >> ----- Original Message ----- >> From: "Jeff Lenk" >> To: >> Sent: Tuesday, November 03, 2009 2:48 PM >> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >> >> >> > >> > Hi Dave, >> > >> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >> > >> > Just as you said freeswitch-1.0.4.exe is the tagged release and >> > freeswitch.exe is a newer svn snapshot. >> > >> > There should be no problems installing the new version allthough best to >> > just try and see! >> > >> > Not sure why the newest one is from October 7th. >> > >> > Jeff >> > >> > >> > Dave Stevenson wrote: >> >> >> >> Hi, >> >> >> >> I have read the Docs on the Wiki >> >> ( >> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> >> but am still not sure of what the different Windows install files are. >> >> Currently, the Windows Installer directory contains :- >> >> >> >> LATEST_SVN_15106 - 6 Bytes >> >> >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> >> >> freeswitch.exe - 32 Megabytes >> >> >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> >> September. The freeswitch.exe file is dated 7th October and think that >> it >> >> contains the minor updates since 3rd September ? >> >> >> >> Could someone who knows FreeSwitch under windows help me understand the >> >> two files please ? >> >> >> >> I chickened out of running the later exe in case it did something to >> the >> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >> with >> >> the old one already installed ? >> >> What will it actually do ? >> >> >> >> regards >> >> Dave >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > -- >> > View this message in context: >> > >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/29d2ebce/attachment.html From stevendt at primrosebank.net Wed Nov 4 13:07:27 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 21:07:27 -0000 Subject: [Freeswitch-users] Gateway Error References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com><6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> <2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> Message-ID: <1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com> Kristian, thanks very much ! After trawling the internet, I had just found the Linksys "RTP Packet Size" and worked out that it was the "ptime" parameter that FreeSwitch flagged as a problem. I had literally just changed that the very minute your post came in ! I'm off not to read that Wiki that looks like it will tell me everything else that I need to know on the SPA-3012, thanks and best regards Dave ----- Original Message ----- From: "Kristian Kielhofner" To: Sent: Wednesday, November 04, 2009 8:57 PM Subject: Re: [Freeswitch-users] Gateway Error http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo On Wed, Nov 4, 2009 at 2:49 PM, Dave Stevenson wrote: > Phew ! > > Thanks Mike, I was very worried there. > > Now, if I just knew which were the "broken defaults", I'd know where to go > next :-) > > Regards > Dave > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, November 04, 2009 7:36 PM > Subject: Re: [Freeswitch-users] Gateway Error > It means you need to go change the setting from the broken defaults, thats > all. > Mike > On Nov 4, 2009, at 1:48 PM, Dave Stevenson wrote: > > Michael et al - and specifically, the FS Developers, > > this is all the more annoying given the fact that the SPA-3102 was bought > specifically to run with FreeSwitch following a recommendation here in the > UK. It was just unwrapped this afternoon :-( > > (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). > > I am setting up a VOIP system at home, and this device sounded like the > ideal gateway to the PSTN. > > What does the error message actually mean - is this device a non-starter > or > are there work-arounds or fixes to the code in progress ? > > Surely the device can't be as "broken" as the message - or am I just being > too hopeful ? > > Regards > Dave > > > > ----- Original Message ----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, November 04, 2009 6:23 PM > Subject: Re: [Freeswitch-users] Gateway Error > > > On Wed, Nov 4, 2009 at 10:03 AM, Dave > Stevenson wrote: >> >> Hi, >> >> I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice >> Gateway >> and am seeing the following error :- >> >> "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they >> meant to say was 20 >> This issue has so far been identified to happen on the following broken >> platforms/devices: >> Linksys/Sigura aka Cisco >> ShoreTel >> Sonus/L3 >> We will try to fix it but some of the devices on this list are so broken >> who know what will happen.." >> >> Having just bought the Gateway specifically for FS, that was a bit of a >> "rude awakening" ! >> >> Does anyone know of a fix in the pipeline, or am I sc***ed already ? > > The cynical among us will say that you were hosed the moment you paid for > a > Linksys device. :) It's very sad but the FS devs find this kind of thing > all > the time. They've literally got all sorts of checks in the code to make > sure > that devices aren't saying one thing and doing something else. Cisco is > not > the only one to do stupid things like this. In any case, just be aware of > it. > > If you want suggestions then list to the others here who can offer their > experiences with various devices they have in production. > -MC > > > ________________________________ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ________________________________ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From hjqlopez at hotmail.com Wed Nov 4 13:39:28 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Wed, 4 Nov 2009 16:39:28 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: Thanks for your time, -The scenario is still the same: Always bypass media. Environment 100% NAT free :-) Call established from A to B through FS. Then...? Blind transfer from B to C (Refer-to: C) RTP should go directly between A and C. -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the REFER-to:C, BUT there is no 2-way audio.? Only RTP from C to A (due to the lack of reINVITE to A, after C answers). Please check SIP diagram here: http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html -What it's wrong with r15332 is there is not such call to C. For sure I know SIP is a protocol, may be my description was not clear but this SIP diagram speaks by itself ;-) http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html -You could check the sofia debug for r15332 here: http://pastebin.com/m6f2b3836 Best regards, Humberto > > I don't know what you are talking about anymore. > > The scenario I had tested is when a call is bridged in bypass_media=true > bridge > and you blind transfer that call back to the dialplan > > as soon as it hits the routing state it will resume media. > > > it has been confirmed to not work and confirmed to have been fixed several > time and if you are still having a problem you must have something blocking > some of your packets or something . > > You have to understand that sip is a protocol and your description is > completely non-standard. > Perhaps you should get a console trace and attach it to a jira. The trace > probably makes more sense to me. > > sofia profile internal siptrace on > console loglevel debug > > reproduce and attach the whole capture. > > > > On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > >> >> Hi, >> >> I tried r15332 and set in the sofia profile: >> >> a) bypass_media_after_bridge=true only >> b) bypass_media_after_bridge=true, param name="media-option" >> value="resume-media-on-hold"/> >> >> >> In both cases FS is hanging up the initial call (A to FS) after accepting >> the REFER to C: >> >> A <- reINVITE with FS' SDP <- FS >> A -> 200 -> FS >> A <- ACK <- FS >> A <- BYE <- FS >> >> The call to C is not even tried. >> >> I found this line is the logs that could give some idea: >> >> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] >> after sending the ACK for the reINVITE >> >> >> Regards, >> >> >> Humberto >> >>>please try r15326 >>>I think i have it working. >>> >>>I recommend for optimal results you set bypass_media_after_bridge=true >>>either as a global or in your DP in place of bypass_media=true >>> >>> >>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >> hotmail.com>wrote: >>> >>>> Hi Mike, >>>> >>>> I re-tried with trunk rev 15319 but I got almost the same behavior: >> There >>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. >> But >>>> still there is no reINVITE for A (with C's SDP) after the call from FS >> to C >>>> is established. >>>> >>>> Anyway, we decided for now to do a different implementation but if you >> want >>>> to explore more in this issue count me in ;-) >>>> >>>> >>>> Thank you very much! >>>> >>>> Humberto >> >> >> _________________________________________________________________ >> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they >> e-mail you. >> http://go.microsoft.com/?linkid=9691817 >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > _________________________________________________________________ > Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now > http://go.microsoft.com/?linkid=9691818 _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook. http://go.microsoft.com/?linkid=9691816 From jerry.richards at teotech.com Wed Nov 4 13:48:51 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 4 Nov 2009 13:48:51 -0800 Subject: [Freeswitch-users] Dial Plan Question In-Reply-To: <4AF0A457.5080702@gmail.com> References: <0A46BCC1ED4C452CAD31DF64A734C492@greyhawk.tonecommander.com> <4AF0A457.5080702@gmail.com> Message-ID: Okay. Say we want 1000 internal user extensions and want them to be configured with individual dial plans that route the call based on the extension's callgroup, time-of-day, and presence. Would be okay to create a static XML dialplan file for each extension, so calls to/from each extension would be routed uniquely based upon these parameters? This approach sounds straightforward to us. Best Regards, Jerry -----Original Message----- From: Shelby Ramsey [mailto:sicfslist at gmail.com] Sent: Tuesday, November 03, 2009 1:45 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dial Plan Question I think the real question is what are you trying to do ... for some things it's very easy to just whip up a static XML file and be done with it. For others you probably want some sort of interaction with a DB. The options here are pretty endless: -- XML curl -- handing off the call to a script call from a static dial plan (use lua if there is going to be any load) -- event_socket -- mod_lcr But ultimately I think it's what you're trying to accomplish that matters. For a PBX install I'd say static files is probably about as easy as it is going to get. For delivering a service you'd probably want interaction with a DB. I've use XML curl a lot and have even starting using direct DB queries from static dialplans using mod_memcache and memcachedb (not memcache ... persistent storage). SDR Jerry Richards wrote: > My understanding of DialPlan/CallRouting is that it can be > accomplished via static XML tags, or alternatively, via a DialPlan > Application that interfaces with the dptools module. > > Question: If my above assumption is true, how does one select one > approach over the other? What is the criteria/considerations that > would govern the decision? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > From brian at freeswitch.org Wed Nov 4 13:59:11 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Nov 2009 15:59:11 -0600 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <1E83D0AE-9587-4CC3-8FBD-073217A839A0@freeswitch.org> What phones are you using? I do this exact same scenario without a problem over and over testing with anthm. So I would like to know what phones you're using. /b On Nov 4, 2009, at 3:39 PM, Humberto Quintana wrote: > > Thanks for your time, > > -The scenario is still the same: > > Always bypass media. > Environment 100% NAT free :-) > Call established from A to B through FS. Then... > Blind transfer from B to C (Refer-to: C) > RTP should go directly between A and C. > > > -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the > REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due > to the lack of reINVITE to A, after C answers). > > Please check SIP diagram here: > > http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html > > > -What it's wrong with r15332 is there is not such call to C. For > sure I know SIP is a protocol, may be my description was not clear > but this SIP diagram speaks by itself ;-) > > http://provision.netcelerate.net/ngrep/ > blindxfer2009-11-04rev15332.html > > > -You could check the sofia debug for r15332 here: > http://pastebin.com/m6f2b3836 > > > Best regards, > > Humberto > >> >> I don't know what you are talking about anymore. >> >> The scenario I had tested is when a call is bridged in >> bypass_media=true >> bridge >> and you blind transfer that call back to the dialplan >> >> as soon as it hits the routing state it will resume media. >> >> >> it has been confirmed to not work and confirmed to have been fixed >> several >> time and if you are still having a problem you must have something >> blocking >> some of your packets or something . >> >> You have to understand that sip is a protocol and your description is >> completely non-standard. >> Perhaps you should get a console trace and attach it to a jira. The >> trace >> probably makes more sense to me. >> >> sofia profile internal siptrace on >> console loglevel debug >> >> reproduce and attach the whole capture. >> >> >> >> On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: >> >>> >>> Hi, >>> >>> I tried r15332 and set in the sofia profile: >>> >>> a) bypass_media_after_bridge=true only >>> b) bypass_media_after_bridge=true, param name="media-option" >>> value="resume-media-on-hold"/> >>> >>> >>> In both cases FS is hanging up the initial call (A to FS) after >>> accepting >>> the REFER to C: >>> >>> A <- reINVITE with FS' SDP <- FS >>> A -> 200 -> FS >>> A <- ACK <- FS >>> A <- BYE <- FS >>> >>> The call to C is not even tried. >>> >>> I found this line is the logs that could give some idea: >>> >>> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >>> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] >>> [RECOVERY_ON_TIMER_EXPIRE] >>> after sending the ACK for the reINVITE >>> >>> >>> Regards, >>> >>> >>> Humberto >>> >>>> please try r15326 >>>> I think i have it working. >>>> >>>> I recommend for optimal results you set >>>> bypass_media_after_bridge=true >>>> either as a global or in your DP in place of bypass_media=true >>>> >>>> >>>> On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >>> hotmail.com>wrote: >>>> >>>>> Hi Mike, >>>>> >>>>> I re-tried with trunk rev 15319 but I got almost the same >>>>> behavior: >>> There >>>>> is now a reINVITE (with FS' SDP) going to A when the REFER is >>>>> accepted. >>> But >>>>> still there is no reINVITE for A (with C's SDP) after the call >>>>> from FS >>> to C >>>>> is established. >>>>> >>>>> Anyway, we decided for now to do a different implementation but >>>>> if you >>> want >>>>> to explore more in this issue count me in ;-) >>>>> >>>>> >>>>> Thank you very much! >>>>> >>>>> Humberto >>> >>> >>> _________________________________________________________________ >>> Windows Live: Friends get your Flickr, Yelp, and Digg updates when >>> they >>> e-mail you. >>> http://go.microsoft.com/?linkid=9691817 >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> _________________________________________________________________ >> Ready. Set. Get a great deal on Windows 7. See fantastic deals on >> Windows 7 now >> http://go.microsoft.com/?linkid=9691818 > > _________________________________________________________________ > Windows Live: Make it easier for your friends to see what you?re up > to on Facebook. > http://go.microsoft.com/?linkid=9691816 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From oseslija at gmail.com Wed Nov 4 14:38:13 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Wed, 4 Nov 2009 23:38:13 +0100 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> Message-ID: <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> Hello, I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS. Phones are registering to CCME and FS simultaneously. So far, everything is working just fine. I think SCCP will be obsolete in the future since even Cisco is working more and more on SIP. OTOH, I really hate SIP images for 79x1 (that's why I put CCME in the first place), where as ones for 79x0 are behaving much better. Regards, Ognjen On Tue, Nov 3, 2009 at 6:52 PM, mm_202 wrote: > FS doesnt support SCCP (from what I gathered, just because no one has > bothered coding it). > > Are there other users out there has use SCCP and FS? (with some > middleware in between) > > If enough people would find a use for it, I'd be willing to actually > code it (esp if someone offered a bounty). > So, would anyone besides me want/use a SCCP endpoint in FS? > > -- mm_202. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/22dd30a9/attachment.html From msc at freeswitch.org Wed Nov 4 14:58:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Nov 2009 14:58:44 -0800 Subject: [Freeswitch-users] Dial Plan Question In-Reply-To: References: <0A46BCC1ED4C452CAD31DF64A734C492@greyhawk.tonecommander.com> <4AF0A457.5080702@gmail.com> Message-ID: <87f2f3b90911041458p789ed510h75205688c0f23e99@mail.gmail.com> On Wed, Nov 4, 2009 at 1:48 PM, Jerry Richards wrote: > > Okay. Say we want 1000 internal user extensions and want them to be > configured with individual dial plans that route the call based on the > extension's callgroup, time-of-day, and presence. Would be okay to create > a > static XML dialplan file for each extension, so calls to/from each > extension > would be routed uniquely based upon these parameters? This approach sounds > straightforward to us. > > Best Regards, > Jerry > > By "static" do you mean "doesn't change very often"? :) I don't see why you couldn't do this, although I'd be interested in knowing how easy/hard it is for you to maintain something like this. My guess is that those who have dialplans this large and complex are probably using mod_xml_curl and serving up their dialplans from another server. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/7749df54/attachment.html From stevendt at primrosebank.net Wed Nov 4 15:59:31 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 23:59:31 -0000 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com><6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com><2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> <1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com> Message-ID: <665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> I am trying to follow the configuration give in the "SPA3102 FreeSwitch HowTo". When I create the 00_spa3102.xml file, FreeSwitch won't load. If I rename the file (to, say .txt) then rename it to an xml once FreeSwitch is up and do a "reloadxml" command, I get an error flagged :- +OK [[error near line 3379]: missing >] I'm pretty sure the error must be in the 00_spa3102.xml file, but I can't see where the error might be - it looks identical to that on the Wiki page ? regards Dave From jmesquita at freeswitch.org Wed Nov 4 16:09:04 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 4 Nov 2009 22:09:04 -0200 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo In-Reply-To: <665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> <33167DE5-D670-46F0-BECA-4802B917E206@jerris.com> <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> <2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> <1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com> <665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> Message-ID: Look at this line on the freeswitch.fsxml and it will tell you exactly where the problem is. Beware that nested comments are not allowed in XML. -- JM On Wed, Nov 4, 2009 at 9:59 PM, Dave Stevenson wrote: > I am trying to follow the configuration give in the "SPA3102 FreeSwitch > HowTo". > > When I create the 00_spa3102.xml file, FreeSwitch won't load. > If I rename the file (to, say .txt) then rename it to an xml once > FreeSwitch > is up and do a "reloadxml" command, I get an error flagged :- > > +OK [[error near line 3379]: missing >] > > I'm pretty sure the error must be in the 00_spa3102.xml file, but I can't > see where the error might be - it looks identical to that on the Wiki page > ? > > regards > Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/a0f7594f/attachment-0001.html From msc at freeswitch.org Wed Nov 4 16:27:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Nov 2009 16:27:36 -0800 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> <33167DE5-D670-46F0-BECA-4802B917E206@jerris.com> <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> <2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> <1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com> <665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> Message-ID: <87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com> FYI, the file JM mentioned is actually in freeswitch/log directory. :) The file "freeswitch.fsxml" is the monster file that has all of the individual XML files included. When you got find line 3379 you'll most likely see a missing ">" char on line 3378 or near there. You'll have a few more of these before you're an expert, I guarantee it. :D It's frequently just a typo. -MC 2009/11/4 Jo?o Mesquita > Look at this line on the freeswitch.fsxml and it will tell you exactly > where the problem is. > > Beware that nested comments are not allowed in XML. > > -- JM > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/81f1315d/attachment.html From stevendt at primrosebank.net Wed Nov 4 16:49:12 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 5 Nov 2009 00:49:12 -0000 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com><2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com><1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com><665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> <87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com> Message-ID: <97FBB4B6002848BCA4F2D89F13626754@bp1.ad.bp.com> Joao & Michael, thanks a lot for the pointers to the combined xml file. Nothing as forgivable as a typo - I'm just dumb ! I had not changed the entry for destination number from the sample file :- To use the actual destination number - duh ! - I had changed the IP address, but not noticed the other required field :-( Still, with your help, I've got it now Regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Thursday, November 05, 2009 12:27 AM Subject: Re: [Freeswitch-users] SPA3102 FreeSwitch HowTo FYI, the file JM mentioned is actually in freeswitch/log directory. :) The file "freeswitch.fsxml" is the monster file that has all of the individual XML files included. When you got find line 3379 you'll most likely see a missing ">" char on line 3378 or near there. You'll have a few more of these before you're an expert, I guarantee it. :D It's frequently just a typo. -MC 2009/11/4 Jo?o Mesquita Look at this line on the freeswitch.fsxml and it will tell you exactly where the problem is. Beware that nested comments are not allowed in XML. -- JM ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/08421800/attachment.html From larclap at yahoo.com Wed Nov 4 17:10:46 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 4 Nov 2009 17:10:46 -0800 Subject: [Freeswitch-users] Copy voicemail greeting In-Reply-To: References: <011501ca5d8c$415fc340$c41f49c0$@com> Message-ID: <01a501ca5db4$cf0024b0$6d006e10$@com> What tool/GUI do you use to edit the db contents? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, November 04, 2009 12:25 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Copy voicemail greeting copy the wav file and insert the record. /b On Nov 4, 2009, at 2:20 PM, Lars Zeb wrote: Is it possible to copy an existing wav greeting from one extension to another? I think something has to be added to db/voicemail_default.db, but it's not a text file. Is it just easier to re-record the message from the 2nd extension? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/f472204e/attachment.html From anthony.minessale at gmail.com Wed Nov 4 17:12:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Nov 2009 19:12:43 -0600 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress In-Reply-To: <4AF1A525.5090909@net-vantage.com> References: <4AF1A525.5090909@net-vantage.com> Message-ID: <191c3a030911041712l7f6362fbw93d1d3695ed433e5@mail.gmail.com> how often? what platform is the machine hardware do you have a console trace by entering console loglevel debug and looking for anything odd? The only causes are Being unable to launch threads from process limits or limitations of the os The profile is restarting. maybe your ip or default gateway is changing edit /usr/local/freeswitch/conf/autoload_configs/sofia.conf.xml and uncomment On Wed, Nov 4, 2009 at 10:00 AM, RA Cohen wrote: > > FreeSWITCH Version 1.0.trunk (15321) > > -- > Roy A Cohen > Network Advantage LLC > www.net-vantage.com > 413.223.9007 option 1 > -------------------------------------------------- > "Bringing Cost-Saving, State-of-the-Art Technology > Solutions to Small and Mid-Size Organizations" > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/130a6aec/attachment.html From ujjval at simplesignal.com Wed Nov 4 18:09:01 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Wed, 4 Nov 2009 18:09:01 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> Message-ID: <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> Any ideas on the below...has anyone implemented the below: Once I have the Moderator and Participants logged on, how do I invoke the moderator previlidges, LIk esay muting everyone/someone or kicking someone out of the Conf and the like? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo Sent: Monday, November 02, 2009 12:52 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Rob: Once I have the Moderator and Participants logged on, how do I invoke the moderator previlidges, LIk esay muting everyone/someone or kicking someone out of the Conf and the like? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Friday, October 30, 2009 9:34 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hm, strange. I haven't seen that before. Can you pastebin your logs at debug level? On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > It's strange... a tcpdump tells me that there is no DTMF from my > provider when using IVR, but when I call into a TN that goes > directly into the Conference App, I see DTMF from the provider. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 7:23 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > I've never had any problem with that. Is your logging at debug level > so you can see the RECV DTFM in the log/fs_cli? Are you calling from > a SIP phone on the pbx, or via a PSTN provider? Maybe your provider > isn't passing them through. > > Make sure your logging is turned up then try something simpler, like > calling the echo application, and see if DTFM comes through. > > Rob > > On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: > >> Rob: >> >> For some reason, I don't see the DTMF appear on the fs_CLI when >> using the below configuration....so it basically timesout. >> >> UK >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, October 26, 2009 9:21 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Thx a lot Rob, reading the wiki your way or using IVR seems correct.. >> =============== >> The wiki also says that the wait-mod might be "used in conjunction >> with an IVR where the moderators are authenticated with an extra >> pass- >> code", which is what I did. I guess that's why I didn't understand >> the point of the +pin. >> ====================== >> >> I will try it out. >> >> Again thx a lot for your help. Will keep everyone posted. >> >> ________________________________________ >> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >> Sent: Friday, October 23, 2009 12:22 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I just re-tested with the pin in my dial plan: >> >> >> >> And it doesn't challenge me for the pin. I just drop right in. I >> figured this is how it was intended, since the wiki says the pin is >> set initially and only challenged in later attempts [by future >> callers]: >> >> "The first time a conference name (confname) is used, it will be >> created on demand, and the pin will be set to what ever is specified >> at that time: the pin in the data string if specified, or if not, the >> "pin" setting in the conference profile, and if that is also >> unspecified, then there is no pin protection. Any later attempt to >> join the conference must specify the same pin number, if one existed >> when it was created. " >> >> >> The wiki also says that the wait-mod might be "used in conjunction >> with an IVR where the moderators are authenticated with an extra >> pass- >> code", which is what I did. I guess that's why I didn't understand >> the point of the +pin. >> >> I'm sure there's a scenario where its used and useful, the wiki just >> doesn't explain it. >> >> Rob >> >> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >> >>> Well first off you're not defining a pine here... >>> >>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>> pin >>> number] >>> >>> That might be why its not asking for a pin. >>> >>> /b >>> >>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>> >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Wed Nov 4 21:55:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Nov 2009 23:55:38 -0600 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> Message-ID: <191c3a030911042155i64faeedmbd72d4fc5f51ca45@mail.gmail.com> but if he wants to code it we wouldn't mind it right? =p On Wed, Nov 4, 2009 at 4:38 PM, Ognjen Seslija wrote: > Hello, > > I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS. > Phones are registering to CCME and FS simultaneously. So far, everything is > working just fine. > > I think SCCP will be obsolete in the future since even Cisco is working > more and more on SIP. OTOH, I really hate SIP images for 79x1 (that's why I > put CCME in the first place), where as ones for 79x0 are behaving much > better. > > Regards, > Ognjen > > > > On Tue, Nov 3, 2009 at 6:52 PM, mm_202 wrote: > >> FS doesnt support SCCP (from what I gathered, just because no one has >> bothered coding it). >> >> Are there other users out there has use SCCP and FS? (with some >> middleware in between) >> >> If enough people would find a use for it, I'd be willing to actually >> code it (esp if someone offered a bounty). >> So, would anyone besides me want/use a SCCP endpoint in FS? >> >> -- mm_202. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/3b315101/attachment.html From ahmedmunir007 at gmail.com Wed Nov 4 22:22:08 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Thu, 5 Nov 2009 11:22:08 +0500 Subject: [Freeswitch-users] Calling more than 1 variable in condition Message-ID: Hi, In my dial plan I've created a variable named SIP_CALL, PSTN_CALL. If SIP_CALL = true, it dials out to sip call, when PSTN_CALL=true, it dials out to landline call, as I provide sample below; The problem I'm facing is how can I apply condition when I've to call more than 1 variables? Like if there are no records in SIP numbering plan table and PSTN numbering plan table so it get the digits and dial out the to carrier (how to apply AND operation in condition?) i.e. Kindly advise for this issue. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/78020414/attachment.html From tculjaga at gmail.com Wed Nov 4 23:28:48 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 5 Nov 2009 08:28:48 +0100 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <191c3a030911042155i64faeedmbd72d4fc5f51ca45@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> <191c3a030911042155i64faeedmbd72d4fc5f51ca45@mail.gmail.com> Message-ID: <65d96fc80911042328g5c25ed51o2aa90f2618ba9638@mail.gmail.com> On Thu, Nov 5, 2009 at 6:55 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > but if he wants to code it we wouldn't mind it right? =p > > > > On Wed, Nov 4, 2009 at 4:38 PM, Ognjen Seslija wrote: > >> Hello, >> >> I would just say, a mod_skinny is more than welcome and i will the 1st one willing to use it. T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/04d03ef1/attachment.html From math.parent at gmail.com Wed Nov 4 23:33:20 2009 From: math.parent at gmail.com (Mathieu Parent) Date: Thu, 5 Nov 2009 08:33:20 +0100 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> Message-ID: <960738410911042333x1e49719fu34f353bf015c9d95@mail.gmail.com> Hello, On Wed, Nov 4, 2009 at 11:38 PM, Ognjen Seslija wrote: > Hello, > > I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS. > Phones are registering to CCME and FS simultaneously. So far, everything is > working just fine. Great. I'm interrested by this simultaneous registering. Do you havve some docs? Mathieu Parent From d_hound at ymail.com Wed Nov 4 23:44:44 2009 From: d_hound at ymail.com (Hound Dog) Date: Wed, 4 Nov 2009 23:44:44 -0800 (PST) Subject: [Freeswitch-users] problem with failover routes for LCR A-Z scenario Message-ID: <727374.16142.qm@web111917.mail.gq1.yahoo.com> I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) it came out a little long , sorry for that :) I found that it difficult/impossible to make LCR use only carriers that I choose scenario is as follows , taking the UK as example for a destination ( prices are not real , just an example ) I have 2 carriers offering routes to the UK , landline and mobile my buying prices Destination carrier1 Price carrier2 price 44 (all UK) $0.01 $0.01 447 (UK mobile) $0.15 $0.19 my selling prices Destination price 44 (all UK) $0.015 447 (UKmobile) $0.17 so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable for UK mobile I can ** only ** make a profit if I use carrier 1 ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) translating it to MOD_LCR information digits rate carrier_id ( other columns ignored ) 44 0.01 1 44 0.01 2 447 0.015 1 this looks good : 44 prefix will be shared between carrier 1 and 2 447 prefix will only go to carreir 1 so it fits perfectly - BUT testing this I get - API CALL [lcr(447965404547)] output: | Digit Match | Carrier | Rate | Codec | CID Regexp | Dialstring | | 447 | carr1 | 0.15 | G711 | | [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.1 | | 44 | carr2 | 0.01 | G711 | | [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.2 | Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) - it all makes sense ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail so far I didnt find a solution for that , so if there is one I love bo pointed there I did think it over a little and came up with 2 options that could be used , and I am also planning to code them and propose patch to maintainers , I would love to get comments on those ( in case there are no existing solution ) option 1 - setting some routes as last option , add another param to the LCR table called last_route , when hitting a route with last_route=1, stop processing additional routes and return your routing decision so far so in our case the route entry with 44 to carrier1 will have last_route=1 , the other 44 routes will have last_route=0 to allow for failover option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. in other words dont failover from a finer route to a wider route. it will need to be a global option and I will be quite simple to use, but will require entering mutiple entries of the same length prefix for each carrier you would like to use its intutive and relatively simple to manage , but requires more lcr entries to get you where you want From jbarou at sqli.com Thu Nov 5 00:40:20 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Thu, 5 Nov 2009 09:40:20 +0100 Subject: [Freeswitch-users] Question about jingle_profiles In-Reply-To: <507898380911041145u431865f8uc8877fce3c2e3778@mail.gmail.com> References: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> <507898380911041145u431865f8uc8877fce3c2e3778@mail.gmail.com> Message-ID: <8048ff7f0911050040n791b59efp33dcc4a4236c71ca@mail.gmail.com> Hi, In the dialplan I have the extension "Local_extension" with "john" and it's working when I call john from the account 1000 with softphone. When I try to make a call from Gtalk to FS I have that in the console : 09-11-05 09:25:58.370659 [DEBUG] switch_rtp.c:2780 Activate VAD codec PCMU 20ms 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1184 (DingaLing/new) State Change CS_INIT -> CS_ROUTING 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal DingaLing/new [BREAK] 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:330 (DingaLing/new) State INIT going to sleep 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 (DingaLing/new) Running State Change CS_ROUTING 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 (DingaLing/new) State ROUTING 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1198 DingaLing/new CHANNEL ROUTING 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:78 DingaLing/new Standard ROUTING 2009-11-05 09:25:58.370659 [INFO] mod_dialplan_xml.c:391 Processing support.voip at gmail.com/gmail.B8861D13->john in context public Dialplan: DingaLing/new parsing [public->unloop] continue=false Dialplan: DingaLing/new Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: DingaLing/new Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: DingaLing/new parsing [public->outside_call] continue=true Dialplan: DingaLing/new Absolute Condition [outside_call] Dialplan: DingaLing/new Action set(outside_call=true) Dialplan: DingaLing/new parsing [public->call_debug] continue=true Dialplan: DingaLing/new Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: DingaLing/new parsing [public->public_extensions] continue=false Dialplan: DingaLing/new Regex (FAIL) [public_extensions] destination_number(john) =~ /^(10[01][0-9])$/ break=on-false Dialplan: DingaLing/new parsing [public->public_did] continue=false Dialplan: DingaLing/new Regex (FAIL) [public_did] destination_number(john) =~ /^(5551212)$/ break=on-false 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:114 (DingaLing/new) State Change CS_ROUTING -> CS_EXECUTE 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal DingaLing/new [BREAK] 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 (DingaLing/new) State ROUTING going to sleep 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 (DingaLing/new) Running State Change CS_EXECUTE 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:340 (DingaLing/new) State EXECUTE 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1215 DingaLing/new CHANNEL EXECUTE 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:151 DingaLing/new Standard EXECUTE EXECUTE DingaLing/new set(outside_call=true) 2009-11-05 09:25:58.381289 [DEBUG] mod_dptools.c:752 DingaLing/new SET [outside_call]=[true] 2009-11-05 09:25:58.381289 [NOTICE] switch_core_state_machine.c:179 Hangup DingaLing/new [CS_EXECUTE] [NORMAL_CLEARING] 2009-11-05 09:25:58.381289 [DEBUG] switch_channel.c:1837 Send signal DingaLing/new [KILL] 2009-11-05 09:25:58.381289 [DEBUG] libdingaling.c:298 Destroyed Session c1722311748 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.381289 [DEBUG] switch_core_session.c:969 Send signal DingaLing/new [BREAK] 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 (DingaLing/new) State HANGUP 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1293 DingaLing/new CHANNEL HANGUP 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:46 DingaLing/new Standard HANGUP, cause: NORMAL_CLEARING 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 (DingaLing/new) State HANGUP going to sleep 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:340 (DingaLing/new) State EXECUTE going to sleep 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 (DingaLing/new) Running State Change CS_HANGUP 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:454 handler already called, skipping state handler. 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:325 (DingaLing/new) State Change CS_HANGUP -> CS_REPORTING 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal DingaLing/new [BREAK] 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 (DingaLing/new) Running State Change CS_REPORTING 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 (DingaLing/new) State REPORTING 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:53 DingaLing/new Standard REPORTING, cause: NORMAL_CLEARING 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 (DingaLing/new) State REPORTING going to sleep 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:319 (DingaLing/new) State Change CS_REPORTING -> CS_DESTROY 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal DingaLing/new [BREAK] 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:1106 Session 1 (DingaLing/new) Locked, Waiting on external entities 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1124 Session 1 (DingaLing/new) Ended 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1126 Close Channel DingaLing/new [CS_DESTROY] 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:413 (DingaLing/new) Running State Change CS_DESTROY 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 (DingaLing/new) State DESTROY 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1231 NUKE RTP 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:60 DingaLing/new Standard DESTROY 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 (DingaLing/new) State DESTROY going to sleep 2009-11-05 09:25:58.459764 [DEBUG] libdingaling.c:1389 Processing 3 packets in retry queue Thanks 2009/11/4 Chris Chen > you have to define the extension "john" or "bob" or whatever number you > want in the dialplan for the context "public". > > Just follow your jingle profile you define. Simple, no other tricks. > > Thanks, > Chris > > On Wed, Nov 4, 2009 at 11:56 AM, Jonathan Barou wrote: > >> Hi everybody, >> >> I actually working on mod_dingaling (gtalk). I can make call from FS to >> Gtalk, and from Gtalk to FS. >> But I have a problem, in jingle_profile I have a file like this : >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> here when I put an user account like >> john or bob its doesn't work whereas I put something like 1000 or 8400 it >> works. >> >> When I tried to put a real phone number It doesn't work too (I have a >> gateway with my PBX). >> >> Somebody know, why it doesn't work with name and work with number ? >> >> Thanks. >> >> >> -- >> Jonathan BAROU >> SQLI LYON - CRCI >> 0472405368 >> jbarou at sqli.com >> lyon.crci at sqli.com >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jonathan BAROU SQLI LYON - CRCI 0472405368 jbarou at sqli.com lyon.crci at sqli.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/fc933f09/attachment-0001.html From oseslija at gmail.com Thu Nov 5 01:33:02 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Thu, 5 Nov 2009 10:33:02 +0100 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <960738410911042333x1e49719fu34f353bf015c9d95@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> <960738410911042333x1e49719fu34f353bf015c9d95@mail.gmail.com> Message-ID: <4468a6770911050133s1f3f1596n1b0f3e17b291080a@mail.gmail.com> Hello, you need to enable Both Reg option for E.164 Registration for extensions (you can do that via CCME's web interface in Configure->Extensions->Extension ). This will make the phone register to SCCP server and Cisco sending SIP REGISTERs to the proxy/trunk configured in the same time phone regs to SCCP. This is valid for subsequent registrations, also. Regards, Ognjen On Thu, Nov 5, 2009 at 8:33 AM, Mathieu Parent wrote: > Hello, > > > On Wed, Nov 4, 2009 at 11:38 PM, Ognjen Seslija > wrote: > > Hello, > > > > I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS. > > Phones are registering to CCME and FS simultaneously. So far, everything > is > > working just fine. > > Great. I'm interrested by this simultaneous registering. Do you havve some > docs? > > Mathieu Parent > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/a13f86ac/attachment.html From oseslija at gmail.com Thu Nov 5 01:27:29 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Thu, 5 Nov 2009 10:27:29 +0100 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <191c3a030911042155i64faeedmbd72d4fc5f51ca45@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> <191c3a030911042155i64faeedmbd72d4fc5f51ca45@mail.gmail.com> Message-ID: <4468a6770911050127p69f8b633i45e72ab9741b3b2c@mail.gmail.com> Exactly right, Tony. Regards, Ognjen On Thu, Nov 5, 2009 at 6:55 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > but if he wants to code it we wouldn't mind it right? =p > > > > On Wed, Nov 4, 2009 at 4:38 PM, Ognjen Seslija wrote: > >> Hello, >> >> I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS. >> Phones are registering to CCME and FS simultaneously. So far, everything is >> working just fine. >> >> I think SCCP will be obsolete in the future since even Cisco is working >> more and more on SIP. OTOH, I really hate SIP images for 79x1 (that's why I >> put CCME in the first place), where as ones for 79x0 are behaving much >> better. >> >> Regards, >> Ognjen >> >> >> >> On Tue, Nov 3, 2009 at 6:52 PM, mm_202 wrote: >> >>> FS doesnt support SCCP (from what I gathered, just because no one has >>> bothered coding it). >>> >>> Are there other users out there has use SCCP and FS? (with some >>> middleware in between) >>> >>> If enough people would find a use for it, I'd be willing to actually >>> code it (esp if someone offered a bounty). >>> So, would anyone besides me want/use a SCCP endpoint in FS? >>> >>> -- mm_202. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/dce49285/attachment.html From rupa at rupa.com Thu Nov 5 05:17:58 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 5 Nov 2009 07:17:58 -0600 Subject: [Freeswitch-users] problem with failover routes for LCR A-Z scenario In-Reply-To: <727374.16142.qm@web111917.mail.gq1.yahoo.com> References: <727374.16142.qm@web111917.mail.gq1.yahoo.com> Message-ID: Now that user rates are supported in mod_lcr, how about an option that says to drop the route if the user_rate is < rate ? This 1) requires you to use custom sql and 2) be able to represent your user rates in that sql (join to user rate table perhaps?) On Thu, Nov 5, 2009 at 1:44 AM, Hound Dog wrote: > I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) > > it came out a little long , sorry for that :) > > > I found that it difficult/impossible to make LCR use only carriers that I choose > > scenario is as follows , taking the UK as example for a destination ?( prices are not real , just an example ) > > I have 2 carriers offering routes to the UK , landline and mobile > > my buying prices > > Destination ? ? ? carrier1 Price ? ?carrier2 price > 44 ?(all UK) ? ? ?$0.01 ? ? ? ? ? ? $0.01 > 447 (UK mobile) ? $0.15 ? ? ? ? ? ? $0.19 > > my selling prices > > Destination ? ? ? ? price > 44 ?(all UK) ? ? ? ?$0.015 > 447 (UKmobile) ? ? ?$0.17 > > so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable > > for UK mobile I can ** only ** make a profit if I use carrier 1 ? ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) > > > translating it to MOD_LCR information > > digits ? ? rate ? ? ? ?carrier_id ? ? ? ( other columns ignored ) > 44 ? ? ? ? 0.01 ? ? ? ?1 > 44 ? ? ? ? 0.01 ? ? ? ?2 > 447 ? ? ? ?0.015 ? ? ? 1 > > this looks good : > ? ? 44 prefix will be shared between carrier 1 and 2 > ? ? 447 prefix will only go to carreir 1 > > so it fits perfectly - BUT > > testing this I get - > > API CALL [lcr(447965404547)] output: > ?| Digit Match | Carrier | Rate ? ? | Codec | CID Regexp | Dialstring | > ?| 447 ? ? ? ? | carr1 ? | 0.15 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.1 | > ?| 44 ? ? ? ? ?| carr2 ? | 0.01 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.2 | > > Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) ?- it all makes sense > > > ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail > > > so far I didnt find a solution for that , so if there is one I love bo pointed there > > > > > > > I did think it over a little and came up with 2 options that could be used , > ?and I am also planning to code them and propose patch to maintainers , > ?I would love to get comments on those ( in case there are no existing solution ) > > > option 1 - setting some routes as last option , add another param to the LCR table called ?last_route , > ?when hitting a route with last_route=1, ?stop processing additional routes and return your routing decision so far > ?so in our case the route entry with 44 to carrier1 will have last_route=1 ?, the other 44 routes will have last_route=0 to allow for failover > > option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. > in other words dont failover from a finer route to a wider route. > it will need to be a global option and I will be quite simple to use, > ?but will require entering mutiple entries of the same length prefix for each carrier you would like to use > ?its intutive and relatively simple to manage , but requires more lcr entries to get you where you want > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From testeador01 at gmail.com Thu Nov 5 05:18:48 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 5 Nov 2009 08:18:48 -0500 Subject: [Freeswitch-users] Question about jingle_profiles In-Reply-To: <8048ff7f0911050040n791b59efp33dcc4a4236c71ca@mail.gmail.com> References: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> <507898380911041145u431865f8uc8877fce3c2e3778@mail.gmail.com> <8048ff7f0911050040n791b59efp33dcc4a4236c71ca@mail.gmail.com> Message-ID: Hello :) if you look at this line 2009-11-05 09:25:58.370659 [INFO] mod_dialplan_xml.c:391 Processing > support.voip at gmail.com/gmail.B8861D13->john in context public you will see that it is looking for john in the public extensions (freeswitch/conf/dialplan/public.xml). The reason why it finds 1000 is because of this portion in public.xml: You need to create a new extension on the public context for "john" or "bob" or whatever other name you want to be able to contact from the public context. I hope this answers your question. 2009/11/5 Jonathan Barou > Hi, > > In the dialplan I have the extension "Local_extension" with "john" and it's > working when I call john from the account 1000 with softphone. > > When I try to make a call from Gtalk to FS I have that in the console : > > > > 09-11-05 09:25:58.370659 [DEBUG] switch_rtp.c:2780 Activate VAD codec PCMU > 20ms > > 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1184 (DingaLing/new) > State Change CS_INIT -> CS_ROUTING > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal > DingaLing/new [BREAK] > > 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:330 > (DingaLing/new) State INIT going to sleep > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 > (DingaLing/new) Running State Change CS_ROUTING > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 > (DingaLing/new) State ROUTING > > 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1198 DingaLing/new > CHANNEL ROUTING > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:78 > DingaLing/new Standard ROUTING > > 2009-11-05 09:25:58.370659 [INFO] mod_dialplan_xml.c:391 Processing > support.voip at gmail.com/gmail.B8861D13->john in context public > > Dialplan: DingaLing/new parsing [public->unloop] continue=false > > Dialplan: DingaLing/new Regex (PASS) [unloop] ${unroll_loops}(true) =~ > /^true$/ break=on-false > > Dialplan: DingaLing/new Regex (FAIL) [unloop] ${sip_looped_call}() =~ > /^true$/ break=on-false > > Dialplan: DingaLing/new parsing [public->outside_call] continue=true > > Dialplan: DingaLing/new Absolute Condition [outside_call] > > Dialplan: DingaLing/new Action set(outside_call=true) > > Dialplan: DingaLing/new parsing [public->call_debug] continue=true > > Dialplan: DingaLing/new Regex (FAIL) [call_debug] ${call_debug}(false) =~ > /^true$/ break=never > > Dialplan: DingaLing/new parsing [public->public_extensions] continue=false > > Dialplan: DingaLing/new Regex (FAIL) [public_extensions] > destination_number(john) =~ /^(10[01][0-9])$/ break=on-false > > Dialplan: DingaLing/new parsing [public->public_did] continue=false > > Dialplan: DingaLing/new Regex (FAIL) [public_did] destination_number(john) > =~ /^(5551212)$/ break=on-false > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:114 > (DingaLing/new) State Change CS_ROUTING -> CS_EXECUTE > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal > DingaLing/new [BREAK] > > 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 > (DingaLing/new) State ROUTING going to sleep > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 > (DingaLing/new) Running State Change CS_EXECUTE > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:340 > (DingaLing/new) State EXECUTE > > 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1215 DingaLing/new > CHANNEL EXECUTE > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:151 > DingaLing/new Standard EXECUTE > > EXECUTE DingaLing/new set(outside_call=true) > > 2009-11-05 09:25:58.381289 [DEBUG] mod_dptools.c:752 DingaLing/new SET > [outside_call]=[true] > > 2009-11-05 09:25:58.381289 [NOTICE] switch_core_state_machine.c:179 Hangup > DingaLing/new [CS_EXECUTE] [NORMAL_CLEARING] > > 2009-11-05 09:25:58.381289 [DEBUG] switch_channel.c:1837 Send signal > DingaLing/new [KILL] > > 2009-11-05 09:25:58.381289 [DEBUG] libdingaling.c:298 Destroyed Session > c1722311748 > > > 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.381289 [DEBUG] switch_core_session.c:969 Send signal > DingaLing/new [BREAK] > > 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 > (DingaLing/new) State HANGUP > > 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1293 DingaLing/new > CHANNEL HANGUP > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:46 > DingaLing/new Standard HANGUP, cause: NORMAL_CLEARING > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 > (DingaLing/new) State HANGUP going to sleep > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:340 > (DingaLing/new) State EXECUTE going to sleep > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 > (DingaLing/new) Running State Change CS_HANGUP > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:454 handler > already called, skipping state handler. > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:325 > (DingaLing/new) State Change CS_HANGUP -> CS_REPORTING > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal > DingaLing/new [BREAK] > > 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 > (DingaLing/new) Running State Change CS_REPORTING > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 > (DingaLing/new) State REPORTING > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:53 > DingaLing/new Standard REPORTING, cause: NORMAL_CLEARING > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 > (DingaLing/new) State REPORTING going to sleep > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:319 > (DingaLing/new) State Change CS_REPORTING -> CS_DESTROY > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal > DingaLing/new [BREAK] > > 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:1106 Session 1 > (DingaLing/new) Locked, Waiting on external entities > > 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1124 Session 1 > (DingaLing/new) Ended > > 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1126 Close > Channel DingaLing/new [CS_DESTROY] > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:413 > (DingaLing/new) Running State Change CS_DESTROY > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 > (DingaLing/new) State DESTROY > > 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1231 NUKE RTP > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:60 > DingaLing/new Standard DESTROY > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 > (DingaLing/new) State DESTROY going to sleep > > 2009-11-05 09:25:58.459764 [DEBUG] libdingaling.c:1389 Processing 3 packets > in retry queue > > Thanks > > > > 2009/11/4 Chris Chen > > you have to define the extension "john" or "bob" or whatever number you >> want in the dialplan for the context "public". >> >> Just follow your jingle profile you define. Simple, no other tricks. >> >> Thanks, >> Chris >> >> On Wed, Nov 4, 2009 at 11:56 AM, Jonathan Barou wrote: >> >>> Hi everybody, >>> >>> I actually working on mod_dingaling (gtalk). I can make call from FS to >>> Gtalk, and from Gtalk to FS. >>> But I have a problem, in jingle_profile I have a file like this : >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> here when I put an user account like >>> john or bob its doesn't work whereas I put something like 1000 or 8400 it >>> works. >>> >>> When I tried to put a real phone number It doesn't work too (I have a >>> gateway with my PBX). >>> >>> Somebody know, why it doesn't work with name and work with number ? >>> >>> Thanks. >>> >>> >>> -- >>> Jonathan BAROU >>> SQLI LYON - CRCI >>> 0472405368 >>> jbarou at sqli.com >>> lyon.crci at sqli.com >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Jonathan BAROU > SQLI LYON - CRCI > 0472405368 > jbarou at sqli.com > lyon.crci at sqli.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/abcfb106/attachment-0001.html From nagalenoj at gmail.com Thu Nov 5 05:28:22 2009 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Thu, 5 Nov 2009 18:58:22 +0530 Subject: [Freeswitch-users] Filtering a particular event. Message-ID: Hi, I've tried to filter the events like below to filter a particular event. 1) register for all events 2) filter for one unique-id 3) filter only one/more events(ex: DTMF & CHANNEL_EXECUTE) So, I want to receive only these events for the specific unique-id. But, I am receiving other events too. I'm using perl ESL outbound. Is it possible to do like this?! -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/f2cd0c29/attachment.html From rupa at rupa.com Thu Nov 5 05:31:00 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 5 Nov 2009 07:31:00 -0600 Subject: [Freeswitch-users] problem with failover routes for LCR A-Z scenario In-Reply-To: References: <727374.16142.qm@web111917.mail.gq1.yahoo.com> Message-ID: Actually, using custom sql, you can implement the filter yourself in the where clause. No need for code changes. On Thu, Nov 5, 2009 at 7:17 AM, Rupa Schomaker wrote: > Now that user rates are supported in mod_lcr, how about an option that > says to drop the route if the user_rate is < rate ? > > This 1) requires you to use custom sql and 2) be able to represent > your user rates in that sql (join to user rate table perhaps?) > > On Thu, Nov 5, 2009 at 1:44 AM, Hound Dog wrote: >> I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) >> >> it came out a little long , sorry for that :) >> >> >> I found that it difficult/impossible to make LCR use only carriers that I choose >> >> scenario is as follows , taking the UK as example for a destination ?( prices are not real , just an example ) >> >> I have 2 carriers offering routes to the UK , landline and mobile >> >> my buying prices >> >> Destination ? ? ? carrier1 Price ? ?carrier2 price >> 44 ?(all UK) ? ? ?$0.01 ? ? ? ? ? ? $0.01 >> 447 (UK mobile) ? $0.15 ? ? ? ? ? ? $0.19 >> >> my selling prices >> >> Destination ? ? ? ? price >> 44 ?(all UK) ? ? ? ?$0.015 >> 447 (UKmobile) ? ? ?$0.17 >> >> so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable >> >> for UK mobile I can ** only ** make a profit if I use carrier 1 ? ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) >> >> >> translating it to MOD_LCR information >> >> digits ? ? rate ? ? ? ?carrier_id ? ? ? ( other columns ignored ) >> 44 ? ? ? ? 0.01 ? ? ? ?1 >> 44 ? ? ? ? 0.01 ? ? ? ?2 >> 447 ? ? ? ?0.015 ? ? ? 1 >> >> this looks good : >> ? ? 44 prefix will be shared between carrier 1 and 2 >> ? ? 447 prefix will only go to carreir 1 >> >> so it fits perfectly - BUT >> >> testing this I get - >> >> API CALL [lcr(447965404547)] output: >> ?| Digit Match | Carrier | Rate ? ? | Codec | CID Regexp | Dialstring | >> ?| 447 ? ? ? ? | carr1 ? | 0.15 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.1 | >> ?| 44 ? ? ? ? ?| carr2 ? | 0.01 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.2 | >> >> Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) ?- it all makes sense >> >> >> ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail >> >> >> so far I didnt find a solution for that , so if there is one I love bo pointed there >> >> >> >> >> >> >> I did think it over a little and came up with 2 options that could be used , >> ?and I am also planning to code them and propose patch to maintainers , >> ?I would love to get comments on those ( in case there are no existing solution ) >> >> >> option 1 - setting some routes as last option , add another param to the LCR table called ?last_route , >> ?when hitting a route with last_route=1, ?stop processing additional routes and return your routing decision so far >> ?so in our case the route entry with 44 to carrier1 will have last_route=1 ?, the other 44 routes will have last_route=0 to allow for failover >> >> option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. >> in other words dont failover from a finer route to a wider route. >> it will need to be a global option and I will be quite simple to use, >> ?but will require entering mutiple entries of the same length prefix for each carrier you would like to use >> ?its intutive and relatively simple to manage , but requires more lcr entries to get you where you want >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -Rupa > -- -Rupa From testeador01 at gmail.com Thu Nov 5 05:35:23 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 5 Nov 2009 08:35:23 -0500 Subject: [Freeswitch-users] Copy voicemail greeting In-Reply-To: <01a501ca5db4$cf0024b0$6d006e10$@com> References: <011501ca5d8c$415fc340$c41f49c0$@com> <01a501ca5db4$cf0024b0$6d006e10$@com> Message-ID: Use sqlite3: http://souptonuts.sourceforge.net/readme_sqlite_tutorial.html [?] 2009/11/4 Lars Zeb > What tool/GUI do you use to edit the db contents? > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Wednesday, November 04, 2009 12:25 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Copy voicemail greeting > > > > copy the wav file and insert the record. > > > > /b > > > > On Nov 4, 2009, at 2:20 PM, Lars Zeb wrote: > > > > Is it possible to copy an existing wav greeting from one extension to > another? I think something has to be added to db/voicemail_default.db, but > it?s not a text file. > > > > Is it just easier to re-record the message from the 2nd extension? > > > > Thanks, Lars > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/ccc317b9/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 96 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/ccc317b9/attachment.gif From jbarou at sqli.com Thu Nov 5 05:37:31 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Thu, 5 Nov 2009 14:37:31 +0100 Subject: [Freeswitch-users] Question about jingle_profiles In-Reply-To: References: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> <507898380911041145u431865f8uc8877fce3c2e3778@mail.gmail.com> <8048ff7f0911050040n791b59efp33dcc4a4236c71ca@mail.gmail.com> Message-ID: <8048ff7f0911050537n2211fbf8n8f09d49168eb21f9@mail.gmail.com> Wonderful, thank you very much Milena. 2009/11/5 Milena > Hello :) > if you look at this line > > 2009-11-05 09:25:58.370659 [INFO] mod_dialplan_xml.c:391 Processing >> support.voip at gmail.com/gmail.B8861D13->john in context public > > > you will see that it is looking for john in the public extensions > (freeswitch/conf/dialplan/public.xml). > > The reason why it finds 1000 is because of this portion in public.xml: > > > > > > > You need to create a new extension on the public context for "john" or > "bob" or whatever other name you want to be able to contact from the public > context. > > I hope this answers your question. > > > 2009/11/5 Jonathan Barou > > Hi, >> >> In the dialplan I have the extension "Local_extension" with "john" and >> it's working when I call john from the account 1000 with softphone. >> >> When I try to make a call from Gtalk to FS I have that in the console : >> >> >> >> 09-11-05 09:25:58.370659 [DEBUG] switch_rtp.c:2780 Activate VAD codec PCMU >> 20ms >> >> 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1184 (DingaLing/new) >> State Change CS_INIT -> CS_ROUTING >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal >> DingaLing/new [BREAK] >> >> 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:330 >> (DingaLing/new) State INIT going to sleep >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 >> (DingaLing/new) Running State Change CS_ROUTING >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 >> (DingaLing/new) State ROUTING >> >> 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1198 DingaLing/new >> CHANNEL ROUTING >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:78 >> DingaLing/new Standard ROUTING >> >> 2009-11-05 09:25:58.370659 [INFO] mod_dialplan_xml.c:391 Processing >> support.voip at gmail.com/gmail.B8861D13->john in context public >> >> Dialplan: DingaLing/new parsing [public->unloop] continue=false >> >> Dialplan: DingaLing/new Regex (PASS) [unloop] ${unroll_loops}(true) =~ >> /^true$/ break=on-false >> >> Dialplan: DingaLing/new Regex (FAIL) [unloop] ${sip_looped_call}() =~ >> /^true$/ break=on-false >> >> Dialplan: DingaLing/new parsing [public->outside_call] continue=true >> >> Dialplan: DingaLing/new Absolute Condition [outside_call] >> >> Dialplan: DingaLing/new Action set(outside_call=true) >> >> Dialplan: DingaLing/new parsing [public->call_debug] continue=true >> >> Dialplan: DingaLing/new Regex (FAIL) [call_debug] ${call_debug}(false) =~ >> /^true$/ break=never >> >> Dialplan: DingaLing/new parsing [public->public_extensions] continue=false >> >> Dialplan: DingaLing/new Regex (FAIL) [public_extensions] >> destination_number(john) =~ /^(10[01][0-9])$/ break=on-false >> >> Dialplan: DingaLing/new parsing [public->public_did] continue=false >> >> Dialplan: DingaLing/new Regex (FAIL) [public_did] destination_number(john) >> =~ /^(5551212)$/ break=on-false >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:114 >> (DingaLing/new) State Change CS_ROUTING -> CS_EXECUTE >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal >> DingaLing/new [BREAK] >> >> 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 >> (DingaLing/new) State ROUTING going to sleep >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 >> (DingaLing/new) Running State Change CS_EXECUTE >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:340 >> (DingaLing/new) State EXECUTE >> >> 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1215 DingaLing/new >> CHANNEL EXECUTE >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:151 >> DingaLing/new Standard EXECUTE >> >> EXECUTE DingaLing/new set(outside_call=true) >> >> 2009-11-05 09:25:58.381289 [DEBUG] mod_dptools.c:752 DingaLing/new SET >> [outside_call]=[true] >> >> 2009-11-05 09:25:58.381289 [NOTICE] switch_core_state_machine.c:179 Hangup >> DingaLing/new [CS_EXECUTE] [NORMAL_CLEARING] >> >> 2009-11-05 09:25:58.381289 [DEBUG] switch_channel.c:1837 Send signal >> DingaLing/new [KILL] >> >> 2009-11-05 09:25:58.381289 [DEBUG] libdingaling.c:298 Destroyed Session >> c1722311748 >> >> >> 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.381289 [DEBUG] switch_core_session.c:969 Send signal >> DingaLing/new [BREAK] >> >> 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 >> (DingaLing/new) State HANGUP >> >> 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1293 DingaLing/new >> CHANNEL HANGUP >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:46 >> DingaLing/new Standard HANGUP, cause: NORMAL_CLEARING >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 >> (DingaLing/new) State HANGUP going to sleep >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:340 >> (DingaLing/new) State EXECUTE going to sleep >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 >> (DingaLing/new) Running State Change CS_HANGUP >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:454 handler >> already called, skipping state handler. >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:325 >> (DingaLing/new) State Change CS_HANGUP -> CS_REPORTING >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal >> DingaLing/new [BREAK] >> >> 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 >> (DingaLing/new) Running State Change CS_REPORTING >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 >> (DingaLing/new) State REPORTING >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:53 >> DingaLing/new Standard REPORTING, cause: NORMAL_CLEARING >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 >> (DingaLing/new) State REPORTING going to sleep >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:319 >> (DingaLing/new) State Change CS_REPORTING -> CS_DESTROY >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal >> DingaLing/new [BREAK] >> >> 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:1106 Session 1 >> (DingaLing/new) Locked, Waiting on external entities >> >> 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1124 Session 1 >> (DingaLing/new) Ended >> >> 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1126 Close >> Channel DingaLing/new [CS_DESTROY] >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:413 >> (DingaLing/new) Running State Change CS_DESTROY >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 >> (DingaLing/new) State DESTROY >> >> 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1231 NUKE RTP >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:60 >> DingaLing/new Standard DESTROY >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 >> (DingaLing/new) State DESTROY going to sleep >> >> 2009-11-05 09:25:58.459764 [DEBUG] libdingaling.c:1389 Processing 3 >> packets in retry queue >> >> Thanks >> >> >> >> 2009/11/4 Chris Chen >> >> you have to define the extension "john" or "bob" or whatever number you >>> want in the dialplan for the context "public". >>> >>> Just follow your jingle profile you define. Simple, no other tricks. >>> >>> Thanks, >>> Chris >>> >>> On Wed, Nov 4, 2009 at 11:56 AM, Jonathan Barou wrote: >>> >>>> Hi everybody, >>>> >>>> I actually working on mod_dingaling (gtalk). I can make call from FS to >>>> Gtalk, and from Gtalk to FS. >>>> But I have a problem, in jingle_profile I have a file like this : >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> here when I put an user account like >>>> john or bob its doesn't work whereas I put something like 1000 or 8400 it >>>> works. >>>> >>>> When I tried to put a real phone number It doesn't work too (I have a >>>> gateway with my PBX). >>>> >>>> Somebody know, why it doesn't work with name and work with number ? >>>> >>>> Thanks. >>>> >>>> >>>> -- >>>> Jonathan BAROU >>>> SQLI LYON - CRCI >>>> 0472405368 >>>> jbarou at sqli.com >>>> lyon.crci at sqli.com >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Jonathan BAROU >> SQLI LYON - CRCI >> 0472405368 >> jbarou at sqli.com >> lyon.crci at sqli.com >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jonathan BAROU SQLI LYON - CRCI 0472405368 jbarou at sqli.com lyon.crci at sqli.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/7587c615/attachment-0001.html From anthony.minessale at gmail.com Thu Nov 5 06:13:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Nov 2009 08:13:46 -0600 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <191c3a030911050613o4a0cd40gcef5163574189561@mail.gmail.com> I did not ask you to send me a ladder diagram. I asked you to send me a console trace from FreeSWITCH using latest trunk (1.0.4 does not help me) 1) start FreeSWITCH 2) run the cli command: console loglevel debug 3) run the cli command: sofia profile internal siptrace on 4) reproduce your issue and put the trace on freeswitch pastebin http://pastebin.freeswitch.org (login and pass are stated in the auth dialog) Also please answer brian's question. What phones and/or sip devices are involved in this call. On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana wrote: > > Thanks for your time, > > -The scenario is still the same: > > Always bypass media. > Environment 100% NAT free :-) > Call established from A to B through FS. Then... > Blind transfer from B to C (Refer-to: C) > RTP should go directly between A and C. > > > -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the > REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the > lack of reINVITE to A, after C answers). > > Please check SIP diagram here: > > http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html > > > -What it's wrong with r15332 is there is not such call to C. For sure I > know SIP is a protocol, may be my description was not clear but this SIP > diagram speaks by itself ;-) > > http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html > > > -You could check the sofia debug for r15332 here: > http://pastebin.com/m6f2b3836 > > > Best regards, > > Humberto > > > > > I don't know what you are talking about anymore. > > > > The scenario I had tested is when a call is bridged in bypass_media=true > > bridge > > and you blind transfer that call back to the dialplan > > > > as soon as it hits the routing state it will resume media. > > > > > > it has been confirmed to not work and confirmed to have been fixed > several > > time and if you are still having a problem you must have something > blocking > > some of your packets or something . > > > > You have to understand that sip is a protocol and your description is > > completely non-standard. > > Perhaps you should get a console trace and attach it to a jira. The trace > > probably makes more sense to me. > > > > sofia profile internal siptrace on > > console loglevel debug > > > > reproduce and attach the whole capture. > > > > > > > > On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > > > >> > >> Hi, > >> > >> I tried r15332 and set in the sofia profile: > >> > >> a) bypass_media_after_bridge=true only > >> b) bypass_media_after_bridge=true, param name="media-option" > >> value="resume-media-on-hold"/> > >> > >> > >> In both cases FS is hanging up the initial call (A to FS) after > accepting > >> the REFER to C: > >> > >> A <- reINVITE with FS' SDP <- FS > >> A -> 200 -> FS > >> A <- ACK <- FS > >> A <- BYE <- FS > >> > >> The call to C is not even tried. > >> > >> I found this line is the logs that could give some idea: > >> > >> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup > >> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] > [RECOVERY_ON_TIMER_EXPIRE] > >> after sending the ACK for the reINVITE > >> > >> > >> Regards, > >> > >> > >> Humberto > >> > >>>please try r15326 > >>>I think i have it working. > >>> > >>>I recommend for optimal results you set bypass_media_after_bridge=true > >>>either as a global or in your DP in place of bypass_media=true > >>> > >>> > >>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana > >> hotmail.com>wrote: > >>> > >>>> Hi Mike, > >>>> > >>>> I re-tried with trunk rev 15319 but I got almost the same behavior: > >> There > >>>> is now a reINVITE (with FS' SDP) going to A when the REFER is > accepted. > >> But > >>>> still there is no reINVITE for A (with C's SDP) after the call from FS > >> to C > >>>> is established. > >>>> > >>>> Anyway, we decided for now to do a different implementation but if you > >> want > >>>> to explore more in this issue count me in ;-) > >>>> > >>>> > >>>> Thank you very much! > >>>> > >>>> Humberto > >> > >> > >> _________________________________________________________________ > >> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they > >> e-mail you. > >> http://go.microsoft.com/?linkid=9691817 > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > _________________________________________________________________ > > Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows > 7 now > > http://go.microsoft.com/?linkid=9691818 > > _________________________________________________________________ > Windows Live: Make it easier for your friends to see what you?re up to on > Facebook. > http://go.microsoft.com/?linkid=9691816 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/cfff9439/attachment.html From rob4manhere at gmail.com Thu Nov 5 06:52:05 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 5 Nov 2009 08:52:05 -0600 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> Message-ID: <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> Hi UK, From what I've done and read, the caller-controls (in conference.conf.xml) can be modified to almost anything you can think of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a caller control just for those with the moderator flag. So unless you want everyone able to mute/kick everyone then you can't do it. The wiki seems to indicate this as well: "Be aware that the caller-controls are applied across the entire conference. You cannot enter one member of the conference using caller- controls ABC and then enter a second member using caller-controls XYZ." http://wiki.freeswitch.org/wiki/Mod_conference I think this might be a limitation of mod_conference. Perhaps one of the pros can chime in if I'm off-base or there's some nifty way to accomplish this. Cheers, Rob On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > Any ideas on the below...has anyone implemented the below: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Ujjval Karihaloo > Sent: Monday, November 02, 2009 12:52 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Rob: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 9:34 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hm, strange. I haven't seen that before. Can you pastebin your logs > at debug level? > > On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > >> It's strange... a tcpdump tells me that there is no DTMF from my >> provider when using IVR, but when I call into a TN that goes >> directly into the Conference App, I see DTMF from the provider. >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 7:23 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I've never had any problem with that. Is your logging at debug level >> so you can see the RECV DTFM in the log/fs_cli? Are you calling from >> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >> isn't passing them through. >> >> Make sure your logging is turned up then try something simpler, like >> calling the echo application, and see if DTFM comes through. >> >> Rob >> >> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >> >>> Rob: >>> >>> For some reason, I don't see the DTMF appear on the fs_CLI when >>> using the below configuration....so it basically timesout. >>> >>> UK >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, October 26, 2009 9:21 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Thx a lot Rob, reading the wiki your way or using IVR seems >>> correct.. >>> =============== >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> ====================== >>> >>> I will try it out. >>> >>> Again thx a lot for your help. Will keep everyone posted. >>> >>> ________________________________________ >>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>> Sent: Friday, October 23, 2009 12:22 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I just re-tested with the pin in my dial plan: >>> >>> >>> >>> And it doesn't challenge me for the pin. I just drop right in. I >>> figured this is how it was intended, since the wiki says the pin is >>> set initially and only challenged in later attempts [by future >>> callers]: >>> >>> "The first time a conference name (confname) is used, it will be >>> created on demand, and the pin will be set to what ever is specified >>> at that time: the pin in the data string if specified, or if not, >>> the >>> "pin" setting in the conference profile, and if that is also >>> unspecified, then there is no pin protection. Any later attempt to >>> join the conference must specify the same pin number, if one existed >>> when it was created. " >>> >>> >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> >>> I'm sure there's a scenario where its used and useful, the wiki just >>> doesn't explain it. >>> >>> Rob >>> >>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>> >>>> Well first off you're not defining a pine here... >>>> >>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>> pin >>>> number] >>>> >>>> That might be why its not asking for a pin. >>>> >>>> /b >>>> >>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rob4manhere at gmail.com Thu Nov 5 06:57:08 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 5 Nov 2009 08:57:08 -0600 Subject: [Freeswitch-users] Wideband / HD phones Message-ID: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> Hey all, Looking at buying some high def phones. Any recommendations (preferably based on experience) for hardware based on product quality, standards compliance, features integration with Freeswitch, etc? Thank you! Rob Forman From brian at freeswitch.org Thu Nov 5 07:07:09 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 09:07:09 -0600 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> Message-ID: <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> Polycom ip6000's or bust! /b On Nov 5, 2009, at 8:57 AM, Rob Forman wrote: > Hey all, > > Looking at buying some high def phones. Any recommendations > (preferably based on experience) for hardware based on product > quality, standards compliance, features integration with Freeswitch, > etc? > > Thank you! > Rob Forman From qinglan_zeng at hotmail.com Thu Nov 5 07:19:37 2009 From: qinglan_zeng at hotmail.com (=?gb2312?B?tPPE4MjL?=) Date: Thu, 5 Nov 2009 15:19:37 +0000 Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: Hi All, I once meet the Skypiax load error issue and some guys infomed me that there is no configuration file for Skypiax. When I follow these intructions -> http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances I still have some difficulties unstanding this: ." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml to Debug\conf\autoload_configs." I did not find such directories in my freeswitch folder. Did not understand what "src" means, I checked the freeswitch folder and did not find such a folder named"src". There is a folder named"mod" under freeswitch while look into "mod" folder there are only some DLL files and can not find endpoints and etc. 2.You'll probably build the "Debug" version I just build this from the precompiled binaries and then launched FS. I'm not sure what I launched is in debug mode or not. If anyone can offer some help that really be appriciated. Thanks Daniel Zeng From: freeswitch-users-request at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 41, Issue 40 To: freeswitch-users at lists.freeswitch.org Date: Thu, 5 Nov 2009 06:57:44 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --??????-- From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 5 Nov 2009 08:13:46 -0600 Subject: Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media I did not ask you to send me a ladder diagram. I asked you to send me a console trace from FreeSWITCH using latest trunk (1.0.4 does not help me) 1) start FreeSWITCH 2) run the cli command: console loglevel debug 3) run the cli command: sofia profile internal siptrace on 4) reproduce your issue and put the trace on freeswitch pastebin http://pastebin.freeswitch.org (login and pass are stated in the auth dialog) Also please answer brian's question. What phones and/or sip devices are involved in this call. On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana wrote: Thanks for your time, -The scenario is still the same: Always bypass media. Environment 100% NAT free :-) Call established from A to B through FS. Then... Blind transfer from B to C (Refer-to: C) RTP should go directly between A and C. -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the lack of reINVITE to A, after C answers). Please check SIP diagram here: http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html -What it's wrong with r15332 is there is not such call to C. For sure I know SIP is a protocol, may be my description was not clear but this SIP diagram speaks by itself ;-) http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html -You could check the sofia debug for r15332 here: http://pastebin.com/m6f2b3836 Best regards, Humberto > > I don't know what you are talking about anymore. > > The scenario I had tested is when a call is bridged in bypass_media=true > bridge > and you blind transfer that call back to the dialplan > > as soon as it hits the routing state it will resume media. > > > it has been confirmed to not work and confirmed to have been fixed several > time and if you are still having a problem you must have something blocking > some of your packets or something . > > You have to understand that sip is a protocol and your description is > completely non-standard. > Perhaps you should get a console trace and attach it to a jira. The trace > probably makes more sense to me. > > sofia profile internal siptrace on > console loglevel debug > > reproduce and attach the whole capture. > > > > On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > >> >> Hi, >> >> I tried r15332 and set in the sofia profile: >> >> a) bypass_media_after_bridge=true only >> b) bypass_media_after_bridge=true, param name="media-option" >> value="resume-media-on-hold"/> >> >> >> In both cases FS is hanging up the initial call (A to FS) after accepting >> the REFER to C: >> >> A <- reINVITE with FS' SDP <- FS >> A -> 200 -> FS >> A <- ACK <- FS >> A <- BYE <- FS >> >> The call to C is not even tried. >> >> I found this line is the logs that could give some idea: >> >> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] >> after sending the ACK for the reINVITE >> >> >> Regards, >> >> >> Humberto >> >>>please try r15326 >>>I think i have it working. >>> >>>I recommend for optimal results you set bypass_media_after_bridge=true >>>either as a global or in your DP in place of bypass_media=true >>> >>> >>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >> hotmail.com>wrote: >>> >>>> Hi Mike, >>>> >>>> I re-tried with trunk rev 15319 but I got almost the same behavior: >> There >>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. >> But >>>> still there is no reINVITE for A (with C's SDP) after the call from FS >> to C >>>> is established. >>>> >>>> Anyway, we decided for now to do a different implementation but if you >> want >>>> to explore more in this issue count me in ;-) >>>> >>>> >>>> Thank you very much! >>>> >>>> Humberto >> >> >> _________________________________________________________________ >> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they >> e-mail you. >> http://go.microsoft.com/?linkid=9691817 >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > _________________________________________________________________ > Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now > http://go.microsoft.com/?linkid=9691818 _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook. http://go.microsoft.com/?linkid=9691816 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 --??????-- From: rob4manhere at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 5 Nov 2009 08:52:05 -0600 Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hi UK, From what I've done and read, the caller-controls (in conference.conf.xml) can be modified to almost anything you can think of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a caller control just for those with the moderator flag. So unless you want everyone able to mute/kick everyone then you can't do it. The wiki seems to indicate this as well: "Be aware that the caller-controls are applied across the entire conference. You cannot enter one member of the conference using caller- controls ABC and then enter a second member using caller-controls XYZ." http://wiki.freeswitch.org/wiki/Mod_conference I think this might be a limitation of mod_conference. Perhaps one of the pros can chime in if I'm off-base or there's some nifty way to accomplish this. Cheers, Rob On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > Any ideas on the below...has anyone implemented the below: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Ujjval Karihaloo > Sent: Monday, November 02, 2009 12:52 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Rob: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 9:34 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hm, strange. I haven't seen that before. Can you pastebin your logs > at debug level? > > On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > >> It's strange... a tcpdump tells me that there is no DTMF from my >> provider when using IVR, but when I call into a TN that goes >> directly into the Conference App, I see DTMF from the provider. >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 7:23 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I've never had any problem with that. Is your logging at debug level >> so you can see the RECV DTFM in the log/fs_cli? Are you calling from >> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >> isn't passing them through. >> >> Make sure your logging is turned up then try something simpler, like >> calling the echo application, and see if DTFM comes through. >> >> Rob >> >> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >> >>> Rob: >>> >>> For some reason, I don't see the DTMF appear on the fs_CLI when >>> using the below configuration....so it basically timesout. >>> >>> UK >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, October 26, 2009 9:21 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Thx a lot Rob, reading the wiki your way or using IVR seems >>> correct.. >>> =============== >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> ====================== >>> >>> I will try it out. >>> >>> Again thx a lot for your help. Will keep everyone posted. >>> >>> ________________________________________ >>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>> Sent: Friday, October 23, 2009 12:22 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I just re-tested with the pin in my dial plan: >>> >>> >>> >>> And it doesn't challenge me for the pin. I just drop right in. I >>> figured this is how it was intended, since the wiki says the pin is >>> set initially and only challenged in later attempts [by future >>> callers]: >>> >>> "The first time a conference name (confname) is used, it will be >>> created on demand, and the pin will be set to what ever is specified >>> at that time: the pin in the data string if specified, or if not, >>> the >>> "pin" setting in the conference profile, and if that is also >>> unspecified, then there is no pin protection. Any later attempt to >>> join the conference must specify the same pin number, if one existed >>> when it was created. " >>> >>> >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> >>> I'm sure there's a scenario where its used and useful, the wiki just >>> doesn't explain it. >>> >>> Rob >>> >>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>> >>>> Well first off you're not defining a pine here... >>>> >>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>> pin >>>> number] >>>> >>>> That might be why its not asking for a pin. >>>> >>>> /b >>>> >>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org --??????-- From: rob4manhere at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 5 Nov 2009 08:57:08 -0600 Subject: [Freeswitch-users] Wideband / HD phones Hey all, Looking at buying some high def phones. Any recommendations (preferably based on experience) for hardware based on product quality, standards compliance, features integration with Freeswitch, etc? Thank you! Rob Forman _________________________________________________________________ ?????????????????msn????? http://ditu.live.com/?form=TL&swm=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/0f85d097/attachment-0001.html From maciej.aniserowicz at gmail.com Thu Nov 5 07:35:51 2009 From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz) Date: Thu, 5 Nov 2009 07:35:51 -0800 (PST) Subject: [Freeswitch-users] Users hanged up for unknown reason In-Reply-To: <87f2f3b90911041016u620ca88bk4f0d6a4ceb339b4b@mail.gmail.com> References: <1257244093831-3937601.post@n2.nabble.com> <87f2f3b90911041016u620ca88bk4f0d6a4ceb339b4b@mail.gmail.com> Message-ID: <1257435351018-3952900.post@n2.nabble.com> OK, I put all of the logs in pastebin. Some more background: I use 2 instances of FS. One of them (we call it "prod") is used internally to connect to "the outside world". The other (called "gateway") is "the outside world" for dev and testing purposes. Here is the previous link again, with my commands and "prod" FS events: http://pastebin.freeswitch.org/10955 Here is a log from "prod" FS: part 1: http://pastebin.freeswitch.org/10993, part 2: http://pastebin.freeswitch.org/10994 Here is a log from "gateway" FS: part 1: http://pastebin.freeswitch.org/10995, part 2: http://pastebin.freeswitch.org/10996 Let me know if this is still not enough information, thanks. Maciej Aniserowicz mercutioviz wrote: > > On Tue, Nov 3, 2009 at 2:28 AM, Maciej Aniserowicz < > maciej.aniserowicz at gmail.com> wrote: > >> >> Hi, >> I have a strange problem. I control FS with commands sent by tcp in >> response >> to events published via tcp. I do something like: >> 1) call 1st user >> 2) call 2nd user >> 3) 1st and 2nd talk >> 4) call another user >> 5) 1st and another talk >> etc... >> >> Sometimes (quite regularly) users are hanged up (with cause >> NORMAL_CLEARING) >> even if they do not hangup manually. >> >> I pasted one such scenario in pastebin >> (http://pastebin.freeswitch.org/10955), it includes logs from commands >> sent >> by me and events received from FS. Could someone take a look and see what >> am >> I doing wrong? >> > Seeing only the events it is difficult to see what triggered them. Can you > repeat these tests and capture the debug output from the CLI? > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Users-hanged-up-for-unknown-reason-tp3937601p3952900.html Sent from the freeswitch-users mailing list archive at Nabble.com. From d_hound at ymail.com Thu Nov 5 07:46:28 2009 From: d_hound at ymail.com (Hound Dog) Date: Thu, 5 Nov 2009 07:46:28 -0800 (PST) Subject: [Freeswitch-users] problem with failover routes for LCR A-Z scenario In-Reply-To: References: <727374.16142.qm@web111917.mail.gq1.yahoo.com> Message-ID: <282966.24351.qm@web111917.mail.gq1.yahoo.com> thanks Rupa , its very creative I saw the new feature of user rates on the trunk , but havent worked with it as im using 1.0.4 will have a look and see if it can be used BTW - I already implemented anti-loop on the LCR with custom-sql , adding the incoming carrier-id to a variable on the incoming dial plan and filtering the LCR carriers not to include same carrier , solved some issues for me when carriers send me traffic that they actually supply will document and send it to you soon thank you v much Ori ----- Original Message ---- From: Rupa Schomaker To: freeswitch-users at lists.freeswitch.org Sent: Thu, November 5, 2009 1:31:00 PM Subject: Re: [Freeswitch-users] problem with failover routes for LCR A-Z scenario Actually, using custom sql, you can implement the filter yourself in the where clause. No need for code changes. On Thu, Nov 5, 2009 at 7:17 AM, Rupa Schomaker wrote: > Now that user rates are supported in mod_lcr, how about an option that > says to drop the route if the user_rate is < rate ? > > This 1) requires you to use custom sql and 2) be able to represent > your user rates in that sql (join to user rate table perhaps?) > > On Thu, Nov 5, 2009 at 1:44 AM, Hound Dog wrote: >> I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) >> >> it came out a little long , sorry for that :) >> >> >> I found that it difficult/impossible to make LCR use only carriers that I choose >> >> scenario is as follows , taking the UK as example for a destination ( prices are not real , just an example ) >> >> I have 2 carriers offering routes to the UK , landline and mobile >> >> my buying prices >> >> Destination carrier1 Price carrier2 price >> 44 (all UK) $0.01 $0.01 >> 447 (UK mobile) $0.15 $0.19 >> >> my selling prices >> >> Destination price >> 44 (all UK) $0.015 >> 447 (UKmobile) $0.17 >> >> so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable >> >> for UK mobile I can ** only ** make a profit if I use carrier 1 ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) >> >> >> translating it to MOD_LCR information >> >> digits rate carrier_id ( other columns ignored ) >> 44 0.01 1 >> 44 0.01 2 >> 447 0.015 1 >> >> this looks good : >> 44 prefix will be shared between carrier 1 and 2 >> 447 prefix will only go to carreir 1 >> >> so it fits perfectly - BUT >> >> testing this I get - >> >> API CALL [lcr(447965404547)] output: >> | Digit Match | Carrier | Rate | Codec | CID Regexp | Dialstring | >> | 447 | carr1 | 0.15 | G711 | | [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.1 | >> | 44 | carr2 | 0.01 | G711 | | [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.2 | >> >> Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) - it all makes sense >> >> >> ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail >> >> >> so far I didnt find a solution for that , so if there is one I love bo pointed there >> >> >> >> >> >> >> I did think it over a little and came up with 2 options that could be used , >> and I am also planning to code them and propose patch to maintainers , >> I would love to get comments on those ( in case there are no existing solution ) >> >> >> option 1 - setting some routes as last option , add another param to the LCR table called last_route , >> when hitting a route with last_route=1, stop processing additional routes and return your routing decision so far >> so in our case the route entry with 44 to carrier1 will have last_route=1 , the other 44 routes will have last_route=0 to allow for failover >> >> option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. >> in other words dont failover from a finer route to a wider route. >> it will need to be a global option and I will be quite simple to use, >> but will require entering mutiple entries of the same length prefix for each carrier you would like to use >> its intutive and relatively simple to manage , but requires more lcr entries to get you where you want >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -Rupa > -- -Rupa _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From rupa at rupa.com Thu Nov 5 07:55:41 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 5 Nov 2009 09:55:41 -0600 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> Message-ID: This is true, BUT it is more flexible than it looks. http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E The caller controls can have a key execute a dialplan extension: execute_application You can set a channel var on the moderator prior to joining to the conf. When the extenion is called, you can check the channel var for moderator and act accordingly. Or you can send an event and monitor with an app over ESL and do whatever you want there (probably using the same channel var trick for knowing who is a mod or not). On Thu, Nov 5, 2009 at 8:52 AM, Rob Forman wrote: > Hi UK, > > ?From what I've done and read, the caller-controls (in > conference.conf.xml) can be modified to almost anything you can think > of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a > caller control just for those with the moderator flag. ?So unless you > want everyone able to mute/kick everyone then you can't do it. > > The wiki seems to indicate this as well: > > "Be aware that the caller-controls are applied across the entire > conference. You cannot enter one member of the conference using caller- > controls ABC and then enter a second member using caller-controls XYZ." > > http://wiki.freeswitch.org/wiki/Mod_conference > > > I think this might be a limitation of mod_conference. ?Perhaps one of > the pros can chime in if I'm off-base or there's some nifty way to > accomplish this. > > Cheers, > Rob > > On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > >> Any ideas on the below...has anyone implemented the below: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, November 02, 2009 12:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Rob: >> >> ? Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 9:34 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hm, strange. ?I haven't seen that before. ?Can you pastebin your logs >> at debug level? >> >> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >> >>> It's strange... a tcpdump tells me that there is no DTMF from my >>> provider when using IVR, but when I call into a TN that goes >>> directly into the Conference App, I see DTMF from the provider. >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 7:23 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I've never had any problem with that. ?Is your logging at debug level >>> so you can see the RECV DTFM in the log/fs_cli? ?Are you calling from >>> a SIP phone on the pbx, or via a PSTN provider? ?Maybe your provider >>> isn't passing them through. >>> >>> Make sure your logging is turned up then try something simpler, like >>> calling the echo application, and see if DTFM comes through. >>> >>> Rob >>> >>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>> >>>> Rob: >>>> >>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>> using the below configuration....so it basically timesout. >>>> >>>> UK >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Ujjval Karihaloo >>>> Sent: Monday, October 26, 2009 9:21 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>> correct.. >>>> =============== >>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. ?I guess that's why I didn't understand >>>> the point of the +pin. >>>> ====================== >>>> >>>> I will try it out. >>>> >>>> Again thx a lot for your help. Will keep everyone posted. >>>> >>>> ________________________________________ >>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>> Sent: Friday, October 23, 2009 12:22 PM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> I just re-tested with the pin in my dial plan: >>>> >>>> >>>> >>>> And it doesn't challenge me for the pin. ?I just drop right in. ?I >>>> figured this is how it was intended, since the wiki says the pin is >>>> set initially and only challenged in later attempts [by future >>>> callers]: >>>> >>>> "The first time a conference name (confname) is used, it will be >>>> created on demand, and the pin will be set to what ever is specified >>>> at that time: the pin in the data string if specified, or if not, >>>> the >>>> "pin" setting in the conference profile, and if that is also >>>> unspecified, then there is no pin protection. Any later attempt to >>>> join the conference must specify the same pin number, if one existed >>>> when it was created. " >>>> >>>> >>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. ?I guess that's why I didn't understand >>>> the point of the +pin. >>>> >>>> I'm sure there's a scenario where its used and useful, the wiki just >>>> doesn't explain it. >>>> >>>> Rob >>>> >>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>> >>>>> Well first off you're not defining a pine here... >>>>> >>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>> pin >>>>> number] >>>>> >>>>> That might be why its not asking for a pin. >>>>> >>>>> /b >>>>> >>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>> >>>>>> ? >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From rupa at rupa.com Thu Nov 5 07:59:49 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 5 Nov 2009 09:59:49 -0600 Subject: [Freeswitch-users] problem with failover routes for LCR A-Z scenario In-Reply-To: <282966.24351.qm@web111917.mail.gq1.yahoo.com> References: <727374.16142.qm@web111917.mail.gq1.yahoo.com> <282966.24351.qm@web111917.mail.gq1.yahoo.com> Message-ID: On Thu, Nov 5, 2009 at 9:46 AM, Hound Dog wrote: > thanks Rupa , > > its very creative > > I saw the new feature of user rates on the trunk , but havent worked with it as im using 1.0.4 > will have a look and see if it can be used Cool, if you have issues let me know. We can work through 'em. > BTW - I already implemented anti-loop on the LCR with custom-sql , adding the incoming carrier-id to a variable on the incoming dial plan and filtering the LCR carriers not to include same carrier , solved some issues for me when carriers send me traffic that they actually supply > > will document and send it to you soon custom sql really is a powerful feature. It does require a bit of thinking, but since it can use any channel variable it is very flexible. > > > thank you v much > Ori > > > > > ----- Original Message ---- > From: Rupa Schomaker > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, November 5, 2009 1:31:00 PM > Subject: Re: [Freeswitch-users] problem with failover routes for LCR A-Z scenario > > Actually, using custom sql, you can implement the filter yourself in > the where clause. ?No need for code changes. > > On Thu, Nov 5, 2009 at 7:17 AM, Rupa Schomaker wrote: >> Now that user rates are supported in mod_lcr, how about an option that >> says to drop the route if the user_rate is < rate ? >> >> This 1) requires you to use custom sql and 2) be able to represent >> your user rates in that sql (join to user rate table perhaps?) >> >> On Thu, Nov 5, 2009 at 1:44 AM, Hound Dog wrote: >>> I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) >>> >>> it came out a little long , sorry for that :) >>> >>> >>> I found that it difficult/impossible to make LCR use only carriers that I choose >>> >>> scenario is as follows , taking the UK as example for a destination ?( prices are not real , just an example ) >>> >>> I have 2 carriers offering routes to the UK , landline and mobile >>> >>> my buying prices >>> >>> Destination ? ? ? carrier1 Price ? ?carrier2 price >>> 44 ?(all UK) ? ? ?$0.01 ? ? ? ? ? ? $0.01 >>> 447 (UK mobile) ? $0.15 ? ? ? ? ? ? $0.19 >>> >>> my selling prices >>> >>> Destination ? ? ? ? price >>> 44 ?(all UK) ? ? ? ?$0.015 >>> 447 (UKmobile) ? ? ?$0.17 >>> >>> so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable >>> >>> for UK mobile I can ** only ** make a profit if I use carrier 1 ? ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) >>> >>> >>> translating it to MOD_LCR information >>> >>> digits ? ? rate ? ? ? ?carrier_id ? ? ? ( other columns ignored ) >>> 44 ? ? ? ? 0.01 ? ? ? ?1 >>> 44 ? ? ? ? 0.01 ? ? ? ?2 >>> 447 ? ? ? ?0.015 ? ? ? 1 >>> >>> this looks good : >>> ? ? 44 prefix will be shared between carrier 1 and 2 >>> ? ? 447 prefix will only go to carreir 1 >>> >>> so it fits perfectly - BUT >>> >>> testing this I get - >>> >>> API CALL [lcr(447965404547)] output: >>> ?| Digit Match | Carrier | Rate ? ? | Codec | CID Regexp | Dialstring | >>> ?| 447 ? ? ? ? | carr1 ? | 0.15 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.1 | >>> ?| 44 ? ? ? ? ?| carr2 ? | 0.01 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.2 | >>> >>> Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) ?- it all makes sense >>> >>> >>> ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail >>> >>> >>> so far I didnt find a solution for that , so if there is one I love bo pointed there >>> >>> >>> >>> >>> >>> >>> I did think it over a little and came up with 2 options that could be used , >>> ?and I am also planning to code them and propose patch to maintainers , >>> ?I would love to get comments on those ( in case there are no existing solution ) >>> >>> >>> option 1 - setting some routes as last option , add another param to the LCR table called ?last_route , >>> ?when hitting a route with last_route=1, ?stop processing additional routes and return your routing decision so far >>> ?so in our case the route entry with 44 to carrier1 will have last_route=1 ?, the other 44 routes will have last_route=0 to allow for failover >>> >>> option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. >>> in other words dont failover from a finer route to a wider route. >>> it will need to be a global option and I will be quite simple to use, >>> ?but will require entering mutiple entries of the same length prefix for each carrier you would like to use >>> ?its intutive and relatively simple to manage , but requires more lcr entries to get you where you want >>> >>> >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> -Rupa >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From pjintheusa at gmail.com Thu Nov 5 08:11:31 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 5 Nov 2009 11:11:31 -0500 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> Message-ID: <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> At $450 on ebay - "Polycom ip6000's AND bust" seems more apt! :) On Thu, Nov 5, 2009 at 10:07 AM, Brian West wrote: > Polycom ip6000's or bust! > > /b > > On Nov 5, 2009, at 8:57 AM, Rob Forman wrote: > > > Hey all, > > > > Looking at buying some high def phones. Any recommendations > > (preferably based on experience) for hardware based on product > > quality, standards compliance, features integration with Freeswitch, > > etc? > > > > Thank you! > > Rob Forman > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/cf930ccb/attachment.html From brian at freeswitch.org Thu Nov 5 08:17:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 10:17:22 -0600 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> Message-ID: Depends... :P If you want quality audio you pay for it... if you want shitty audio you pay for it too just a little less... so your choices are awesome audio or ok audio. /b On Nov 5, 2009, at 10:11 AM, Phillip Jones wrote: > At $450 on ebay - "Polycom ip6000's AND bust" seems more apt! :) From steveu at coppice.org Thu Nov 5 08:43:12 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 06 Nov 2009 00:43:12 +0800 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> Message-ID: <4AF300A0.8000206@coppice.org> Hi Phillip, I'm not sure why Brian suggested the IP6000. Its a big chunky conference room speakerphone (the large triangular type). Very nice, but probably not what you were looking for. If your idea of high def is G.722 there are more conventional phones for half that price. If your idea of high def is something genuinely high definition, like a G.722.1C phone, the choice is rather limited, unless you pay an arm and a leg. Steve On 11/06/2009 12:11 AM, Phillip Jones wrote: > At $450 on ebay - "Polycom ip6000's AND bust" seems more apt! :) > > On Thu, Nov 5, 2009 at 10:07 AM, Brian West > wrote: > > Polycom ip6000's or bust! > > /b > > On Nov 5, 2009, at 8:57 AM, Rob Forman wrote: > > > Hey all, > > > > Looking at buying some high def phones. Any recommendations > > (preferably based on experience) for hardware based on product > > quality, standards compliance, features integration with Freeswitch, > > etc? > > > > Thank you! > > Rob Forman > From mkitchin.public at gmail.com Thu Nov 5 08:46:34 2009 From: mkitchin.public at gmail.com (mkitchin.public at gmail.com) Date: Thu, 05 Nov 2009 10:46:34 -0600 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <63de75710911041057x472d44aj9bc52bb460a8c8cd@mail.gmail.com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> <025101ca5d10$f81228c0$e8367a40$@com> <20091104064201.GA15804@jdc.jasonjgw.net> <027f01ca5d6c$9c1603a0$d4420ae0$@com> <63de75710911041057x472d44aj9bc52bb460a8c8cd@mail.gmail.com> Message-ID: <4AF3016A.2020100@gmail.com> I hate to say it, but I had to give in and try sipx. the ease of provisioning phones and the ability for helpdesk staff to reset passwords and such through a gui looks like it is too good for me to pass on. mm_202 wrote: > I had the exact same problem with the Cisco phones not being able to > receive calls. > > I fixed it by messing around with the NAT settings in the internal > sofia profile. From what I remember, > I just removed the line > and everything worked fine. > > -- mm_202. > > On Wed, Nov 4, 2009 at 11:33 AM, Peter J. Zandvoort > wrote: > >> Absolutely agreed. To use Matthew's original car metaphor: When you just got >> your learner's permit, the old Chevy may be a better choice than the >> Ferrari. >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason >> White >> Sent: Wednesday, November 04, 2009 1:42 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones >> >> Peter J. Zandvoort wrote: >> >>> After looking at various asterisk distributions, SipX, 3CX and >>> what-have-you, I've come to the conclusion that FreeSWITCH is by far the >>> most advanced platform out there. Its architecture and performance is >>> literally light years ahead of the rest and I have yet to come up with >>> something that it can't do. But all that comes at a price: The learning >>> curve is like scaling a brick wall. >>> >> The most flexible and sophisticated tools tend to have this characteristic, >> the best solution to which is a supportive community and good documentation. >> FreeSWITCH has the community; the documentation is improving thanks to >> ongoing >> efforts to extend, clarify and enhance the wiki. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From larclap at yahoo.com Thu Nov 5 09:27:56 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 5 Nov 2009 09:27:56 -0800 Subject: [Freeswitch-users] FS hangup Message-ID: <00b401ca5e3d$50b7a8b0$f226fa10$@com> I just updated to v15372 from v15311. When calling into FreeSWITCH, it hangs up the call rather than going to voicemail (line 262 in pastebin). I don't know what might be causing this. Can anyone help? Thanks, Lars http://pastebin.freeswitch.org/11006 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/60a65779/attachment.html From dujinfang at gmail.com Thu Nov 5 09:32:25 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Nov 2009 01:32:25 +0800 Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: <23f91030911050932m2695b23eg6a253954b109f16d@mail.gmail.com> 2009/11/5 ??? > Hi All, > > I once meet the Skypiax load error issue and some guys infomed me that > there is no configuration file for Skypiax. > > When I follow these intructions -> > > http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances > > I still have some difficulties unstanding this: > ." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml > to Debug\conf\autoload_configs." > *I did not find such directories in my freeswitch folder. Did not > understand what "src" means, I checked the freeswitch folder and did not > find such a folder named"src". There is a folder named"mod" under freeswitch > while look into "mod" folder there are only some DLL files and can not find > endpoints and etc.* > > src means source code. you can check out from svn trunk or download the source code from files.freeswitch.org. or follow fisheye: http://fisheye.freeswitch.org/browse/FreeSWITCH > 2.You'll probably build the "Debug" version > *I just build this from the precompiled binaries and then launched FS. > I'm not sure what I launched is in debug mode or not. > * > > If anyone can offer some help that really be appriciated. > I never try to do that on windows, but we are running skype trunks on Linux. Also there's a Chinese google group about FreeSWITCH: http://groups.google.com/group/freeswitch-cn And also please don't include long unrelated quoted text in your mail. It wastes time to read and hence some read emails on cellphone. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/bdba7242/attachment.html From gmaruzz at celliax.org Thu Nov 5 09:41:41 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 5 Nov 2009 18:41:41 +0100 Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: <7b197bef0911050941r1299922amba1f3f1a4b204674@mail.gmail.com> 2009/11/5 ??? : > I once meet the Skypiax load error issue and some guys infomed me that there > is no configuration file for Skypiax. Daniel, maybe this will sound not nice to your hears, but really, you better find a friend that stay there with you and help you and teach you. You will never be able to do things with FS with your present level of skill. FS (and more so mod_skypiax) is not a consumer grade package. It requires a level of skill that you do not have, at the moment. Maybe is sad, but in my opinion is true. I tell you this just to avoid you a lot of frustrations. -gm > > When I follow these intructions -> > http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances > > I still have some difficulties unstanding this: > ." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml to > Debug\conf\autoload_configs." > I did not find such directories in my freeswitch folder. Did not understand > what "src" means, I checked the freeswitch folder and did not find such a > folder named"src". There is a folder named"mod" under freeswitch while look > into "mod" folder there are only some DLL files and can not find endpoints > and etc. > > 2.You'll probably build the "Debug" version > I just build this from the precompiled binaries and then launched FS. I'm > not sure what I launched is in debug mode or not. > > If anyone can offer some help that really be appriciated. > > Thanks > Daniel Zeng > From: freeswitch-users-request at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 41, Issue 40 > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 5 Nov 2009 06:57:44 -0800 > > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > --??????-- > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date : Thu, 5 Nov 2009 08:13:46 -0600 > Subject: Re: [Freeswitch-users] no REINVITE on Blind Transfer with > bypass_media > > I did not ask you to send me a ladder diagram. > I asked you to send me a console trace from FreeSWITCH using latest trunk > (1.0.4 does not help me) > > 1) start FreeSWITCH > 2) run the cli command: console loglevel debug > > 3) run the cli command: sofia profile internal siptrace on > 4) reproduce your issue and put the trace on freeswitch pastebin > http://pastebin.freeswitch.org (login and pass are stated in the auth > dialog) > > > > Also please answer brian's question. What phones and/or sip devices are > involved in this call. > > > > On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana > wrote: > > > > Thanks for your time, > > > > -The scenario is still the same: > > > > Always bypass media. > > Environment 100% NAT free :-) > > Call established from A to B through FS. Then... > > Blind transfer from B to C (Refer-to: C) > > RTP should go directly between A and C. > > > > > > -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the > REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the > lack of reINVITE to A, after C answers). > > > > Please check SIP diagram here: > > > > http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html > > > > > > -What it's wrong with r15332 is there is not such call to C. For sure I know > SIP is a protocol, may be my description was not clear but this SIP diagram > speaks by itself ;-) > > > > http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html > > > > > > -You could check the sofia debug for r15332 here: > > http://pastebin.com/m6f2b3836 > > > > > > Best regards, > > > > Humberto > > > >> > >> I don't know what you are talking about anymore. > >> > >> The scenario I had tested is when a call is bridged in bypass_media=true > >> bridge > >> and you blind transfer that call back to the dialplan > >> > >> as soon as it hits the routing state it will resume media. > >> > >> > >> it has been confirmed to not work and confirmed to have been fixed several > >> time and if you are still having a problem you must have something >> blocking > >> some of your packets or something . > >> > >> You have to understand that sip is a protocol and your description is > >> completely non-standard. > >> Perhaps you should get a console trace and attach it to a jira. The trace > >> probably makes more sense to me. > >> > >> sofia profile internal siptrace on > >> console loglevel debug > >> > >> reprodu ce and attach the whole capture. > >> > >> > >> > >> On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > >> > >>> > >>> Hi, > >>> > >>> I tried r15332 and set in the sofia profile: > >>> > >>> a) bypass_media_after_bridge=true only > >>> b) bypass_media_after_bridge=true, param name="media-option" > >>> value="resume-media-on-hold"/> > >>> > >>> > >>> In both cases FS is hanging up the initial call (A to FS) after accepting > >>> the REFER to C: > >>> > >>> A <- reINVITE with FS' SDP <- FS > >>> A -> 200 -> FS > >>> A <- ACK <- FS > >>> A <- BYE <- FS > >>> > >>> The call to C is not even tried. > >>> > >>> I found this line is the logs that could give some idea: > >>> > >>> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup > >>> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] >>> [RECOVERY_ON_TIMER_EXPIRE] > >>> after sending the ACK for the reINVITE > >>> > >>> > >>> Regards, > >>> > >>> > >>> Humberto > >>> > >>>>please try r15326 > >>>>I think i have it working. > >>>> > >>>>I recommend for optimal results you set bypass_media_after_bridge=true > >>>>either as a global or in your DP in place of bypass_media=true > >>>> > >>>> > >>>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana > >>> hotmail.com>wrote: > >>>> > >>>>> Hi Mike, > >>>>> > >>>>> I re-tried with trunk rev 15319 but I got almost the same behavior: > >>> There > >>>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. > >>> But > >>>>> still there is no reINVITE for A (with C's SDP) after the call from FS > >>> to C > >>>>> is established. > >>>>> > >>>>> Anyway, we decided for now to do a different implementation but if you > >>> want > >>>>> to explore more in this issue count me in ;-) > >>>>> > >>>>> > >>>>> Thank you very much! > >>>>> > >>>>> Humberto > >>> > >>> > >>> _____________________________________ ____________________________ > >>> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they > >>> e-mail you. > >>> http://go.microsoft.com/?linkid=9691817 > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> _________________________________________________________________ > >> Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows >> 7 now > >> http://go.microsoft.com/?linkid=9691818 > > > > _________________________________________________________________ > > Windows Live: Make it easier for your friends to see what you're up to on > Facebook. > > http://go.microsoft.com/?linkid=9691816 > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > --??????-- > From: rob4manhere at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 5 Nov 2009 08:52:05 -0600 > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hi UK, > > From what I've done and read, the caller-controls (in > conference.conf.xml) can be modified to almost anything you can think > of, BUT, > they are mapped 1-to-1 to a conference- ie you can't map a > caller control just for those with the moderator flag. So unless you > want everyone able to mute/kick everyone then you can't do it. > > The wiki seems to indicate this as well: > > "Be aware that the caller-controls are applied across the entire > conference. You cannot enter one member of the conference using caller- > controls ABC and then enter a second member using caller-controls XYZ." > > http://wiki.freeswitch.org/wiki/Mod_conference > > > I think this might be a limitation of mod_conference. Perhaps one of > the pros can chime in if I'm off-base or there's some nifty way to > accomplish this. > > Cheers, > Rob > > On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > >> Any ideas on the below...has anyone implemented the below: >> >> Once I have the Moderator and Participants logg > ed on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, November 02, 2009 12:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Rob: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman> Sent: Friday, October 30, 2009 9:34 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hm, strange. I haven't seen that before. Can you pastebin your logs >> at debug level? >> >> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >> >>> It's strange... a tcpdump tells me that there is no DTMF from my >>> provider when using IVR, but when I call into a TN that goes >>> directly into the Conference App, I see DTMF from the provider. >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 7:23 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference w > ith Moderator >>> >>> I've never had any problem with that. Is your logging at debug level >>> so you can see the RECV DTFM in the log/fs_cli? Are you calling from >>> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >>> isn't passing them through. >>> >>> Make sure your logging is turned up then try something simpler, like >>> calling the echo application, and see if DTFM comes through. >>> >>> Rob >>> >>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>> >>>> Rob: >>>> >>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>> using the below configuration....so it basically timesout. >>>> >>>> UK >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Ujjval Karihaloo >>>> Sent: Monday, October 26, 2009 9:21 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>> correct.. >>>> =============== >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> ====================== >>>> >>>> I will try it out. >>>> >>>> Again thx a lot for your help. Will keep everyone posted. >>>> >>>> _______________________ > _________________ >>>> From: freeswitch-users-bounces at lists.freeswitch.org >>>> [freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>> Sent: Friday, October 23, 2009 12:22 PM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> I just re-tested with the pin in my dial plan: >>>> >>>> >>>> >>>> And it doesn't challenge me for the pin. I just drop right in. I >>>> figured this is how it was intended, since the wiki says the pin is >>>> set initially and only challenged in later attempts [by future >>>> callers]: >>>> >>>> "The first time a conference name (confname) is used, i > t will be >>>> created on demand, and the pin will be set to what ever is specified >>>> at that time: the pin in the data string if specified, or if not, >>>> the >>>> "pin" setting in the conference profile, and if that is also >>>> unspecified, then there is no pin protection. Any later attempt to >>>> join the conference must specify the same pin number, if one existed >>>> when it was created. " >>>> >>>> >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> >>>> I'm sure there's a scenario where its used and useful, the wiki just >>>> doesn't explain it. >>>> >>> > > Rob >>>> >>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>> >>>>> Well first off you're not defining a pine here... >>>>> >>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>> pin >>>>> number] >>>>> >>>>> That might be why its not asking for a pin. >>>>> >>>>> /b >>>>> >>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> __________ > _____________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.free > switch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/ > freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > > --??????-- > From: rob4manhere at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: T hu, 5 Nov 2009 08:57:08 -0600 > Subject: [Freeswitch-users] Wideband / HD phones > > Hey all, > > Looking at buying some high def phones. Any recommendations > (preferably based on experience) for hardware based on product > quality, standards compliance, features integration with Freeswitch, > etc? > > Thank you! > Rob Forman > > > > ________________________________ > ??????????MSN??? ????? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From shiyanov at gmail.com Thu Nov 5 09:44:28 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 5 Nov 2009 20:44:28 +0300 Subject: [Freeswitch-users] Dialpan: try.. finally analogs Message-ID: Hello! I have to deal with classic problem: "Leaking stream handle" in FS console. I also know the reason - firstly channel is sent to the extension with "playback" and later it is transfered to another extensions with "execute_extension" or, another trouble-case - channel is bridged to some addres. I do not ask (but I wish to) why FS doesn't close stream automatically when channel is gone. I ask whether it is possible to use some "try.. finally" construction in diaplan? If "yes" then I can simply stop playback in the "finally" block.. Any thoughs? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/2e5e5682/attachment.html From frank at carmickle.com Thu Nov 5 09:45:30 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 5 Nov 2009 12:45:30 -0500 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <4AF300A0.8000206@coppice.org> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> <4AF300A0.8000206@coppice.org> Message-ID: <20091105174529.GO10757@base.carmickle.com> On Fri, Nov 06, Steve Underwood wrote: > If your idea of high def is G.722 there are more conventional phones for > half that price. And there is portaudio in freeswitch itself. With a usb headset and celt at 48k how can you go wrong? --FC From anthony.minessale at gmail.com Thu Nov 5 09:47:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Nov 2009 11:47:47 -0600 Subject: [Freeswitch-users] FS hangup In-Reply-To: <00b401ca5e3d$50b7a8b0$f226fa10$@com> References: <00b401ca5e3d$50b7a8b0$f226fa10$@com> Message-ID: <191c3a030911050947o6bbc3e97j24898aa50951f54@mail.gmail.com> do you have continue_on_fail set? if you do you have to include no_answer,busy etc once you set it, you have to set *everything* you want. On Thu, Nov 5, 2009 at 11:27 AM, Lars Zeb wrote: > I just updated to v15372 from v15311. When calling into FreeSWITCH, it > hangs up the call rather than going to voicemail (line 262 in pastebin). I > don?t know what might be causing this. > > > > Can anyone help? > > > > Thanks, Lars > > > > http://pastebin.freeswitch.org/11006 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/6069e3b1/attachment.html From shiyanov at gmail.com Thu Nov 5 09:49:49 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 5 Nov 2009 20:49:49 +0300 Subject: [Freeswitch-users] Filtering a particular event. In-Reply-To: References: Message-ID: it's possible. Mod_socket commands: event myevents event plain On Thu, Nov 5, 2009 at 4:28 PM, Nagalenoj H. wrote: > Hi, > I've tried to filter the events like below to filter a particular event. > > 1) register for all events > 2) filter for one unique-id > 3) filter only one/more events(ex: DTMF & CHANNEL_EXECUTE) > > So, I want to receive only these events for the specific unique-id. But, I > am receiving other events too. I'm using perl ESL outbound. > Is it possible to do like this?! > > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/09492375/attachment.html From shiyanov at gmail.com Thu Nov 5 09:52:15 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 5 Nov 2009 20:52:15 +0300 Subject: [Freeswitch-users] Java example In-Reply-To: <85845D7B-9D9D-4BFA-ACCA-0F28DA4EBA9E@freeswitch.org> References: <44498.1257162831@entvoice.com> <85845D7B-9D9D-4BFA-ACCA-0F28DA4EBA9E@freeswitch.org> Message-ID: let's say: starpound is using FS for business process automation + telephony On Mon, Nov 2, 2009 at 10:07 PM, Brian West wrote: > Is starpound involved in the FS Community? > > /b > > > On Nov 2, 2009, at 12:51 PM, Artem Shiyanov wrote: > > Here is rather big and, let's say, complete example of mod_java usage: > https://starpound.svn.sourceforge.net/svnroot/starpound/trunk/src/fs2agi > The goal of this project is to be a proxy between FreeSwitch and server > application which knows Asterisk AGI. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/44ada5a5/attachment.html From brian at freeswitch.org Thu Nov 5 09:55:31 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 11:55:31 -0600 Subject: [Freeswitch-users] Java example In-Reply-To: References: <44498.1257162831@entvoice.com> <85845D7B-9D9D-4BFA-ACCA-0F28DA4EBA9E@freeswitch.org> Message-ID: But they should be more involved in the Community if possible. /b On Nov 5, 2009, at 11:52 AM, Artem Shiyanov wrote: > let's say: starpound is using FS for business process automation + > telephony > > > > On Mon, Nov 2, 2009 at 10:07 PM, Brian West > wrote: > Is starpound involved in the FS Community? > > /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/28d14896/attachment.html From dujinfang at gmail.com Thu Nov 5 09:57:59 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Nov 2009 01:57:59 +0800 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> Message-ID: <23f91030911050957m796fe88fj5da881875c010e6b@mail.gmail.com> Ciao Giovanni, Do you still plan to merge this? 2009/9/5 Giovanni Maruzzelli > Seven, > > thanks a lot for your efforts. > > I will merge it in the next days, and I will take care that it will > not breaks Windows or Linux. > > If I find problems I will wait for you having more time in the future. > > I send you my super best wishes for your personal things to go well > and solves in the best of the possible ways. > > ciao for now, > > -giovanni > > > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > > > On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: > > gm, > > > > Thanks a lot you can merge into the mainline. I check into my branch > > because it's currently not as useful as on Linux and Windows and the > > solution is not good. But it works and it is a good start that > > mod_skypiax runs on OSX. Sure it would be easier for people want to > > test and improve it if it been merged into trunk. I think you can make > > a diff by > > > > svn diff -r 14472:14772 > http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax > > > > FYI for personal reason I won't have much time put on this in the > > coming month. Actually the code was done a few weeks ago but i only > > got a chance to commit it yesterday. Sure that is not to say I cannot > > do but fixes. But can you please make sure it won't break Linux/ > > windows build when you merge the code? I haven't have a chance to test > > all of them yet. > > > > -7- > > > > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: > >> Seeeeeeeven! > >> > >> I saw the modification you made on the wiki page... > >> > >> You made it, mod_skypiax runs on OSX!!!! > >> > >> Let's merge your mods on the mainline, pleaaaase ;-))) > >> > >> -giovanni > >> > >> > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/5c3ba1c2/attachment-0001.html From gmaruzz at celliax.org Thu Nov 5 10:03:43 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 5 Nov 2009 19:03:43 +0100 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <23f91030911050957m796fe88fj5da881875c010e6b@mail.gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> <23f91030911050957m796fe88fj5da881875c010e6b@mail.gmail.com> Message-ID: <7b197bef0911051003t13e363edqff7b76ecfc099ed5@mail.gmail.com> On Thu, Nov 5, 2009 at 6:57 PM, Seven Du wrote: > Ciao Giovanni, > > Do you still plan to merge this? Sorry Seven, I've lost track of this, and now I'm so sick I'm completely un-useful ;). But yes, I would like to do it, if you think it is in a useful state. Can you please create a Jira and attach an svn diff, so in the next days I can merge it? -giovanni > > 2009/9/5 Giovanni Maruzzelli >> >> Seven, >> >> thanks a lot for your efforts. >> >> I will merge it in the next days, and I will take care that it will >> not breaks Windows or Linux. >> >> If I find problems I will wait for you having more time in the future. >> >> I send you my super best wishes for your personal things to go well >> and solves in the best of the possible ways. >> >> ciao for now, >> >> -giovanni >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> >> >> >> On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: >> > gm, >> > >> > Thanks a lot you can merge into the mainline. I check into my branch >> > because it's currently not as useful as on Linux and Windows and the >> > solution is not good. But it works and it is a good start that >> > mod_skypiax runs on OSX. Sure it would be easier for people want to >> > test and improve it if it been merged into trunk. I think you can make >> > a diff by >> > >> > svn diff -r 14472:14772 >> > http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax >> > >> > FYI for personal reason I won't have much time put on this in the >> > coming month. Actually the code was done a few weeks ago but i only >> > got a chance to commit it yesterday. Sure that is not to say I cannot >> > do but fixes. But can you please make sure it won't break Linux/ >> > windows build when you merge the code? I haven't have a chance to test >> > all of them yet. >> > >> > -7- >> > >> > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: >> >> Seeeeeeeven! >> >> >> >> I saw the modification you made on the wiki page... >> >> >> >> You made it, mod_skypiax runs on OSX!!!! >> >> >> >> Let's merge your mods on the mainline, pleaaaase ;-))) >> >> >> >> -giovanni >> >> >> >> >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> Cell : +39-347-2665618 >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From larclap at yahoo.com Thu Nov 5 10:17:03 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 5 Nov 2009 10:17:03 -0800 Subject: [Freeswitch-users] FS hangup In-Reply-To: <191c3a030911050947o6bbc3e97j24898aa50951f54@mail.gmail.com> References: <00b401ca5e3d$50b7a8b0$f226fa10$@com> <191c3a030911050947o6bbc3e97j24898aa50951f54@mail.gmail.com> Message-ID: <00d201ca5e44$2d59b000$880d1000$@com> Thanks for the help. Yes, I am using a lua script to handle inbound calls with continue_on_fail set to true: session:execute("set", "continue_on_fail=true"); I changed it to: session:execute("set", "continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DEST INATION"); and it works OK now. Did something change between v15311 to v15372 to make this behave differently? I ask because it worked with "true" in the earlier version. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, November 05, 2009 9:48 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS hangup do you have continue_on_fail set? if you do you have to include no_answer,busy etc once you set it, you have to set *everything* you want. On Thu, Nov 5, 2009 at 11:27 AM, Lars Zeb wrote: I just updated to v15372 from v15311. When calling into FreeSWITCH, it hangs up the call rather than going to voicemail (line 262 in pastebin). I don't know what might be causing this. Can anyone help? Thanks, Lars http://pastebin.freeswitch.org/11006 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/ba3ea9f5/attachment.html From anthony.minessale at gmail.com Thu Nov 5 10:35:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Nov 2009 12:35:47 -0600 Subject: [Freeswitch-users] FS hangup In-Reply-To: <00d201ca5e44$2d59b000$880d1000$@com> References: <00b401ca5e3d$50b7a8b0$f226fa10$@com> <191c3a030911050947o6bbc3e97j24898aa50951f54@mail.gmail.com> <00d201ca5e44$2d59b000$880d1000$@com> Message-ID: <191c3a030911051035y739919c7w2663025e93d64976@mail.gmail.com> yes sounds like a bug. I think i redid it and forgot to check for "true" still =0 On Thu, Nov 5, 2009 at 12:17 PM, Lars Zeb wrote: > Thanks for the help. Yes, I am using a lua script to handle inbound calls > with continue_on_fail set to true: > > > > session:execute("set", "continue_on_fail=true"); > > > > I changed it to: > > > > session:execute("set", > "continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"); > > > > and it works OK now. > > > > Did something change between v15311 to v15372 to make this behave > differently? I ask because it worked with ?true? in the earlier version. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Thursday, November 05, 2009 9:48 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FS hangup > > > > do you have continue_on_fail set? > if you do you have to include no_answer,busy etc once you set it, you have > to set *everything* you want. > > > On Thu, Nov 5, 2009 at 11:27 AM, Lars Zeb wrote: > > I just updated to v15372 from v15311. When calling into FreeSWITCH, it > hangs up the call rather than going to voicemail (line 262 in pastebin). I > don?t know what might be causing this. > > > > Can anyone help? > > > > Thanks, Lars > > > > http://pastebin.freeswitch.org/11006 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/6e60e188/attachment-0001.html From anthony.minessale at gmail.com Thu Nov 5 10:40:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Nov 2009 12:40:03 -0600 Subject: [Freeswitch-users] FS hangup In-Reply-To: <191c3a030911051035y739919c7w2663025e93d64976@mail.gmail.com> References: <00b401ca5e3d$50b7a8b0$f226fa10$@com> <191c3a030911050947o6bbc3e97j24898aa50951f54@mail.gmail.com> <00d201ca5e44$2d59b000$880d1000$@com> <191c3a030911051035y739919c7w2663025e93d64976@mail.gmail.com> Message-ID: <191c3a030911051040p31e04e4cj4f27c2821edce64@mail.gmail.com> fixed in 15376 On Thu, Nov 5, 2009 at 12:35 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > yes sounds like a bug. > I think i redid it and forgot to check for "true" still =0 > > > > On Thu, Nov 5, 2009 at 12:17 PM, Lars Zeb wrote: > >> Thanks for the help. Yes, I am using a lua script to handle inbound >> calls with continue_on_fail set to true: >> >> >> >> session:execute("set", "continue_on_fail=true"); >> >> >> >> I changed it to: >> >> >> >> session:execute("set", >> "continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"); >> >> >> >> and it works OK now. >> >> >> >> Did something change between v15311 to v15372 to make this behave >> differently? I ask because it worked with ?true? in the earlier version. >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony >> Minessale >> *Sent:* Thursday, November 05, 2009 9:48 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] FS hangup >> >> >> >> do you have continue_on_fail set? >> if you do you have to include no_answer,busy etc once you set it, you have >> to set *everything* you want. >> >> >> On Thu, Nov 5, 2009 at 11:27 AM, Lars Zeb wrote: >> >> I just updated to v15372 from v15311. When calling into FreeSWITCH, it >> hangs up the call rather than going to voicemail (line 262 in pastebin). I >> don?t know what might be causing this. >> >> >> >> Can anyone help? >> >> >> >> Thanks, Lars >> >> >> >> http://pastebin.freeswitch.org/11006 >> >> >> >> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 >> i386 GNU/Linux >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/2960fde8/attachment.html From hjqlopez at hotmail.com Thu Nov 5 10:39:59 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Thu, 5 Nov 2009 13:39:59 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: <403475.98637.qm@web51107.mail.re2.yahoo.com> References: <403475.98637.qm@web51107.mail.re2.yahoo.com> Message-ID: >> -You could check the sofia debug for r15332 here: http://pastebin.freeswitch.org/11008 Phone/Devices: The caller is the DID provider's Switch. The callee (which also sends the REFER) is Asterisk 1.4.26. Testing with other devices(linksys SPA962,Grandstream GXV3000) gathers the same results. > I did not ask you to send me a ladder diagram. > I asked you to send me a console trace from FreeSWITCH using latest trunk > (1.0.4 does not help me) > > 1) start FreeSWITCH > 2) run the cli command: console loglevel debug > 3) run the cli command: sofia profile internal siptrace on > 4) reproduce your issue and put the trace on freeswitch pastebin > http://pastebin.freeswitch.org (login and pass are stated in the auth > dialog) > > Also please answer brian's question. What phones and/or sip devices are > involved in this call. > > > > On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana wrote: > >> >> Thanks for your time, >> >> -The scenario is still the same: >> >> Always bypass media. >> Environment 100% NAT free :-) >> Call established from A to B through FS. Then... >> Blind transfer from B to C (Refer-to: C) >> RTP should go directly between A and C. >> >> >> -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the >> REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the >> lack of reINVITE to A, after C answers). >> >> Please check SIP diagram here: >> >> http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html >> >> >> -What it's wrong with r15332 is there is not such call to C. For sure I >> know SIP is a protocol, may be my description was not clear but this SIP >> diagram speaks by itself ;-) >> >> http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html >> >> >> -You could check the sofia debug for r15332 here: >> http://pastebin.com/m6f2b3836 >> >> >> Best regards, >> >> Humberto >> >>> >>> I don't know what you are talking about anymore. >>> >>> The scenario I had tested is when a call is bridged in bypass_media=true >>> bridge >>> and you blind transfer that call back to the dialplan >>> >>> as soon as it hits the routing state it will resume media. >>> >>> >>> it has been confirmed to not work and confirmed to have been fixed >> several >>> time and if you are still having a problem you must have something >> blocking >>> some of your packets or something . >>> >>> You have to understand that sip is a protocol and your description is >>> completely non-standard. >>> Perhaps you should get a console trace and attach it to a jira. The trace >>> probably makes more sense to me. >>> >>> sofia profile internal siptrace on >>> console loglevel debug >>> >>> reproduce and attach the whole capture. >>> >>> >>> >>> On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: >>> >>>> >>>> Hi, >>>> >>>> I tried r15332 and set in the sofia profile: >>>> >>>> a) bypass_media_after_bridge=true only >>>> b) bypass_media_after_bridge=true, param name="media-option" >>>> value="resume-media-on-hold"/> >>>> >>>> >>>> In both cases FS is hanging up the initial call (A to FS) after >> accepting >>>> the REFER to C: >>>> >>>> A <- reINVITE with FS' SDP <- FS >>>> A -> 200 -> FS >>>> A <- ACK <- FS >>>> A <- BYE <- FS >>>> >>>> The call to C is not even tried. >>>> >>>> I found this line is the logs that could give some idea: >>>> >>>> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >>>> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] >> [RECOVERY_ON_TIMER_EXPIRE] >>>> after sending the ACK for the reINVITE >>>> >>>> >>>> Regards, >>>> >>>> >>>> Humberto >>>> >>>>>please try r15326 >>>>>I think i have it working. >>>>> >>>>>I recommend for optimal results you set bypass_media_after_bridge=true >>>>>either as a global or in your DP in place of bypass_media=true >>>>> >>>>> >>>>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >>>> hotmail.com>wrote: >>>>> >>>>>> Hi Mike, >>>>>> >>>>>> I re-tried with trunk rev 15319 but I got almost the same behavior: >>>> There >>>>>> is now a reINVITE (with FS' SDP) going to A when the REFER is >> accepted. >>>> But >>>>>> still there is no reINVITE for A (with C's SDP) after the call from FS >>>> to C >>>>>> is established. >>>>>> >>>>>> Anyway, we decided for now to do a different implementation but if you >>>> want >>>>>> to explore more in this issue count me in ;-) >>>>>> >>>>>> >>>>>> Thank you very much! >>>>>> >>>>>> Humberto > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com _________________________________________________________________ Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now http://go.microsoft.com/?linkid=9691818 From codecomplete at free.fr Thu Nov 5 10:43:54 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 5 Nov 2009 10:43:54 -0800 (PST) Subject: [Freeswitch-users] Does OpenZap support CTR21? Message-ID: <26217371.post@talk.nabble.com> Hello As an alternative to more expensive alternatives like OpenVox or Sangoma, I'd like to order an X100P clone from www.x100p.com for use in France. According to a PDF on the site, the reason this card gets bad reviews is that "the Silicon labs Si3012/Si3035 DAA chip used in the original Digium X100P card and low cost X100P clone cards only supports FCC mode. However, the Si3014/Si3034 DAA chip used on the X100P SE supports global line standards." As for software, "the Silicon labs Si3014/Si3034 DAA chip used in the X100P SE supports 600 Ohm impedance and complex impedance to meet CTR21 line standards. However, the Zaptel wcfxo driver only supports CTR21 mode with 600 Ohm AC termination, which may or may not be the correct setting depending on the country and the phone system in use." So... does someone know if OpenZap, which is apparently required in addition to Zaptel/Dahdi for FreeSwitch to work PCI TDM cards, supports CTR21? Thank you. -- View this message in context: http://old.nabble.com/Does-OpenZap-support-CTR21--tp26217371p26217371.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu Nov 5 10:47:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Nov 2009 10:47:38 -0800 Subject: [Freeswitch-users] Calling more than 1 variable in condition In-Reply-To: References: Message-ID: <87f2f3b90911051047u34559431idd898ee9e687a011@mail.gmail.com> On Wed, Nov 4, 2009 at 10:22 PM, Ahmed Munir wrote: > Hi, > > In my dial plan I've created a variable named SIP_CALL, PSTN_CALL. If > SIP_CALL = true, it dials out to sip call, when PSTN_CALL=true, it dials out > to landline call, as I provide sample below; > > > > > > > > > > > > The problem I'm facing is how can I apply condition when I've to call more > than 1 variables? Like if there are no records in SIP numbering plan table > and PSTN numbering plan table so it get the digits and dial out the to > carrier (how to apply AND operation in condition?) i.e. > > > > > > > > AND operations are very simple - just stack the conditions: <-- actions here --> note that you must close the first condition's tag! BTW, this is covered in the dialplan section of the wiki. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/052f3f50/attachment-0001.html From michal.bielicki at halo2.pl Thu Nov 5 11:05:10 2009 From: michal.bielicki at halo2.pl (Michal Bielicki) Date: Thu, 5 Nov 2009 20:05:10 +0100 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <20091105174529.GO10757@base.carmickle.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> <4AF300A0.8000206@coppice.org> <20091105174529.GO10757@base.carmickle.com> Message-ID: <4CBF4621-B12A-46F3-9C1B-90AB1D158C0A@halo2.pl> I'd rather go with the ip7000 since it has better audio gear in it. For a deskphone everything Polycom >= ip450 is absolutely wideband enough for a deskphone. Personally I am currently a total fan of the VVX which is a video deskphone with the same audio as a IP6000 Am 05.11.2009 um 18:45 schrieb Frank Carmickle: > On Fri, Nov 06, Steve Underwood wrote: >> If your idea of high def is G.722 there are more conventional >> phones for >> half that price. > > And there is portaudio in freeswitch itself. With a usb headset and > celt at 48k how can you go wrong? > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org Michal Bielicki HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa t. +48228753290 | f. +48228753291 michal.bielicki at halokwadrat.pl | w. www.halokwadrat.pl Knowledge & Low Prices. Guaranteed! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/cb9586f0/attachment.html From info at daccii.it Thu Nov 5 12:35:12 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Thu, 05 Nov 2009 21:35:12 +0100 Subject: [Freeswitch-users] Transfer call to group Message-ID: <4AF33700.1040209@daccii.it> Hi, actually i'm trying to setup an IVR that, when the choice is done, transfer the call to a group, really simply. Here the dialplan in default context to handle call to group (four extensions, one for group, from 2001 to 2004) http://pastebin.freeswitch.org/11014 Here the output log http://pastebin.freeswitch.org/11015 When i call the group directly from a telephone in the default context or when the ivr transfer me to the group i didn't get nothing, looking to log you can see (line 153) EXECUTE sofia/internal/15 at 192.168.0.77 bridge() Data for bridge application is ${group_call(${dialed_extension}+F@${domain_name})} It's, probably, a stupid error, but the only other way to accomplish this is to bridge individually phones using | as separator but i would to mantain a single extension to handle this stuff. Thanks for your support! Best Regards, Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/ebbe66a4/attachment.vcf From mike at jerris.com Thu Nov 5 13:19:42 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Nov 2009 16:19:42 -0500 Subject: [Freeswitch-users] Dialpan: try.. finally analogs In-Reply-To: References: Message-ID: It cleans up after itself fine, but it is an indication of some issue in the code we need to address. if you can reproduce this in svn trunk, please file a bug on jira.freeswitch.org with details how to reproduce. mike On Nov 5, 2009, at 12:44 PM, Artem Shiyanov wrote: > Hello! > > I have to deal with classic problem: "Leaking stream handle" in FS > console. I also know the reason - firstly channel is sent to the > extension with "playback" and later it is transfered to another > extensions with "execute_extension" or, another trouble-case - > channel is bridged to some addres. > I do not ask (but I wish to) why FS doesn't close stream > automatically when channel is gone. > I ask whether it is possible to use some "try.. finally" > construction in diaplan? If "yes" then I can simply stop playback in > the "finally" block.. > > Any thoughs? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Thu Nov 5 13:20:58 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Nov 2009 16:20:58 -0500 Subject: [Freeswitch-users] Does OpenZap support CTR21? In-Reply-To: <26217371.post@talk.nabble.com> References: <26217371.post@talk.nabble.com> Message-ID: <7FD19B47-C121-48CD-98C2-2830BFDF1068@jerris.com> This would be specific to the zaptel driver for that card, not openzap. mike On Nov 5, 2009, at 1:43 PM, Fred-145 wrote: > > Hello > > As an alternative to more expensive alternatives like OpenVox or > Sangoma, > I'd like to order an X100P clone from www.x100p.com for use in France. > > According to a PDF on the site, the reason this card gets bad > reviews is > that "the Silicon labs Si3012/Si3035 DAA chip used in the original > Digium > X100P card and low cost X100P clone cards only supports FCC mode. > However, > the Si3014/Si3034 DAA chip used on the X100P SE supports global line > standards." > > As for software, "the Silicon labs Si3014/Si3034 DAA chip used in > the X100P > SE supports 600 Ohm impedance and complex impedance to meet CTR21 line > standards. However, the Zaptel wcfxo driver only supports CTR21 mode > with > 600 Ohm AC termination, which may or may not be the correct setting > depending on the country and the phone system in use." > > So... does someone know if OpenZap, which is apparently required in > addition > to Zaptel/Dahdi for FreeSwitch to work PCI TDM cards, supports CTR21? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/99ef82d2/attachment.html From frank at carmickle.com Thu Nov 5 14:20:18 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 5 Nov 2009 17:20:18 -0500 Subject: [Freeswitch-users] [ERR] mod_portaudio.c:974 Cannot find an input device Message-ID: <20091105222017.GP10757@base.carmickle.com> Hello I updated to 15376 added some build depends and still no joy. Any more pointers. Thanks. --FC From Russell.Mosemann at cune.org Thu Nov 5 14:20:31 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Thu, 5 Nov 2009 22:20:31 -0000 Subject: [Freeswitch-users] DAHDI issue Message-ID: <20091105222031.562F2424FBD@mail.cune.org> Debian 5.0.3 FreeSWITCH Version 1.0.trunk (15376M) openzap and libpri-1.4.10.2 dahdi-linux-complete-2.2.0.2+2.2.0 Digium Wildcard TE110P T1/E1 Card (running as a T1) This was working with zaptel. I thought that I would upgrade from zaptel to DAHDI, but it's generating "no such device or address" errors. FS is running as root but can't seem to see the channels. I have unloaded and loaded the drivers. Permissions look fine. The dahdi tools can see the card. Any insights? http://pastebin.freeswitch.org/11016 -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From Russell.Mosemann at cune.org Thu Nov 5 14:42:45 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Thu, 5 Nov 2009 22:42:45 -0000 Subject: [Freeswitch-users] DAHDI issue Message-ID: <20091105224245.4CC6F421B07@mail.cune.org> > I thought that I would upgrade from zaptel to DAHDI, After I send the message, the answer comes to me. I guess that's the way things work. :-) I had forgotten to define the channels in /etc/dahdi/system.conf. Here are the settings, and things are working. Thanks for listening. :-) span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us echocanceller=mg2,1-23 -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From JCasale at activenetwerx.com Thu Nov 5 14:54:27 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 5 Nov 2009 22:54:27 +0000 Subject: [Freeswitch-users] sip profile question Message-ID: The internal.xml also has an "ext-rtp-ip" variable and in trying to understand what this is for (my version of fs is <1) I noticed in trunks conf file it is explained. So the available options that I have given my setup is multihomed with a lan/wan setup where the wan interface is dynamic would be a fqdn for fs to lookup, or auto/auto-nat. How exactly does auto and auto-nat work so I may know of its going to work properly/reliably in my scenario. Thanks! jlc From brian at freeswitch.org Thu Nov 5 15:00:10 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:00:10 -0600 Subject: [Freeswitch-users] sip profile question In-Reply-To: References: Message-ID: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> auto-nat tries to use upnp/nat-pmp to figure it out... auto will just put your IP in there. The other values can be stun:host or an IP. The docs in trunk show this now... its really simple to understand but you should NEVER have to set that unless you have a nat scenario that requires you to lie about your IP and such to traverse the nat. /b On Nov 5, 2009, at 4:54 PM, Joseph L. Casale wrote: > The internal.xml also has an "ext-rtp-ip" variable and in trying to > understand what this is for (my version of fs is <1) I noticed in > trunks > conf file it is explained. So the available options that I have given > my setup is multihomed with a lan/wan setup where the wan interface is > dynamic would be a fqdn for fs to lookup, or auto/auto-nat. > > How exactly does auto and auto-nat work so I may know of its going to > work properly/reliably in my scenario. > > Thanks! > jlc From jerry.richards at teotech.com Thu Nov 5 15:00:50 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 5 Nov 2009 15:00:50 -0800 Subject: [Freeswitch-users] Bug in Freeswitch/scripts/gentls_cert.in build file? In-Reply-To: <19C997BDE7BB44419A2B3A1BBFEA1643@greyhawk.tonecommander.com> References: <19C997BDE7BB44419A2B3A1BBFEA1643@greyhawk.tonecommander.com> Message-ID: <776FA39C275F417B90D423E4F0866F09@greyhawk.tonecommander.com> Here is what is believed to be a bug found by Robert Hadley found in Freeswitch1.0.4/scripts/gentls_cert.in build file: Fix for "gentls_cert remove" to work: [scripts]# diff gentls_cert.in gentls_cert.in~ 129c129 < if [ -d "${CONFDIR}/CA" ]; then --- > if [ ! -d "${CONFDIR}/CA" ]; then Best Regards, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/41115d6a/attachment.html From brian at freeswitch.org Thu Nov 5 15:06:01 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:06:01 -0600 Subject: [Freeswitch-users] Bug in Freeswitch/scripts/gentls_cert.in build file? In-Reply-To: <776FA39C275F417B90D423E4F0866F09@greyhawk.tonecommander.com> References: <19C997BDE7BB44419A2B3A1BBFEA1643@greyhawk.tonecommander.com> <776FA39C275F417B90D423E4F0866F09@greyhawk.tonecommander.com> Message-ID: <3F79E299-417E-461A-9DB8-10852D90B6BC@freeswitch.org> In the future please post issues to jira.freeswitch.org along with a diff -u from the root freeswitch source directory. This already seems to be fixed in svn trunk can you verify. Thanks, Brian On Nov 5, 2009, at 5:00 PM, Jerry Richards wrote: > > Here is what is believed to be a bug found by Robert Hadley found > in Freeswitch1.0.4/scripts/gentls_cert.in build file: > > Fix for "gentls_cert remove" to work: > [scripts]# diff gentls_cert.in gentls_cert.in~ > 129c129 > < if [ -d "${CONFDIR}/CA" ]; then > --- > > if [ ! -d "${CONFDIR}/CA" ]; then > > > Best Regards, > Jerry > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/69e8bd52/attachment.html From JCasale at activenetwerx.com Thu Nov 5 15:20:41 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 5 Nov 2009 23:20:41 +0000 Subject: [Freeswitch-users] sip profile question In-Reply-To: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> References: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> Message-ID: >auto-nat tries to use upnp/nat-pmp to figure it out... auto will just >put your IP in there. > >The other values can be stun:host or an IP. > >The docs in trunk show this now... its really simple to understand but >you should NEVER have to set that unless you have a nat scenario that >requires you to lie about your IP and such to traverse the nat. Thanks for the fast reply Brian, so bear with me here... I am just about to go live w/ my first fs box as I move away from a year or two with Asterisk. So I don't have upnp/nat-pmp, I guess "auto" would be my next choice, but if the box is multihomed, how does it decide which of the two (well more as I am going to use vlans) ip's to stick in there? I guess I could use a public stun server, but if there is a self contained way for me to handle it, I would rather do that so that I don't have to worry about someone else's stun server being up so my fs box functions. Thanks! jlc From brian at freeswitch.org Thu Nov 5 15:27:41 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:27:41 -0600 Subject: [Freeswitch-users] sip profile question In-Reply-To: References: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> Message-ID: <5897F88C-7691-4B38-B51B-4403745C2573@freeswitch.org> If you're on a public IP you have no need for ext-rtp-ip or ext-sip-ip REMOVE them. If your multi homed then you'll need to set them.. we don't listen on 0.0.0.0 you'll have to start a profile for each IP you wish to listen on. /b On Nov 5, 2009, at 5:20 PM, Joseph L. Casale wrote: > > Thanks for the fast reply Brian, so bear with me here... I am just > about > to go live w/ my first fs box as I move away from a year or two with > Asterisk. > > So I don't have upnp/nat-pmp, I guess "auto" would be my next > choice, but if > the box is multihomed, how does it decide which of the two (well > more as I > am going to use vlans) ip's to stick in there? > > I guess I could use a public stun server, but if there is a self > contained > way for me to handle it, I would rather do that so that I don't have > to worry > about someone else's stun server being up so my fs box functions. > > Thanks! > jlc From carlos.talbot at gmail.com Thu Nov 5 15:28:14 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Thu, 5 Nov 2009 17:28:14 -0600 Subject: [Freeswitch-users] FusionPBX Message-ID: <5800526b0911051528p4ed099bdy112776128681477f@mail.gmail.com> FYI, the latest Windows SVN build now includes the option to configure FusionPBX, a port of the pfsense/FreeSWITCH gui: http://fusionpbx.com/index.php If you plan to install it someplace other than the default location of C:/FreeSWITCH just make sure to update the paths in "Admin, System Settings" from the FusionPBX web interface. The default username for the GUI is *admin*, password *fusionpbx* Here's the link: http://files.freeswitch.org/windows_installer/freeswitch.exe At this time FusionPBX utilizes sqlite for its data store. The author, mcrane, plans to release a new version soon with support for a MySQL, or PostgreSQL backend. regards, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/da306ac7/attachment.html From JCasale at activenetwerx.com Thu Nov 5 15:40:57 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 5 Nov 2009 23:40:57 +0000 Subject: [Freeswitch-users] evaluate variable through cli Message-ID: How does one show the assigned value that a variable such as $${local_ip_v4} or $${domain} might have through the cli? Thanks, jlc From mrene_lists at avgs.ca Thu Nov 5 15:43:11 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 5 Nov 2009 15:43:11 -0800 Subject: [Freeswitch-users] evaluate variable through cli In-Reply-To: References: Message-ID: global_getvar local_ip_v4 Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 5-Nov-09, at 3:40 PM, Joseph L. Casale wrote: > How does one show the assigned value that a variable such as > $${local_ip_v4} or $${domain} might have through the cli? > > Thanks, > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jerry.richards at teotech.com Thu Nov 5 15:49:30 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 5 Nov 2009 15:49:30 -0800 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP Message-ID: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry From JCasale at activenetwerx.com Thu Nov 5 15:45:40 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 5 Nov 2009 23:45:40 +0000 Subject: [Freeswitch-users] sip profile question In-Reply-To: <5897F88C-7691-4B38-B51B-4403745C2573@freeswitch.org> References: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> <5897F88C-7691-4B38-B51B-4403745C2573@freeswitch.org> Message-ID: >If you're on a public IP you have no need for ext-rtp-ip or ext-sip-ip >REMOVE them. If your multi homed then you'll need to set them.. we >don't listen on 0.0.0.0 you'll have to start a profile for each IP you >wish to listen on. I am multihomed, and the wan nic is dynamic. Is there any way for me to control how it guesses the IP of a `specific` interface without the use of a third party (stun etc). Thanks, jlc From brian at freeswitch.org Thu Nov 5 15:57:18 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:57:18 -0600 Subject: [Freeswitch-users] sip profile question In-Reply-To: References: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> <5897F88C-7691-4B38-B51B-4403745C2573@freeswitch.org> Message-ID: <68EF3129-6E7B-488A-BD7F-2CFBB6EE7EC3@freeswitch.org> Just use ${local_ip_v4} then. and enable auto-restart on the sofia.conf.xml /b On Nov 5, 2009, at 5:45 PM, Joseph L. Casale wrote: > I am multihomed, and the wan nic is dynamic. Is there any way for me > to > control how it guesses the IP of a `specific` interface without the > use > of a third party (stun etc). > > Thanks, > jlc From brian at freeswitch.org Thu Nov 5 15:58:30 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:58:30 -0600 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP In-Reply-To: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> References: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> Message-ID: <23D4A048-8D9F-49A5-B86E-C1CA2B8FAFDB@freeswitch.org> This all depends on what you're doing in your dialplan if you do stuff like record it requires media and will trigger it. A sip trace or some such debug would be more helpful then a terse description of a problem. /b On Nov 5, 2009, at 5:49 PM, Jerry Richards wrote: > > I am trying to make a call through a Gateway that sends the INVITE > with no > SDP and ONLY wants the 200 OK w/SDP when the callee answers. > > For some reason, Freeswitch answers the call with 200 OK w/SDP even > before > the callee answers the phone. Is this to provide ringback? Can I > disable > that action? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Thu Nov 5 15:58:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:58:51 -0600 Subject: [Freeswitch-users] evaluate variable through cli In-Reply-To: References: Message-ID: <39C7F95D-477F-40B5-AAA1-34B80A0C2A94@freeswitch.org> vars.xml but watch out the core will trump local_ip_v4 if it happens to change. /b On Nov 5, 2009, at 5:40 PM, Joseph L. Casale wrote: > How does one show the assigned value that a variable such as > $${local_ip_v4} or $${domain} might have through the cli? > > Thanks, > jlc > > ______ From dujinfang at gmail.com Thu Nov 5 18:04:45 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Nov 2009 10:04:45 +0800 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <7b197bef0911051003t13e363edqff7b76ecfc099ed5@mail.gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> <23f91030911050957m796fe88fj5da881875c010e6b@mail.gmail.com> <7b197bef0911051003t13e363edqff7b76ecfc099ed5@mail.gmail.com> Message-ID: <23f91030911051804p31d73789o7e3a14c43c857eb1@mail.gmail.com> 2009/11/6 Giovanni Maruzzelli > On Thu, Nov 5, 2009 at 6:57 PM, Seven Du wrote: > > Ciao Giovanni, > > > > Do you still plan to merge this? > > Sorry Seven, > > I've lost track of this, and now I'm so sick I'm completely un-useful ;). > > That's OK, we all have a lot of things to do each day. > But yes, I would like to do it, if you think it is in a useful state. > > Can you please create a Jira and attach an svn diff, so in the next > days I can merge it? > > I'd like to create a jira and I think it would be easier if you can directly merge from branch. However the branch is a bit old and it would need some days if you need svn diff based on the current trunk. Thanks. > -giovanni > > > > > 2009/9/5 Giovanni Maruzzelli > >> > >> Seven, > >> > >> thanks a lot for your efforts. > >> > >> I will merge it in the next days, and I will take care that it will > >> not breaks Windows or Linux. > >> > >> If I find problems I will wait for you having more time in the future. > >> > >> I send you my super best wishes for your personal things to go well > >> and solves in the best of the possible ways. > >> > >> ciao for now, > >> > >> -giovanni > >> > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> > >> > >> > >> On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: > >> > gm, > >> > > >> > Thanks a lot you can merge into the mainline. I check into my branch > >> > because it's currently not as useful as on Linux and Windows and the > >> > solution is not good. But it works and it is a good start that > >> > mod_skypiax runs on OSX. Sure it would be easier for people want to > >> > test and improve it if it been merged into trunk. I think you can make > >> > a diff by > >> > > >> > svn diff -r 14472:14772 > >> > > http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax > >> > > >> > FYI for personal reason I won't have much time put on this in the > >> > coming month. Actually the code was done a few weeks ago but i only > >> > got a chance to commit it yesterday. Sure that is not to say I cannot > >> > do but fixes. But can you please make sure it won't break Linux/ > >> > windows build when you merge the code? I haven't have a chance to test > >> > all of them yet. > >> > > >> > -7- > >> > > >> > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: > >> >> Seeeeeeeven! > >> >> > >> >> I saw the modification you made on the wiki page... > >> >> > >> >> You made it, mod_skypiax runs on OSX!!!! > >> >> > >> >> Let's merge your mods on the mainline, pleaaaase ;-))) > >> >> > >> >> -giovanni > >> >> > >> >> > >> >> > >> >> > >> >> Sincerely, > >> >> > >> >> Giovanni Maruzzelli > >> >> Cell : +39-347-2665618 > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/ff1b4043/attachment.html From msc at freeswitch.org Thu Nov 5 19:36:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Nov 2009 19:36:07 -0800 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> Message-ID: <87f2f3b90911051936k40fee09ds7d0bd237094d1df1@mail.gmail.com> If you need a really cheap entry-level phone that does Polycom's HD Siren codecs then check out the IP 335 that just came out. It's very basic but I'm hearing good things from people who've used them. -MC On Thu, Nov 5, 2009 at 6:57 AM, Rob Forman wrote: > Hey all, > > Looking at buying some high def phones. Any recommendations > (preferably based on experience) for hardware based on product > quality, standards compliance, features integration with Freeswitch, > etc? > > Thank you! > Rob Forman > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/1aeee1ef/attachment-0001.html From JCasale at activenetwerx.com Thu Nov 5 20:26:36 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Fri, 6 Nov 2009 04:26:36 +0000 Subject: [Freeswitch-users] sip profile question In-Reply-To: <68EF3129-6E7B-488A-BD7F-2CFBB6EE7EC3@freeswitch.org> References: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> <5897F88C-7691-4B38-B51B-4403745C2573@freeswitch.org> <68EF3129-6E7B-488A-BD7F-2CFBB6EE7EC3@freeswitch.org> Message-ID: >Just use ${local_ip_v4} then. > >and enable auto-restart on the sofia.conf.xml Cool, it seems to always use the public ip, quite reliably. That is what I am after (why), is there something in the code that forces it to favor for example, non RFC 1918 addresses? It works, I just want to understand exactly how and why rather than be oblivious:) Thanks for all the advice! jlc From mctch at yahoo.com Thu Nov 5 21:02:21 2009 From: mctch at yahoo.com (Mark Crane) Date: Thu, 5 Nov 2009 21:02:21 -0800 (PST) Subject: [Freeswitch-users] FusionPBX In-Reply-To: <5800526b0911051528p4ed099bdy112776128681477f@mail.gmail.com> Message-ID: <616176.22865.qm@web56404.mail.re3.yahoo.com> Screenshots for the FusionPBX graphical interface http://fusionpbx.com/files/fusionpbx_com/screenshots/index.php --- On Thu, 11/5/09, Carlos Talbot wrote: From: Carlos Talbot Subject: [Freeswitch-users] FusionPBX To: freeswitch-users at lists.freeswitch.org Date: Thursday, November 5, 2009, 4:28 PM FYI, the latest Windows SVN build now includes the option to configure FusionPBX, a port of the pfsense/FreeSWITCH gui:?http://fusionpbx.com/index.php If you plan to install it someplace other than the default location of C:/FreeSWITCH just make sure to update the paths in "Admin, System Settings" from the FusionPBX web interface. The default username for the GUI is admin, password fusionpbx Here's the link:?http://files.freeswitch.org/windows_installer/freeswitch.exe At this time FusionPBX utilizes sqlite for its data store. The author, mcrane, plans to release a new version soon with support for a MySQL, or PostgreSQL backend. regards, Carlos -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/34c2a75e/attachment.html From lakindia89 at gmail.com Thu Nov 5 22:29:02 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Fri, 6 Nov 2009 11:59:02 +0530 Subject: [Freeswitch-users] Events in mod_perl Message-ID: <7d79b3930911052229k2828ff7dic6d5b887a4897c8c@mail.gmail.com> Hi all, Is there any way to receive events while running a perl program with the help of mod_perl?? I've seen some functions related to sending and receiving events in the mod_perl wiki. But I don't know how to use that. Any help!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/5300ed7c/attachment.html From msc at freeswitch.org Fri Nov 6 01:44:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Nov 2009 01:44:26 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conf Call - Nov 6 Message-ID: <87f2f3b90911060144h1a5e4e96id53fab67d26d5ec1@mail.gmail.com> FYI, The agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_11_06 It's a light agenda this week. Also, I'm out of town and will only be on the call for about an hour before I have to drop off. I'll be checking in off and on. Everyone is welcome to bring items for discussion. Talk to you tomorrow. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/bd6efbc7/attachment.html From claudiu at globtel.ro Fri Nov 6 02:27:00 2009 From: claudiu at globtel.ro (Claudiu Filip) Date: Fri, 6 Nov 2009 12:27:00 +0200 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP In-Reply-To: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> References: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> Message-ID: <1766871856.20091106122700@globtel.ro> Hi Jerry, Have a look at 3pcc-enable option in your sip profile. You may want to set it "proxy", even if it's not that RFC compliant and has some issues with codec negotiation (FS advertise global_codecs to both parties and it may result in having different codecs on each leg => transcoding or call drop if transcoding not possible). Best regards, Claudiu Filip claudiu at departamentul.it Friday, November 6, 2009, 1:49:30 AM, you wrote: Jerry> I am trying to make a call through a Gateway that sends the INVITE with no Jerry> SDP and ONLY wants the 200 OK w/SDP when the callee answers. Jerry> For some reason, Freeswitch answers the call with 200 OK w/SDP even before Jerry> the callee answers the phone. Is this to provide ringback? Can I disable Jerry> that action? Jerry> Best Regards, Jerry> Jerry Jerry> _______________________________________________ Jerry> FreeSWITCH-users mailing list Jerry> FreeSWITCH-users at lists.freeswitch.org Jerry> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users Jerry> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users Jerry> http://www.freeswitch.org From mattdfong at gmail.com Fri Nov 6 04:32:59 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 6 Nov 2009 19:32:59 +0700 Subject: [Freeswitch-users] uuid_broadcast (with mux) or simultanous uuid_displace? Message-ID: <4256bf830911060432r7ca79601x220b050598b28929@mail.gmail.com> I have an application with two channels bridged, and I want to play an audio (.wav) file to both of the channels but have the ability to combine audio sources so the two participants can talk over the audio being played. when I tried to do this with uuid_broadcast specifying --both legs, it did not mix the audio. I believe this can be done with uuid_displace with the mux argument, but uuid_displace only works on one channel, (there is no --both argument). So, is it possible to execute uuid_displace twice at the same time, one for each uuid? If so, is there a single command I can give to FreeSWITCH that will combine 2 commands to be executed simultaneously? (like a & in linux) Or is there another way of performing what I'm trying to do that I'm over looking? Thanks. --matt Hello Hunter Predictive Dialer - http://www.hellohunter.com - Voice Broadcasting -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/2baffa41/attachment.html From stevendt at primrosebank.net Fri Nov 6 06:20:19 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Fri, 6 Nov 2009 14:20:19 -0000 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo Wiki References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com><2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com><1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com><665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com><87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com> <97FBB4B6002848BCA4F2D89F13626754@bp1.ad.bp.com> Message-ID: Hi, I am having some (limited) success with setting up my system ! I have got an SPA-3102 connected and working after a fashion, but I don't understand how to move on. I am setting up an internal only VOIP system - it will (hopefully) use the SPA3102 to take calls from the PSTN and put them through FreeSwitch to transfer them to VOIP. The gateway will work in reverse, taking internal VOIP calls and passing them to the PSTN. So far, based on the "SPA3102 FreeSwitch HowTo Wiki" and some related information, I can get FreeSwitch to see the incoming call and pass it to the extention defined in the SPA3102 Dialplan, i.e., it rings extension 1001. I can make outgoing calls by dialing the PSTN Extension (1000) and then manually entering the PSTN number. I have an "out of the box" FreeSwitch installation, including extensions, dialplans etc. As well as being new to FreeSwitch, all of this phone stuff is new to me - Dialplans so far look like a black art ! My questions are :- Making outgoing calls. Do I need to enter 1000 everytime I make a call - I'm thinking that I should be able to setup a dialplan which knows that if I'm not entering an internal (VOIP) number (format say, xxxx), it should automatically reroute to the SPA3102 through extension 1000? I probably want to set it up so that the user maybe has to dial a number, say 9, for the outside line, but not the 1000 number. When I try appending a "real" number to 1000, I get a "CALL_REJECTED" error in the console and a "bad number" tone, I suspect that the tone is coming from FreeSwitch and not the SPA3102 - would that be right ? Receiving Incoming Calls As shown in the Wiki, the SPA3102 rings phone extension 1001. I want all internal VOIP phones to ring and be available for answer from all phones - again, I think this should be possible, but I have no idea how to achieve it. I'm guessing that I'd create a "dummy" extension that the SPA3102 would call, which a dialplan would then distribute to a group of VOIP extensions ? Any help/pointers would be really appreciated. I will probably try IRC later, but with the time difference, it's a bit awkward for me (I'm in the UK) - I can't face many more late nights ! Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/ad2cef33/attachment-0001.html From stevendt at primrosebank.net Fri Nov 6 06:20:33 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Fri, 6 Nov 2009 14:20:33 -0000 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo Wiki References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com><2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com><1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com><665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com><87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com> <97FBB4B6002848BCA4F2D89F13626754@bp1.ad.bp.com> Message-ID: <41A5CF92E4E94547BCE301E0F5A5B79B@bp1.ad.bp.com> Hi, I am having some (limited) success with setting up my system ! I have got an SPA-3102 connected and working after a fashion, but I don't understand how to move on. I am setting up an internal only VOIP system - it will (hopefully) use the SPA3102 to take calls from the PSTN and put them through FreeSwitch to transfer them to VOIP. The gateway will work in reverse, taking internal VOIP calls and passing them to the PSTN. So far, based on the "SPA3102 FreeSwitch HowTo Wiki" and some related information, I can get FreeSwitch to see the incoming call and pass it to the extention defined in the SPA3102 Dialplan, i.e., it rings extension 1001. I can make outgoing calls by dialing the PSTN Extension (1000) and then manually entering the PSTN number. I have an "out of the box" FreeSwitch installation, including extensions, dialplans etc. As well as being new to FreeSwitch, all of this phone stuff is new to me - Dialplans so far look like a black art ! My questions are :- Making outgoing calls. Do I need to enter 1000 everytime I make a call - I'm thinking that I should be able to setup a dialplan which knows that if I'm not entering an internal (VOIP) number (format say, xxxx), it should automatically reroute to the SPA3102 through extension 1000? I probably want to set it up so that the user maybe has to dial a number, say 9, for the outside line, but not the 1000 number. When I try appending a "real" number to 1000, I get a "CALL_REJECTED" error in the console and a "bad number" tone, I suspect that the tone is coming from FreeSwitch and not the SPA3102 - would that be right ? Receiving Incoming Calls As shown in the Wiki, the SPA3102 rings phone extension 1001. I want all internal VOIP phones to ring and be available for answer from all phones - again, I think this should be possible, but I have no idea how to achieve it. I'm guessing that I'd create a "dummy" extension that the SPA3102 would call, which a dialplan would then distribute to a group of VOIP extensions ? Any help/pointers would be really appreciated. I will probably try IRC later, but with the time difference, it's a bit awkward for me (I'm in the UK) - I can't face many more late nights ! Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/a999ee69/attachment.html From rob4manhere at gmail.com Fri Nov 6 06:32:07 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Fri, 6 Nov 2009 08:32:07 -0600 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <87f2f3b90911051936k40fee09ds7d0bd237094d1df1@mail.gmail.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <87f2f3b90911051936k40fee09ds7d0bd237094d1df1@mail.gmail.com> Message-ID: <993FA8DA-7A3A-4FF7-BA33-168CFF96B4FC@gmail.com> Great- good to know. Thanks for all the responses! Rob On Nov 5, 2009, at 9:36 PM, Michael Collins wrote: > If you need a really cheap entry-level phone that does Polycom's HD > Siren codecs then check out the IP 335 that just came out. It's very > basic but I'm hearing good things from people who've used them. > -MC > > On Thu, Nov 5, 2009 at 6:57 AM, Rob Forman > wrote: > Hey all, > > Looking at buying some high def phones. Any recommendations > (preferably based on experience) for hardware based on product > quality, standards compliance, features integration with Freeswitch, > etc? > > Thank you! > Rob Forman > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/aa3ed3a1/attachment.html From mariusz_kolo at wp.pl Fri Nov 6 06:38:31 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=) Date: Fri, 06 Nov 2009 15:38:31 +0100 Subject: [Freeswitch-users] Problem with hangin bri In-Reply-To: <4AEEC368.5020204@wp.pl> References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com> <87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com> <4AE8B3B6.9000106@wp.pl> <87f2f3b90910281528t633b765x7c2bb858f41ce154@mail.gmail.com> <4AEEC368.5020204@wp.pl> Message-ID: <4AF434E7.6070904@wp.pl> Hi I'm testing behaviour on our Bri Card and i see that only incoming calls hangin up channels pastebin: http://pastebin.freeswitch.org/11020 At start oz dump for 1 and 2 channels are: API CALL [oz(dump 1 1)] output: span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE API CALL [oz(dump 1 2)] output: span_id: 1 chan_id: 2 physical_span_id: 1 physical_chan_id: 2 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE after first incoming call, system chose 2 channel (in log: 2009-11-06 14:46:34.341595 .... Processing 609381316->717949433 ....) we have: API CALL [oz(dump 1 2)] output: span_id: 1 chan_id: 2 physical_span_id: 1 physical_chan_id: 2 type: B state: HANGUP last_state: PROGRESS cid_date: cid_name: 609381316 cid_num: 609381316 ani: 609381316 aniII: dnis: 717949433 rdnis: cause: NORMAL_CLEARING after second incoming call, system chose 1 channel propably because 2 channel is still in use (in log: 2009-11-06 14:47:07.361596 .... Processing 609381316->717949433 ...) we have: API CALL [oz(dump 1 1)] output: span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: B state: HANGUP last_state: PROGRESS cid_date: cid_name: 609381316 cid_num: 609381316 ani: 609381316 aniII: dnis: 717949433 rdnis: cause: NORMAL_CLEARING All next incoming calls has warning 2009-11-06 14:47:40.093836 [WARNING] ozmod_isdn.c:829 Channel 1:2 ~ 1:2 is already in use waiting for it to become available. All next outbound calls has: Warning 2009-11-06 14:48:16.925599 [ERR] mod_openzap.c:1154 No channels available Only unload mod_openzap and load mod_openzap release channels I think channel's state after call should be DOWN not HANGUP If i make only outgoing calls channel's state returns to DOWN Please check it out. I hope my logs can help Thanks Mariusz Ko?odziejczyk pisze: > Hi > > pastebin: > > http://pastebin.freeswitch.org/10926 > and > http://pastebin.freeswitch.org/10927 > > .We invoke calls from one voip phone to cell phone, and vice versa, but > when i make inbound and outbound connection in nearly same time > something goes wrong with chanells > > Thanks > > > > Michael Collins pisze: > >> Thanks. Can you collect debug logs of this happening? See >> http://wiki.freeswitch.org/wiki/Reporting_Bugs for helpful tips on >> collecting debug information. Use pastebin to dump all the log info >> and reply here with the link. We don't have too many BRI users but I >> believe there are a few so hopefully we can help you get up and running. >> -MC >> >> 2009/10/28 Mariusz Ko?odziejczyk > > >> >> Hi >> >> I'm also working on this project, so i can answer your questions >> >> Which version of FreeSWITCH are you running? >> >> FreeSWITCH Version 1.0.trunk (15246) >> >> Which PRI library are you using? >> openzap Native stack >> >> openzap.conf >> >> [span zt BRI1] >> trunk_type => bri >> b-channel => 1-2 >> d-channel=> 3 >> >> openzap.conf >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Which BRI card are you using? >> >> Producer: http://www.phoniceq.com/ >> card model: http://quadbri.phoniceq.com/ >> >> Card instalation process (instruction from producer) >> >> 1) download bristuff staff from >> >> http://junghanns.net/downloads/bristuff-0.4.0-RC3h.tar.gz >> or >> http://junghanns.net/downloads/bristuff-0.3.0-PRE-1y-z.tar.gz >> >> unpack it and go to bristuff-* >> >> 2) download patcher from >> http://quadbri.phoniceq.com/driver/bristuff/qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch >> >> patch it using >> >> patch -p0 < qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch >> >> 3) you can check card using zttest (result should be 99.x) >> >> Producer has said, that we are first client, it wants to use this >> card in freeswitch >> >> we are using 1 port (S/T interface). Our NT is "NT1 plus 2b1q" >> >> >> Thanks >> >> Michael Collins pisze: >> > Okay, obligatory questions: >> > Which version of FreeSWITCH are you running? >> > Which PRI library are you using? >> > Which BRI card are you using? >> > >> > -MC >> > >> > On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawli?ski >> >> > >> wrote: >> > >> > Hi, >> > I have some problems with bri status. I have 3 chanel isdn >> modem, >> > and zaptel compatible quad bri card. I can invoke calls from my >> > voip phone to cell phone, and vice versa, but when i make >> inbound >> > and outbound connection in nearly same time something goes wrong >> > with chanells and after few calls all of them has hangup status. >> > >> > There is log about that in attachement, see "is already in use >> > waiting for it to become available." phrase. Time of this >> event is >> > about 14:43:35. Unload and Load open_zap module helped, but its >> > not an solution because of lost connections. >> > >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> -- >> Mariusz Ko?odziejczyk >> >> Advanced Developing Architecture S.C. >> >> tel. : +48 609 381 316 >> e-mail : mariusz_kolo at wp.pl >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- Mariusz Ko?odziejczyk Advanced Developing Architecture S.C. tel. : +48 609 381 316 e-mail : mariusz_kolo at wp.pl From gshfreesw at gmail.com Fri Nov 6 07:11:34 2009 From: gshfreesw at gmail.com (Shameem Shiek) Date: Fri, 6 Nov 2009 10:11:34 -0500 Subject: [Freeswitch-users] Register multiple DID/extensions with the same provider. Message-ID: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> Hello, I have created a provider configuration file XML file *in conf/directory/default/myprovider.xml . *This configuration has the line: How do I add another DID from the same provider? Do I add another line like this ? or create a new provider XML config file? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/f877531d/attachment.html From brian at freeswitch.org Fri Nov 6 07:23:19 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Nov 2009 09:23:19 -0600 Subject: [Freeswitch-users] Register multiple DID/extensions with the same provider. In-Reply-To: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> References: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> Message-ID: <72FFFD32-4A62-474E-A9D5-497660FD380B@freeswitch.org> Does your provider require you to register once for every did? /b On Nov 6, 2009, at 9:11 AM, Shameem Shiek wrote: > Hello, > > I have created a provider configuration file XML file in conf/ > directory/default/myprovider.xml . This configuration has the line: > > > > > How do I add another DID from the same provider? Do I add another > line like this ? or create a new provider XML config file? > > Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/e013af09/attachment-0001.html From rob4manhere at gmail.com Fri Nov 6 07:30:38 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Fri, 6 Nov 2009 09:30:38 -0600 Subject: [Freeswitch-users] Register multiple DID/extensions with the same provider. In-Reply-To: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> References: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> Message-ID: Hi Shameem, If you're just getting another DID under the same provider registration, that should go under your public dialplan (conf/dialplan/ public) and then route to the extension or application of your choice. Rob On Nov 6, 2009, at 9:11 AM, Shameem Shiek wrote: > Hello, > > I have created a provider configuration file XML file in conf/ > directory/default/myprovider.xml . This configuration has the line: > > > > > How do I add another DID from the same provider? Do I add another > line like this ? or create a new provider XML config file? > > Thanks! > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/9660f825/attachment.html From ujjval at simplesignal.com Fri Nov 6 07:59:42 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Fri, 6 Nov 2009 07:59:42 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175B572244@EXMBXCLUS01.citservers.local> Any examples I can refer to for this? Like for Channel vars and execute_application calls? Does this all need to be doen in dialplan.public.xml or also in other config files? Sorry: I am still learning the Freeswitch world. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Thursday, November 05, 2009 8:56 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator This is true, BUT it is more flexible than it looks. http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E The caller controls can have a key execute a dialplan extension: execute_application You can set a channel var on the moderator prior to joining to the conf. When the extenion is called, you can check the channel var for moderator and act accordingly. Or you can send an event and monitor with an app over ESL and do whatever you want there (probably using the same channel var trick for knowing who is a mod or not). On Thu, Nov 5, 2009 at 8:52 AM, Rob Forman wrote: > Hi UK, > > ?From what I've done and read, the caller-controls (in > conference.conf.xml) can be modified to almost anything you can think > of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a > caller control just for those with the moderator flag. ?So unless you > want everyone able to mute/kick everyone then you can't do it. > > The wiki seems to indicate this as well: > > "Be aware that the caller-controls are applied across the entire > conference. You cannot enter one member of the conference using caller- > controls ABC and then enter a second member using caller-controls XYZ." > > http://wiki.freeswitch.org/wiki/Mod_conference > > > I think this might be a limitation of mod_conference. ?Perhaps one of > the pros can chime in if I'm off-base or there's some nifty way to > accomplish this. > > Cheers, > Rob > > On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > >> Any ideas on the below...has anyone implemented the below: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, November 02, 2009 12:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Rob: >> >> ? Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 9:34 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hm, strange. ?I haven't seen that before. ?Can you pastebin your logs >> at debug level? >> >> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >> >>> It's strange... a tcpdump tells me that there is no DTMF from my >>> provider when using IVR, but when I call into a TN that goes >>> directly into the Conference App, I see DTMF from the provider. >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 7:23 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I've never had any problem with that. ?Is your logging at debug level >>> so you can see the RECV DTFM in the log/fs_cli? ?Are you calling from >>> a SIP phone on the pbx, or via a PSTN provider? ?Maybe your provider >>> isn't passing them through. >>> >>> Make sure your logging is turned up then try something simpler, like >>> calling the echo application, and see if DTFM comes through. >>> >>> Rob >>> >>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>> >>>> Rob: >>>> >>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>> using the below configuration....so it basically timesout. >>>> >>>> UK >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Ujjval Karihaloo >>>> Sent: Monday, October 26, 2009 9:21 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>> correct.. >>>> =============== >>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. ?I guess that's why I didn't understand >>>> the point of the +pin. >>>> ====================== >>>> >>>> I will try it out. >>>> >>>> Again thx a lot for your help. Will keep everyone posted. >>>> >>>> ________________________________________ >>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>> Sent: Friday, October 23, 2009 12:22 PM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> I just re-tested with the pin in my dial plan: >>>> >>>> >>>> >>>> And it doesn't challenge me for the pin. ?I just drop right in. ?I >>>> figured this is how it was intended, since the wiki says the pin is >>>> set initially and only challenged in later attempts [by future >>>> callers]: >>>> >>>> "The first time a conference name (confname) is used, it will be >>>> created on demand, and the pin will be set to what ever is specified >>>> at that time: the pin in the data string if specified, or if not, >>>> the >>>> "pin" setting in the conference profile, and if that is also >>>> unspecified, then there is no pin protection. Any later attempt to >>>> join the conference must specify the same pin number, if one existed >>>> when it was created. " >>>> >>>> >>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. ?I guess that's why I didn't understand >>>> the point of the +pin. >>>> >>>> I'm sure there's a scenario where its used and useful, the wiki just >>>> doesn't explain it. >>>> >>>> Rob >>>> >>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>> >>>>> Well first off you're not defining a pine here... >>>>> >>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>> pin >>>>> number] >>>>> >>>>> That might be why its not asking for a pin. >>>>> >>>>> /b >>>>> >>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>> >>>>>> ? >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jmesquita at freeswitch.org Fri Nov 6 08:31:09 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 6 Nov 2009 14:31:09 -0200 Subject: [Freeswitch-users] Events in mod_perl In-Reply-To: <7d79b3930911052229k2828ff7dic6d5b887a4897c8c@mail.gmail.com> References: <7d79b3930911052229k2828ff7dic6d5b887a4897c8c@mail.gmail.com> Message-ID: I don't know what you are trying to do exactly but I think that you might need to you ESL instead. Why don't you take a look at all the examples inside ${SVNROOT}/libs/esl and see if that fits you? I have a hunch that it would. JM On Fri, Nov 6, 2009 at 4:29 AM, lakshmanan ganapathy wrote: > Hi all, > Is there any way to receive events while running a perl program with > the help of mod_perl?? > > I've seen some functions related to sending and receiving events in the > mod_perl wiki. But I don't know how to use that. > Any help!!! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/fa2c4b02/attachment.html From andrewkt at aktzero.com Fri Nov 6 09:59:47 2009 From: andrewkt at aktzero.com (Andrew Thompson) Date: Fri, 06 Nov 2009 12:59:47 -0500 Subject: [Freeswitch-users] Register multiple DID/extensions with the same provider. In-Reply-To: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> References: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> Message-ID: <4AF46413.7070905@aktzero.com> On 11/6/2009 10:11 AM, Shameem Shiek wrote: > How do I add another DID from the same provider? Do I add another line > like this ? or create a new provider XML config file? I have two providers that send me DIDs in different ways. One of them does not provide the number that was dialed, so I have to sub-account and register uniquely for each DID. The other does provide the number dialed and I create seperate extensions in conf/dialplan/public/NPANXXXXXX.xml. TLDR: Test inbound calls and see if you get unique destination_number for each DID. -- Andrew Thompson From mrene_lists at avgs.ca Fri Nov 6 10:33:17 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 6 Nov 2009 10:33:17 -0800 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP In-Reply-To: <1766871856.20091106122700@globtel.ro> References: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> <1766871856.20091106122700@globtel.ro> Message-ID: Are you recording? I recall a recent change to force answer whenever record_session is called. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 6-Nov-09, at 2:27 AM, Claudiu Filip wrote: > > > Hi Jerry, > > > Have a look at 3pcc-enable option in your sip profile. You may want > to set it "proxy", even if it's not that RFC compliant and has some > issues with codec negotiation (FS advertise global_codecs to both > parties and it may result in having different codecs on each leg => > transcoding or call drop if transcoding not possible). > > > Best regards, > > Claudiu Filip > claudiu at departamentul.it > > > Friday, November 6, 2009, 1:49:30 AM, you wrote: > Jerry> I am trying to make a call through a Gateway that sends the > INVITE with no > Jerry> SDP and ONLY wants the 200 OK w/SDP when the callee answers. > > Jerry> For some reason, Freeswitch answers the call with 200 OK w/ > SDP even before > Jerry> the callee answers the phone. Is this to provide ringback? > Can I disable > Jerry> that action? > > Jerry> Best Regards, > Jerry> Jerry > > > Jerry> _______________________________________________ > Jerry> FreeSWITCH-users mailing list > Jerry> FreeSWITCH-users at lists.freeswitch.org > Jerry> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > Jerry> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > Jerry> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From codecomplete at free.fr Fri Nov 6 11:16:40 2009 From: codecomplete at free.fr (Fred-145) Date: Fri, 6 Nov 2009 11:16:40 -0800 (PST) Subject: [Freeswitch-users] Does OpenZap support CTR21? In-Reply-To: <7FD19B47-C121-48CD-98C2-2830BFDF1068@jerris.com> References: <26217371.post@talk.nabble.com> <7FD19B47-C121-48CD-98C2-2830BFDF1068@jerris.com> Message-ID: <26230864.post@talk.nabble.com> Thanks for the feedback. I've never installed this type of card with Freeswitch. Am I correct in understanding that I just need to download and compile the latest Zaptel/Dahdi source from the Asterisk web site, and then install Freeswitch and OpenZap, and it'll work (provided the card works with my hardware)? -- View this message in context: http://old.nabble.com/Does-OpenZap-support-CTR21--tp26217371p26230864.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Russell.Mosemann at cune.org Fri Nov 6 12:04:58 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Fri, 6 Nov 2009 20:04:58 -0000 Subject: [Freeswitch-users] Does OpenZap support CTR21? In-Reply-To: <26230864.post@talk.nabble.com> Message-ID: <20091106200458.AACDC3E5BEF@mail.cune.org> Fred-145 said: > I've never installed this type of card with Freeswitch. Am I correct in > understanding that I just need to download and compile the latest > Zaptel/Dahdi source from the Asterisk web site, and then install Freeswitch > and OpenZap, and it'll work (provided the card works with my hardware)? Yes, it should just work. I'd recommend Dahdi (complete), because Zaptel is not being developed anymore. Check the wiki for little things you have to set in various files, such as /etc/dahdi/modules /etc/dahdi/system.conf wherever/freeswitch/conf/openzap.xml wherever/freeswitch/conf/zt.conf (shouldn't have to change it) wherever/freeswitch/autoload_configs/openzap.conf.xml -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From orien at tx.rr.com Fri Nov 6 15:21:23 2009 From: orien at tx.rr.com (Orien Love) Date: Fri, 06 Nov 2009 17:21:23 -0600 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: References: Message-ID: <4AF4AF73.8070804@tx.rr.com> First of all, Thanks to the help I received on my pfSense installation, especially to Michael. I have a basic test system up and running. I am still waiting on some hardware but the base system is working!!!! I am looking on advice on how to set up a simple office PBX, 20 phones and 4 outside lines.with 2 or 3 "operator" phones and the rest will be extensions. Here is my plan, please let me know if it does not make sense, or if I am going about it System Hardware 4 spa3000's to handle the outside lines. 2-3 polycom 601 phones with expansion modules (Operator phones) 18 polycom 330 or other phones for desks. 2-24 port cisco POE switches 1 pfSense server. System Design. Extension Numbers 2xx Outside line access 1xxxxxxxxxx groups 3xx auto-attendent ??? here are my questions #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of memory handle this proposed system? (Here is the MB I am thing of using MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) #2 how do I pool my spa 3000 FXO lines so that the outgoing calls use the first available line? also how do insure that metro (non long distance) calls go to a specific line if available? I have learned a lot on how to set up Polycom 601 phones, I am planning on writing a how to document, is there any specific format? Thanks Orien From stevendt at primrosebank.net Fri Nov 6 15:59:09 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Fri, 6 Nov 2009 23:59:09 -0000 Subject: [Freeswitch-users] Valid Dial Strings References: <4AF4AF73.8070804@tx.rr.com> Message-ID: <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> Hi, can someone pointme to where the valid dialing strings are specified ? I'm assuming that something, somewhere, tells FS that numbers are invalid before they get dialed ? regards Dave From anthony.minessale at gmail.com Fri Nov 6 16:02:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Nov 2009 18:02:06 -0600 Subject: [Freeswitch-users] uuid_broadcast (with mux) or simultanous uuid_displace? In-Reply-To: <4256bf830911060432r7ca79601x220b050598b28929@mail.gmail.com> References: <4256bf830911060432r7ca79601x220b050598b28929@mail.gmail.com> Message-ID: <191c3a030911061602s4d1ca7fh9222460f8c54a951@mail.gmail.com> no, sorry. We have to draw the line somewhere. You are stepping outside the boundary of what you can do in a 2 channel bridge. Transfer them into a conference and do it there. On Fri, Nov 6, 2009 at 6:32 AM, Matthew Fong wrote: > I have an application with two channels bridged, and I want to play an > audio (.wav) file to both of the channels but have the ability to combine > audio sources so the two participants can talk over the audio being played. > when I tried to do this with uuid_broadcast specifying --both legs, it did > not mix the audio. > > I believe this can be done with uuid_displace with the mux argument, but > uuid_displace only works on one channel, (there is no --both argument). So, > is it possible to execute uuid_displace twice at the same time, one for each > uuid? If so, is there a single command I can give to FreeSWITCH that will > combine 2 commands to be executed simultaneously? (like a & in linux) Or is > there another way of performing what I'm trying to do that I'm over looking? > > Thanks. > > --matt > Hello Hunter > Predictive Dialer - http://www.hellohunter.com - Voice Broadcasting > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/e13d7647/attachment.html From anthony.minessale at gmail.com Fri Nov 6 16:08:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Nov 2009 18:08:48 -0600 Subject: [Freeswitch-users] Dialpan: try.. finally analogs In-Reply-To: References: Message-ID: <191c3a030911061608w5be8af61y7bc10fe2d23dfc4a@mail.gmail.com> If you know the reason, why are you so puzzled by it? I think you should not assume you understand what is happening unless you really do. I think you need to provide an exact description of what you are doing so I can explain to you where you are making the mistake. Make sure you are on latest SVN and reproduce this in a console log for us and add an exact description of what you are doing in detail. On Thu, Nov 5, 2009 at 11:44 AM, Artem Shiyanov wrote: > Hello! > > I have to deal with classic problem: "Leaking stream handle" in FS console. > I also know the reason - firstly channel is sent to the extension with > "playback" and later it is transfered to another extensions with > "execute_extension" or, another trouble-case - channel is bridged to some > addres. > I do not ask (but I wish to) why FS doesn't close stream automatically when > channel is gone. > I ask whether it is possible to use some "try.. finally" construction in > diaplan? If "yes" then I can simply stop playback in the "finally" block.. > > Any thoughs? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/f48d592f/attachment.html From lakindia89 at gmail.com Fri Nov 6 21:38:44 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Sat, 7 Nov 2009 11:08:44 +0530 Subject: [Freeswitch-users] Events in mod_perl In-Reply-To: References: <7d79b3930911052229k2828ff7dic6d5b887a4897c8c@mail.gmail.com> Message-ID: <7d79b3930911062138l262db9bema29c620a7b8cef94@mail.gmail.com> Ya. I have done that event processing with ESL. But I wanted to know, whether in mod_perl, we can get the events and process it or not. I've seen function's like events_get etc.. But I don't know how to use those things. In mod_perl if I'm able to get the events, then it will be easier for me. Is it possible!!! 2009/11/6 Jo?o Mesquita > I don't know what you are trying to do exactly but I think that you might > need to you ESL instead. > > Why don't you take a look at all the examples inside ${SVNROOT}/libs/esl > and see if that fits you? I have a hunch that it would. > > JM > > On Fri, Nov 6, 2009 at 4:29 AM, lakshmanan ganapathy > wrote: > >> Hi all, >> Is there any way to receive events while running a perl program with >> the help of mod_perl?? >> >> I've seen some functions related to sending and receiving events in the >> mod_perl wiki. But I don't know how to use that. >> Any help!!! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/9b5f49d3/attachment.html From mayamatakeshi at gmail.com Sat Nov 7 04:47:36 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Sat, 7 Nov 2009 21:47:36 +0900 Subject: [Freeswitch-users] ParkServer: how to publish orbit status Message-ID: <15b9404e0911070447r78b95d9cv7178d0ce735d7d80@mail.gmail.com> Hello, it is my understanding that FreeSWITCH doesn't provide a ParkServer per se. So, to provide for this, I will have an entry in the dialplan to play MOH on the channel continuously till someone retrieve the call. And then, I need to publish a NOTIFY to all subscribed users informing the status of the park orbit. >From the wiki (http://wiki.freeswitch.org/wiki/PRESENCE_IN_event_example), it seems I could use "sendevent PRESENCE_IN" for this, but I was unable to figure out how the message must be sent to fill all required attributes in the xml. >From the examples, I managed to send a NOTIFY with a content like this confirmed but I actually need to send it like this: confirmed So, is it possible to set the lacking attributes? (I've tried to set content-length and pass the xml to sendevent, but nothing changed). br, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/add11ed6/attachment-0001.html From ruda at ruda.com.br Sat Nov 7 05:24:38 2009 From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=) Date: Sat, 7 Nov 2009 10:24:38 -0300 Subject: [Freeswitch-users] Fwd: Microsoft Exchange 2007 - Moved Temporarily In-Reply-To: <7600794b0911070523n41a0351co8a31bfc42128e683@mail.gmail.com> References: <7600794b0911070523n41a0351co8a31bfc42128e683@mail.gmail.com> Message-ID: <7600794b0911070524m191bf5dfmb8e42f8c775b2c29@mail.gmail.com> I'm having a problem when using Microsoft Exchange 2007 with the FreeSWITCH 1.0.trunk Windows on the same server (127.0.0.1) I set everything right and I am analyzing packages Loopback, only when Exchange sends the FreeSWITCH Moved Temporarily not move. He is sending the INVITE at the same time send port to the port that was moved. How can I fix this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/a845e807/attachment.html From rupa at rupa.com Sat Nov 7 05:43:03 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Sat, 7 Nov 2009 05:43:03 -0800 Subject: [Freeswitch-users] Fwd: Microsoft Exchange 2007 - Moved Temporarily In-Reply-To: <7600794b0911070524m191bf5dfmb8e42f8c775b2c29@mail.gmail.com> References: <7600794b0911070523n41a0351co8a31bfc42128e683@mail.gmail.com> <7600794b0911070524m191bf5dfmb8e42f8c775b2c29@mail.gmail.com> Message-ID: I don't believe this is a supported configuration. If you really must run both on the same server, put FS and Exchange on different IP addresses and see if that works. SIP client/server on the same IP has issues. On Sat, Nov 7, 2009 at 5:24 AM, Rud? Cunha wrote: > I'm having a problem when using Microsoft Exchange 2007 with the FreeSWITCH > 1.0.trunk Windows on the same server (127.0.0.1) > I set everything right and I am analyzing packages Loopback, only when > Exchange sends the FreeSWITCH Moved Temporarily not move. He is sending the > INVITE at the same time send port to the port that was moved. > > How can I fix this? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From steve at justfone.com Sat Nov 7 06:26:38 2009 From: steve at justfone.com (Steven Brown) Date: Sat, 7 Nov 2009 14:26:38 +0000 Subject: [Freeswitch-users] leg_delay_start Message-ID: <3e6d7b0c0911070626g32af550fsd80e99d8266a7aa8@mail.gmail.com> Hi I've been trying to experiment with leg_delay_start when bridging to two mobiles via a gateway, however regardless of settings both legs are bridged immediately. I noticed a previous post on problems with leg_delay_start which seemed to go unanswered, just wondered if there is a known issue or if its something I'm doing wrong. Using FS 1.0.3 Dialplan extract as follows : Any pointers appreciated Thanks Steven Brown email steve at justfone.com office 08707706968 mobile 07768755409 fax 07884636663 Justfone - Company Reg. No. : 3926817 Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW The contents of this e-mail may be privileged and are confidential. It may not be disclosed to or used by anyone other than the addressee(s), nor copied in any way. If received in error, please advise sender, then delete it from your system. Internet email communications are not secure and therefore Justfone do not accept legal responsibility for the contents of this message. Any views or opinions presented are solely those of the author and do not necessarily represent those of Justfone unless otherwise specifically stated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/a8086ca8/attachment.html From stevendt at primrosebank.net Sat Nov 7 06:42:09 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Sat, 7 Nov 2009 14:42:09 -0000 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo Wiki - HELP Please! References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com><2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com><1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com><665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com><87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com><97FBB4B6002848BCA4F2D89F13626754@bp1.ad.bp.com> <41A5CF92E4E94547BCE301E0F5A5B79B@bp1.ad.bp.com> Message-ID: <7B9E5C0A81154EDB8B2F2979EAF0BA17@bp1.ad.bp.com> Follow up to previous post..... regarding making outgoing calls. I ***think*** that I have configured a dialplan that allows the user to dial out but the requests seem to be getting rejected by the SPA3102. I can dial 0 and the FreeSwitch attendant will connect to the PSTN line (FreeSwitch reports that the call has been answered). Similarly, I can dial 1000 - the SPA3102 extension number with similar results. However, if a try to dial an external number, the gateway rejects the call. I have captured some of the debug log but the info in there is way over my head, can anyone help me understand what it's telling me please ? regards Dave ----- Original Message ----- From: Dave Stevenson To: freeswitch-users at lists.freeswitch.org Sent: Friday, November 06, 2009 2:20 PM Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo Wiki Hi, I am having some (limited) success with setting up my system ! I have got an SPA-3102 connected and working after a fashion, but I don't understand how to move on. I am setting up an internal only VOIP system - it will (hopefully) use the SPA3102 to take calls from the PSTN and put them through FreeSwitch to transfer them to VOIP. The gateway will work in reverse, taking internal VOIP calls and passing them to the PSTN. So far, based on the "SPA3102 FreeSwitch HowTo Wiki" and some related information, I can get FreeSwitch to see the incoming call and pass it to the extention defined in the SPA3102 Dialplan, i.e., it rings extension 1001. I can make outgoing calls by dialing the PSTN Extension (1000) and then manually entering the PSTN number. I have an "out of the box" FreeSwitch installation, including extensions, dialplans etc. As well as being new to FreeSwitch, all of this phone stuff is new to me - Dialplans so far look like a black art ! My questions are :- Making outgoing calls. Do I need to enter 1000 everytime I make a call - I'm thinking that I should be able to setup a dialplan which knows that if I'm not entering an internal (VOIP) number (format say, xxxx), it should automatically reroute to the SPA3102 through extension 1000? I probably want to set it up so that the user maybe has to dial a number, say 9, for the outside line, but not the 1000 number. When I try appending a "real" number to 1000, I get a "CALL_REJECTED" error in the console and a "bad number" tone, I suspect that the tone is coming from FreeSwitch and not the SPA3102 - would that be right ? Receiving Incoming Calls As shown in the Wiki, the SPA3102 rings phone extension 1001. I want all internal VOIP phones to ring and be available for answer from all phones - again, I think this should be possible, but I have no idea how to achieve it. I'm guessing that I'd create a "dummy" extension that the SPA3102 would call, which a dialplan would then distribute to a group of VOIP extensions ? Any help/pointers would be really appreciated. I will probably try IRC later, but with the time difference, it's a bit awkward for me (I'm in the UK) - I can't face many more late nights ! Regards Dave ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/f2a55514/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Copy of freeswitch.zip Type: application/x-zip-compressed Size: 3248 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/f2a55514/attachment.bin From fsdevlist at seanf.me Sat Nov 7 00:59:57 2009 From: fsdevlist at seanf.me (Sean Ferguson) Date: Sat, 7 Nov 2009 03:59:57 -0500 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote Message-ID: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> FreeSWITCH seems to be unable to read MP3 files, citing that it's an unknown format. Looking through the log, I found this during startup: 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_sync_wrote** There don't seem to be any compile-time errors, yet I can't seem to eliminate this issue. Any help would be appreciated. From john_re at fastmail.us Sat Nov 7 05:52:53 2009 From: john_re at fastmail.us (john_re) Date: Sat, 07 Nov 2009 05:52:53 -0800 Subject: [Freeswitch-users] Hi - Nov 7 TODAY & Nov 22 - Join Global FreeSW GNU(Linux) HW Culture meeting via VOIP - BerkeleyTIP GlobalTIP - For Forwarding Message-ID: <1257601973.13636.1344039469@webmail.messagingengine.com> Hi Anthony, FreeSWITCH list members. :) 1) Anthony, is it you who's in charge of this mail list & group? 2) Starting June 2008 I've been putting together a global Free SW, HW & Culture group, BerkeleyTIP, & GlobalTIP, which meets via VOIP globally. :) It's purpose is educational, productive & social. TIP = Talks, Installfest & Project/Programming Party. I do all this in my small "spare/free" time as a contribution to the communit(ies). {Thanks for all the SW! :) } http://sites.google.com/site/berkeleytip/ The meetings are on the 1st Saturday & 3rd Sunday of each month, from 12N-3PM Pacific usa UTC-8hr time. We are currently looking at moving from Asterisk to freeswitch on our voip server box. http://sites.google.com/site/berkeleytip/remote-attendance I'm writing to invite you all to join with the global BTIP community & attend online. Join #berkeleytip on irc.freenode.net, & we'll help you get on our VOIP conference. [I'm sure you all could probably help us way more than we could help you. ;) ] We also encourage all groups (such as the freeswitch community) to hold a simultaneous meeting, & join together with us all online. I know you already have a friday voip conference, iirc. If all free sw groups had a simultaneous meeting to BerkeleyTIP, it would be like a global meeting/conference - everyone would know that at that time they could meet all their friends & community members online on voip conferences, from all the free sw projects in the world. 3) May I send to this mailing list the monthly BTIPGlobal announcement? This month's announcement is included right below. Best Wishes, John Re =================================================================== CONTENTS: Meeting days/times & Howto - Mark your calendar's dates; Videos; Hot topics; Opportunities; Announcement Flyers; New webpages ===== Come join in with the Global Free SW HW & Culture community at the BerkeleyTIP/GlobalTIP meeting, via VOIP. Two meetings this month: Sat Nov 7, 12Noon - 3PM Pacific Time (=UTC-8) Sun Nov 22, 12Noon - 3PM Pacific Time (=UTC-8) Mark your calendars, 1st Sat, 3rd Sun every month. {Note: 4th Sunday this November, to give 2 week spacing.} Join online #berkeleytip on irc.freenode.net & we'll help you get your voip HW & SW working: http://sites.google.com/site/berkeleytip/remote-attendance Or come to the FreeSpeech Cafe at UC Berkeley in person meeting. Join the global mailing list http://groups.google.com/group/BerkTIPGlobal I hope to see you there. :) ===== Talk Videos for November 2009: Django Development - Richard Kiss, Eddy Mulyono, Glen Jarvis, Simeon Franklin; BayPiggies Python for scientific research, discussion with Guido van Rossum; UCBSciPy Netbooks - Michael Gorven, Dave Mackie, and Jonathan Carter; CLUG Japan Linux Symposium Keynote, Linus Torvalds & Jim Zemlin; Linux Foundation http://sites.google.com/site/berkeleytip/talk-videos Download & watch them before the meetings, discuss at the meetings. Thanks to all the Speakers, Videographers, & Groups! :) [Record your local meeting! Put the video online, & email me for inclusion for next month. :) ] ===== Hot topics: Ubuntu 9.10 - Problems? Fixes? Upgrade? Install? Freeswitch VOIP server - setup for BTIP Flyers & outreach to UCBerkeley. Outreach to other UC campuses next semester. ===== Opportunities - Learn new, or increase your job skills, &/or volunteer & help the community: Set up any of: a BTIP Mailing List, web server/site, Freeswitch VOIP server, or Virtual Private Network & SSL ===== Announcement Flyers: Print & Post them in your community. 4/5 available - Freedom, Karmic Koala, Free Culture, SciPy, OLPC. See bottom of page: http://groups.google.com/group/BerkTIPGlobal ===== New BTIP Webpages @ http://sites.google.com/site/berkeleytip/ UC Campus local groups; Free Hardware; System Administration; Announcement Flyers; Opportunities For Forwarding - You are invited to forward this announcement wherever it would be appreciated. From mike at jerris.com Sat Nov 7 09:47:22 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 7 Nov 2009 12:47:22 -0500 Subject: [Freeswitch-users] Events in mod_perl In-Reply-To: <7d79b3930911062138l262db9bema29c620a7b8cef94@mail.gmail.com> References: <7d79b3930911052229k2828ff7dic6d5b887a4897c8c@mail.gmail.com> <7d79b3930911062138l262db9bema29c620a7b8cef94@mail.gmail.com> Message-ID: You can use EventConsumer class for this, I am afraid its not very documented, but I do recall either a sample or discussion on the mailing list that you should be able to find. Mike On Nov 7, 2009, at 12:38 AM, lakshmanan ganapathy wrote: > Ya. I have done that event processing with ESL. But I wanted to > know, whether in mod_perl, we can get the events and process it or > not. I've seen function's like events_get etc.. But I don't know how > to use those things. > > In mod_perl if I'm able to get the events, then it will be easier > for me. > Is it possible!!! > > 2009/11/6 Jo?o Mesquita > I don't know what you are trying to do exactly but I think that you > might need to you ESL instead. > > Why don't you take a look at all the examples inside ${SVNROOT}/libs/ > esl and see if that fits you? I have a hunch that it would. > > JM > > On Fri, Nov 6, 2009 at 4:29 AM, lakshmanan ganapathy > wrote: > Hi all, > Is there any way to receive events while running a perl program > with the help of mod_perl?? > > I've seen some functions related to sending and receiving events in > the mod_perl wiki. But I don't know how to use that. > Any help!!! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/9fec2f91/attachment.html From mike at jerris.com Sat Nov 7 09:50:23 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 7 Nov 2009 12:50:23 -0500 Subject: [Freeswitch-users] leg_delay_start In-Reply-To: <3e6d7b0c0911070626g32af550fsd80e99d8266a7aa8@mail.gmail.com> References: <3e6d7b0c0911070626g32af550fsd80e99d8266a7aa8@mail.gmail.com> Message-ID: <057FECFA-306B-4710-93BA-6998AB15E5B8@jerris.com> Those vars were not even available in 1.0.3. I can't recall if they were in 1.0.4 or if you will need to use the latest 1.0.5 pre-release. Mike On Nov 7, 2009, at 9:26 AM, Steven Brown wrote: > Hi > > I've been trying to experiment with leg_delay_start when bridging to > two mobiles via a gateway, however regardless of settings both legs > are bridged immediately. I noticed a previous post on problems with > leg_delay_start which seemed to go unanswered, just wondered if > there is a known issue or if its something I'm doing wrong. > > Using FS 1.0.3 > > Dialplan extract as follows : > > > > Any pointers appreciated From mike at jerris.com Sat Nov 7 09:52:00 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 7 Nov 2009 12:52:00 -0500 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> Message-ID: <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> looks like ogg devel packages are installed but ogg lib is not? On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: > FreeSWITCH seems to be unable to read MP3 files, citing that it's an > unknown format. Looking through the log, I found this during startup: > > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error > Loading module /usr/local/freeswitch/mod/mod_shout.so > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: > ogg_sync_wrote** > > There don't seem to be any compile-time errors, yet I can't seem to > eliminate this issue. Any help would be appreciated. From msc at freeswitch.org Sat Nov 7 10:49:27 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 7 Nov 2009 10:49:27 -0800 Subject: [Freeswitch-users] Valid Dial Strings In-Reply-To: <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> Message-ID: <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" wrote: > Hi, > > can someone pointme to where the valid dialing strings are specified ? > For SIP dialstrings check here: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings Also, if you send us examples of what you've tried we can help you figure out what's wrong. > I'm assuming that something, somewhere, tells FS that numbers are > invalid > before they get dialed ? Pastebin some debug logs of what's happening. Check out this page which has lots of useful information on how to collect information: http://wiki.freeswitch.org/wiki/Reporting_Bugs It sounds like it's just a matter of figuring out how to configure your specific setup. Please report back with more information and we'll be happy to help. -MC From msc at freeswitch.org Sat Nov 7 11:02:45 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 7 Nov 2009 11:02:45 -0800 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: <4AF4AF73.8070804@tx.rr.com> References: <4AF4AF73.8070804@tx.rr.com> Message-ID: <21938E73-E566-431B-A0EA-7DE1731E1F8B@freeswitch.org> On Nov 6, 2009, at 3:21 PM, Orien Love wrote: > First of all, Thanks to the help I received on my pfSense > installation, > especially to Michael. I have a basic test system up and running. I > am > still waiting on some hardware but the base system is working!!!! > > I am looking on advice on how to set up a simple office PBX, 20 phones > and 4 outside lines.with 2 or 3 "operator" phones and the rest will be > extensions. > > Here is my plan, please let me know if it does not make sense, or if I > am going about it > > System Hardware > 4 spa3000's to handle the outside lines. > 2-3 polycom 601 phones with expansion modules (Operator phones) > 18 polycom 330 or other phones for desks. > 2-24 port cisco POE switches > 1 pfSense server. > > System Design. > > Extension Numbers 2xx > Outside line access 1xxxxxxxxxx > groups 3xx > auto-attendent ??? > > here are my questions > #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB > of > memory handle this proposed system? (Here is the MB I am thing of > using > MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) > #2 how do I pool my spa 3000 FXO lines so that the outgoing calls > use the first available line? also how do insure that metro (non long > distance) calls go to a specific line if available? > > I have learned a lot on how to set up Polycom 601 phones, I am > planning > on writing a how to document, is there any specific format? > > Thanks Orien > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From msc at freeswitch.org Sat Nov 7 11:07:55 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 7 Nov 2009 11:07:55 -0800 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: <4AF4AF73.8070804@tx.rr.com> References: <4AF4AF73.8070804@tx.rr.com> Message-ID: <92D8199B-5094-4170-9208-CD299BAF9D31@freeswitch.org> On Nov 6, 2009, at 3:21 PM, Orien Love wrote: > First of all, Thanks to the help I received on my pfSense > installation, > especially to Michael. I have a basic test system up and running. I > am > still waiting on some hardware but the base system is working!!!! > > I am looking on advice on how to set up a simple office PBX, 20 phones > and 4 outside lines.with 2 or 3 "operator" phones and the rest will be > extensions. > > Here is my plan, please let me know if it does not make sense, or if I > am going about it > > System Hardware > 4 spa3000's to handle the outside lines. > 2-3 polycom 601 phones with expansion modules (Operator phones) > 18 polycom 330 or other phones for desks. > 2-24 port cisco POE switches > 1 pfSense server. > > System Design. > > Extension Numbers 2xx > Outside line access 1xxxxxxxxxx > groups 3xx > auto-attendent ??? > > here are my questions > #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB > of > memory handle this proposed system? (Here is the MB I am thing of > using > MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) The FS devs don't endorse or recommend any specific hardware. However, many FreeSWITCH user are quite vocal about the hardware they prefer so we will let them speak. Just remember that YMMV depending on your specific setup. That being said, it does not sound like your HW reqs are very intense. Some of our members who've used the atom can probably give you specific feedback. > #2 how do I pool my spa 3000 FXO lines so that the outgoing calls > use the first available line? also how do insure that metro (non long > distance) calls go to a specific line if available? > > I have learned a lot on how to set up Polycom 601 phones, I am > planning > on writing a how to document, is there any specific format? > > Thanks Orien > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From msc at freeswitch.org Sat Nov 7 11:19:25 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 7 Nov 2009 11:19:25 -0800 Subject: [Freeswitch-users] suggestions for hardware. Message-ID: <194F9101-604D-46A9-9530-AD3C47CFC54C@freeswitch.org> On Nov 6, 2009, at 3:21 PM, Orien Love wrote: > First of all, Thanks to the help I received on my pfSense > installation, > especially to Michael. I have a basic test system up and running. I > am > still waiting on some hardware but the base system is working!!!! > > I am looking on advice on how to set up a simple office PBX, 20 phones > and 4 outside lines.with 2 or 3 "operator" phones and the rest will be > extensions. > > Here is my plan, please let me know if it does not make sense, or if I > am going about it > > System Hardware > 4 spa3000's to handle the outside lines. > 2-3 polycom 601 phones with expansion modules (Operator phones) > 18 polycom 330 or other phones for desks. > 2-24 port cisco POE switches > 1 pfSense server. > > System Design. > > Extension Numbers 2xx > Outside line access 1xxxxxxxxxx > groups 3xx > auto-attendent ??? > > here are my questions > #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of > memory handle this proposed system? (Here is the MB I am thing of > using > MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) > #2 how do I pool my spa 3000 FXO lines so that the outgoing calls > use the first available line? also how do insure that metro (non long > distance) calls go to a specific line if available? Sorry for the multiple posts. (Stinking iPhone without a slide-out keyboard.) This can be done with a little dialplan logic and using the pipe sep list of dialstrings. Check out the bridge examples here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall > > I have learned a lot on how to set up Polycom 601 phones, I am > planning > on writing a how to document, is there any specific format? > I looked around and I didn't see any other fully documented examples of hardware setup and config. My recommendation is to create a new wiki page and do your write-up. When you're done let me know and we will figure out how to index it. -MC > Thanks Orien > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From steve at justfone.com Sat Nov 7 11:59:02 2009 From: steve at justfone.com (Steven Brown) Date: Sat, 7 Nov 2009 19:59:02 +0000 Subject: [Freeswitch-users] leg_delay_start Message-ID: <3e6d7b0c0911071159q56faf627w710fcede6d6c031b@mail.gmail.com> Thanks Mike, I should have checked that, I've just done a make current on my other FS box and tested on it and can confirm that leg_delay_start works a treat, exactly as I need. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/9d896ea2/attachment.html From stevendt at primrosebank.net Sat Nov 7 13:34:26 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Sat, 7 Nov 2009 21:34:26 -0000 Subject: [Freeswitch-users] Valid Dial Strings References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> Message-ID: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> Hi Michael, thanks for the reply. I think that I have got to the bottom of how to allow numbers to get to the VOIP gateway - at the moment, my dialplan just allows any. The big problem is that the VOIP Gateway (Linksys 3102) rejects any calls to it from VOIP to the PSTN and I don't know why. I have posted a dump to the pastebin, hopefully, the messages in there will allow someone to see what the problem is and give me some pointers on how I might fix it regards Dave ----- Original Message ----- From: "Michael S Collins" To: Sent: Saturday, November 07, 2009 6:49 PM Subject: Re: [Freeswitch-users] Valid Dial Strings > > On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" > wrote: > >> Hi, >> >> can someone pointme to where the valid dialing strings are specified ? >> > For SIP dialstrings check here: > http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings > > Also, if you send us examples of what you've tried we can help you > figure out what's wrong. > >> I'm assuming that something, somewhere, tells FS that numbers are >> invalid >> before they get dialed ? > > Pastebin some debug logs of what's happening. Check out this page > which has lots of useful information on how to collect information: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It sounds like it's just a matter of figuring out how to configure > your specific setup. Please report back with more information and > we'll be happy to help. > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Sat Nov 7 18:20:42 2009 From: dujinfang at gmail.com (Seven Du) Date: Sun, 8 Nov 2009 10:20:42 +0800 Subject: [Freeswitch-users] Announce FreeSWITCH-CN - the Chinese community Message-ID: <23f91030911071820w43f72b68lb81e1785f3ac780b@mail.gmail.com> ALL, FreeSWITCH-CN is a non-official, non-profit Chinese community. There was some arguments of language specified sites vs. a central site, freeswitch.org, on this list. However, facts are that people would like to find information in their native language and here are already some language specified community exists - it, ru es etc.. And I'm sure that given time we will get more and more people involve in around the world. In my opinion, I still perfer to host documents on the office wiki as long as it support multi-language translations which I think would be quick as Janitors already starting to re-organizing the wiki. And make http://www.freeswitch.org.cn just a landing page - search engines hit the page and we will guide users to the proper information. Also there's a google group: http://groups.google.com/group/freeswitch-cn?hl=en available. Thanks Anthony and his team's permission of us using the FreeSWITCH logo and the sexy domain name in a non-profit condition. Thanks all involved in this project to make the community stronger and keep it rolling. Join us and leave your advices to make it better. -7- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091108/f90e08fb/attachment.html From lei.tlfly at gmail.com Sat Nov 7 19:36:09 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Sun, 8 Nov 2009 11:36:09 +0800 Subject: [Freeswitch-users] Announce FreeSWITCH-CN - the Chinese community In-Reply-To: <23f91030911071820w43f72b68lb81e1785f3ac780b@mail.gmail.com> References: <23f91030911071820w43f72b68lb81e1785f3ac780b@mail.gmail.com> Message-ID: <50c41b4e0911071936o3c963a2av9cdfb8fe66bfbd5@mail.gmail.com> Congratulations? 2009/11/8 Seven Du > ALL, > > FreeSWITCH-CN is a non-official, non-profit Chinese community. > > There was some arguments of language specified sites vs. a central site, > freeswitch.org, on this list. However, facts are that people would like to > find information in their native language and here are already some language > specified community exists - it, ru es etc.. And I'm sure that given time > we will get more and more people involve in around the world. > > In my opinion, I still perfer to host documents on the office wiki as long > as it support multi-language translations which I think would be quick as > Janitors already starting to re-organizing the wiki. And make > http://www.freeswitch.org.cn just a landing page - search engines hit the > page and we will guide users to the proper information. > > Also there's a google group: > http://groups.google.com/group/freeswitch-cn?hl=en available. > > Thanks Anthony and his team's permission of us using the FreeSWITCH logo > and the sexy domain name in a non-profit condition. Thanks all involved in > this project to make the community stronger and keep it rolling. > > Join us and leave your advices to make it better. > > -7- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091108/13078c6a/attachment.html From yehavi.bourvine at gmail.com Sun Nov 8 00:46:13 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 8 Nov 2009 10:46:13 +0200 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information Message-ID: Hello, While trying to display the *called party *name on SNOM phones I've found that the field sent to the phone needs to be changed slightly in order to make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and Cisco work ok. Just wanted to let the developers know... And now a question: We have SNOM phones monitoring other extensions (BLF feature). When a call comes in, the monitoring phones get notification, but the name field (identity display) contains the calling extension number and not its display name. Can this be fixed? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091108/85f87fea/attachment.html From list.subscription at alexrambau.com Sun Nov 8 02:42:38 2009 From: list.subscription at alexrambau.com (Alex Rambau) Date: Sun, 8 Nov 2009 03:42:38 -0700 Subject: [Freeswitch-users] Event Socket Timeout - Outbound Message-ID: Currently, is there any way to set the timeout on an outbound event socket? In case, for whatever reason, the socket application at 192.168.1.108:4444 is unresponsive or offline, I would like the call to not wait the extraordinary amount of time it takes to timeout so that I can handle it in other ways. My dial plan entry is as follows: Thanks in advance, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091108/e31a2912/attachment.html From mcampbellsmith at gmail.com Sun Nov 8 03:59:32 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 8 Nov 2009 22:59:32 +1100 Subject: [Freeswitch-users] Extension: No audio Message-ID: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> Hi! I have FS natted and am connecting with an 'external' extension that is registered to FS. ie the extension 2000 is registered on the internet with a public IP through my router to FS (192.168.1.120 IP address). uPnP works and I see that the extension is registered successfully. The problem is that I do not get any audio When looking at the SIP trace, I see the INVITE but do not see a TRYING or RINGING message. The extension is actually ringing. I modified the RTP port range on the remote end to match the RTP ports of freeswitch. I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 If anyone has an idea what needs to be set to get audio, help appreciated. Thanks! From god.nirvana at gmail.com Sun Nov 8 08:34:09 2009 From: god.nirvana at gmail.com (god.nirvana) Date: Mon, 9 Nov 2009 00:34:09 +0800 Subject: [Freeswitch-users] javascript parameter Message-ID: <200911090034059066890@gmail.com> hi all: how can i get the value of the myArg1 myArg2 in test.js. like this originate sofia/example/1000 at somewhere.com '&javascript(test.js myArg1 myArg2)' thanks! 2009-11-09 god.nirvana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/e372f11e/attachment-0001.html From brian at freeswitch.org Sun Nov 8 09:21:18 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 8 Nov 2009 11:21:18 -0600 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information In-Reply-To: References: Message-ID: Can you elaborate on this and provide a patch on jira? /b On Nov 8, 2009, at 2:46 AM, Yehavi Bourvine wrote: > Hello, > > While trying to display the called party name on SNOM phones I've > found that the field sent to the phone needs to be changed slightly > in order to make SNOM work: Insetad of sending P-Assterted-Identity > SNOM expects Remote-Party-ID. I changed it in mod_sofia and now > SNOM, Polycom and Cisco work ok. Just wanted to let the developers > know... > > And now a question: We have SNOM phones monitoring other > extensions (BLF feature). When a call comes in, the monitoring > phones get notification, but the name field (identity display) > contains the calling extension number and not its display name. Can > this be fixed? > > Thanks! __Yehavi: From bruce at nani.ca Sun Nov 8 01:12:26 2009 From: bruce at nani.ca (Bruce Fletcher) Date: Sun, 8 Nov 2009 01:12:26 -0800 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 Message-ID: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> I'm trying to get through the noobie tutorial that c888 recommends in IRC, but PortAudio doesn't seem to build properly on Mac OS X 10.6. It failed due to some code that wasn't 64bit ready, apparently. The error I got was exactly the same as this 4 month old error from MacPorts: http://trac.macports.org/ticket/20338#comment:4 Someone there produced the following patch, which almost got me going again: http://trac.macports.org/changeset/53938 This has to be manually applied because the line numbers don't match up between the MacPorts and FreeSWITCH versions of PortAudio. At least, not in the configure script. This still doesn't quite get portaudio built, but I found a suggestion to configure it with -- disable-shared here: http://music.columbia.edu/pipermail/portaudio/2008-February/008188.html This finally got me to the point of having a compiled version of portaudio, but of course with that much hacking around nothing is going to go smoothly, is it. I can boot freeswitch fine and 'load mod_portaudio' seems to work, but when I try to hear some MOH output, I get this: freeswitch at Media-Centre.local> pa call 9999 2009-11-08 00:50:51.894512 [NOTICE] switch_channel.c:613 New Channel portaudio/9999 [3079018b-0df8-4c61-bffa-2f2eb681d06d] Assertion failed: (sizeof( UInt32 ) == sizeof( long )), function ringBufferIOProc, file src/hostapi/coreaudio/pa_mac_core.c, line 1713. I'd be happy to poke around on the PortAudio site and see if there is any useful information or patches there, but it's really late now so I just wanted to document where I've gotten to in case someone else has the same problem or a better clue on how to fix things. This is all using version 15396 out of subversion, if it matters. Thanks, - Bruce From brian at freeswitch.org Sun Nov 8 09:26:09 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 8 Nov 2009 11:26:09 -0600 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> Message-ID: <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> The problem is the patch isn't backwards compatible and blows away any chance of being so. We have looked at this... and that patch IS NOT RIGHT. /b On Nov 8, 2009, at 3:12 AM, Bruce Fletcher wrote: > I'm trying to get through the noobie tutorial that c888 recommends in > IRC, but PortAudio doesn't seem to build properly on Mac OS X 10.6. > It failed due to some code that wasn't 64bit ready, apparently. The > error I got was exactly the same as this 4 month old error from > MacPorts: > > http://trac.macports.org/ticket/20338#comment:4 > > Someone there produced the following patch, which almost got me going > again: > > http://trac.macports.org/changeset/53938 > > This has to be manually applied because the line numbers don't match > up between the MacPorts and FreeSWITCH versions of PortAudio. At > least, not in the configure script. This still doesn't quite get > portaudio built, but I found a suggestion to configure it with -- > disable-shared here: > > http://music.columbia.edu/pipermail/portaudio/2008-February/008188.html > > This finally got me to the point of having a compiled version of > portaudio, but of course with that much hacking around nothing is > going to go smoothly, is it. I can boot freeswitch fine and 'load > mod_portaudio' seems to work, but when I try to hear some MOH output, > I get this: > > freeswitch at Media-Centre.local> pa call 9999 > 2009-11-08 00:50:51.894512 [NOTICE] switch_channel.c:613 New Channel > portaudio/9999 [3079018b-0df8-4c61-bffa-2f2eb681d06d] > Assertion failed: (sizeof( UInt32 ) == sizeof( long )), function > ringBufferIOProc, file src/hostapi/coreaudio/pa_mac_core.c, line 1713. > > I'd be happy to poke around on the PortAudio site and see if there is > any useful information or patches there, but it's really late now so I > just wanted to document where I've gotten to in case someone else has > the same problem or a better clue on how to fix things. > > This is all using version 15396 out of subversion, if it matters. > > Thanks, > - Bruce > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From bruce at nani.ca Sun Nov 8 10:03:01 2009 From: bruce at nani.ca (Bruce Fletcher) Date: Sun, 8 Nov 2009 10:03:01 -0800 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> Message-ID: <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> OK, I'll ignore that MacPorts patch for now and try to find a better approach. I'll look into this further tonight, but this morning I found a more recent promising patch on the PortAudio site: http://www.portaudio.com/trac/changeset/1418 It seems to push some data types to 32 bit regardless of platform, which might work better than the MacPorts approach of migrating some data structures to 64 bit. At any rate, this patch being on the PortAudio site suggests it might be a more approved fix. I'll keep plugging at this in my free time and report any significant progress back to the list. Thanks, - Bruce On 2009-11-08, at 9:26 AM, Brian West wrote: > The problem is the patch isn't backwards compatible and blows away any > chance of being so. We have looked at this... and that patch IS NOT > RIGHT. > > /b > > On Nov 8, 2009, at 3:12 AM, Bruce Fletcher wrote: > >> I'm trying to get through the noobie tutorial that c888 recommends in >> IRC, but PortAudio doesn't seem to build properly on Mac OS X 10.6. >> It failed due to some code that wasn't 64bit ready, apparently. The >> error I got was exactly the same as this 4 month old error from >> MacPorts: >> >> http://trac.macports.org/ticket/20338#comment:4 >> >> Someone there produced the following patch, which almost got me going >> again: >> >> http://trac.macports.org/changeset/53938 >> >> This has to be manually applied because the line numbers don't match >> up between the MacPorts and FreeSWITCH versions of PortAudio. At >> least, not in the configure script. This still doesn't quite get >> portaudio built, but I found a suggestion to configure it with -- >> disable-shared here: >> >> http://music.columbia.edu/pipermail/portaudio/2008-February/008188.html >> >> This finally got me to the point of having a compiled version of >> portaudio, but of course with that much hacking around nothing is >> going to go smoothly, is it. I can boot freeswitch fine and 'load >> mod_portaudio' seems to work, but when I try to hear some MOH output, >> I get this: >> >> freeswitch at Media-Centre.local> pa call 9999 >> 2009-11-08 00:50:51.894512 [NOTICE] switch_channel.c:613 New Channel >> portaudio/9999 [3079018b-0df8-4c61-bffa-2f2eb681d06d] >> Assertion failed: (sizeof( UInt32 ) == sizeof( long )), function >> ringBufferIOProc, file src/hostapi/coreaudio/pa_mac_core.c, line >> 1713. >> >> I'd be happy to poke around on the PortAudio site and see if there is >> any useful information or patches there, but it's really late now >> so I >> just wanted to document where I've gotten to in case someone else has >> the same problem or a better clue on how to fix things. >> >> This is all using version 15396 out of subversion, if it matters. >> >> Thanks, >> - Bruce >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From frank at carmickle.com Sun Nov 8 11:22:42 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sun, 8 Nov 2009 14:22:42 -0500 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> Message-ID: <20091108192242.GA10757@base.carmickle.com> Hello I am also having trouble with portaudio. I still haven't figured out what it is that is wrong. I have had it working in the past on this same machine same install of debian lenny. Now it just reports [ERR] mod_portaudio.c:964 Cannot find an input device Also seeing this on a fedor 12 64 bit machine but not a sid 64 bit machine or a fedora 11 32 bit machine. I'm not sure if it's the same problem but please do keep me informed about what you find. Thanks --Frank From mike at jerris.com Sun Nov 8 12:25:28 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 8 Nov 2009 15:25:28 -0500 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> Message-ID: <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> If you can figure out a clean way for us to do this with proper ifdefs in tree in a way that will not break others that would be the most preferred. Mike On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote: > OK, I'll ignore that MacPorts patch for now and try to find a better > approach. > > I'll look into this further tonight, but this morning I found a more > recent promising patch on the PortAudio site: > > http://www.portaudio.com/trac/changeset/1418 > > It seems to push some data types to 32 bit regardless of platform, > which might work better than the MacPorts approach of migrating some > data structures to 64 bit. At any rate, this patch being on the > PortAudio site suggests it might be a more approved fix. > > I'll keep plugging at this in my free time and report any significant > progress back to the list. > > Thanks, > - Bruce > From mike at jerris.com Sun Nov 8 12:28:21 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 8 Nov 2009 15:28:21 -0500 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> Message-ID: You don't have ext-rtp-ip set in your config. Mike On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: > Hi! > > I have FS natted and am connecting with an 'external' extension that > is registered to FS. ie the extension 2000 is registered on the > internet with a public IP through my router to FS (192.168.1.120 IP > address). uPnP works and I see that the extension is registered > successfully. > > The problem is that I do not get any audio > > When looking at the SIP trace, I see the INVITE but do not see a > TRYING or RINGING message. The extension is actually ringing. I > modified the RTP port range on the remote end to match the RTP ports > of freeswitch. > > I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 > > If anyone has an idea what needs to be set to get audio, help > appreciated. > > Thanks! From mcampbellsmith at gmail.com Sun Nov 8 13:59:44 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 9 Nov 2009 08:59:44 +1100 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> Message-ID: <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> Hi Mike, I should have put that in also. I do have external_rtp_ip set in my config. I have it set to my domain name: I should also mention that if I use flaphone.com (which registers with an external IP address), then I get audio. In sofia, I see my IP addresses: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.1.120 Ext-RTP-IP 124.xxx.xxx.xxx SIP-IP 192.168.1.120 Ext-SIP-IP 124.xxx.xxx.x URL sip:mod_sofia at 192.168.1.120:5060 BIND-URL sip:mod_sofia at 192.168.1.120:5060 HOLD-MUSIC silence OUTBOUND-PROXY N/A CODECS G726-32,G722,PCMU,PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris wrote: > You don't have ext-rtp-ip set in your config. > > Mike > > On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: > >> Hi! >> >> I have FS natted and am connecting with an 'external' extension that >> is registered to FS. ?ie the extension 2000 is registered on the >> internet with a public IP through my router to FS (192.168.1.120 IP >> address). ?uPnP works and I see that the extension is registered >> successfully. >> >> The problem is that I do not get any audio >> >> When looking at the SIP trace, I see the INVITE but do not see a >> TRYING or RINGING message. ?The extension is actually ringing. ?I >> modified the RTP port range on the remote end to match the RTP ports >> of freeswitch. >> >> I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 >> >> If anyone has an idea what needs to be set to get audio, help >> appreciated. >> >> Thanks! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Sun Nov 8 16:41:42 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 8 Nov 2009 19:41:42 -0500 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> Message-ID: <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> Your packet traces would disagree with the statements below. It is sending your internal address in rtp, so its not set correctly on whatever profile your using to call out, MIke On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: > Hi Mike, > > I should have put that in also. > > I do have external_rtp_ip set in my config. I have it set to my > domain name: > > > I should also mention that if I use flaphone.com (which registers with > an external IP address), then I get audio. In sofia, I see my IP > addresses: > > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > Name internal > Domain Name N/A > DBName sofia_reg_internal > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_from > RTP-IP 192.168.1.120 > Ext-RTP-IP 124.xxx.xxx.xxx > SIP-IP 192.168.1.120 > Ext-SIP-IP 124.xxx.xxx.x > URL sip:mod_sofia at 192.168.1.120:5060 > BIND-URL sip:mod_sofia at 192.168.1.120:5060 > HOLD-MUSIC silence > OUTBOUND-PROXY N/A > CODECS G726-32,G722,PCMU,PCMA > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT true > STUN-ENABLED true > STUN-AUTO-DISABLE false > > On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris > wrote: >> You don't have ext-rtp-ip set in your config. >> >> Mike >> >> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: >> >>> Hi! >>> >>> I have FS natted and am connecting with an 'external' extension that >>> is registered to FS. ie the extension 2000 is registered on the >>> internet with a public IP through my router to FS (192.168.1.120 IP >>> address). uPnP works and I see that the extension is registered >>> successfully. >>> >>> The problem is that I do not get any audio >>> >>> When looking at the SIP trace, I see the INVITE but do not see a >>> TRYING or RINGING message. The extension is actually ringing. I >>> modified the RTP port range on the remote end to match the RTP ports >>> of freeswitch. >>> >>> I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 >>> >>> If anyone has an idea what needs to be set to get audio, help >>> appreciated. >>> >>> Thanks! >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mcampbellsmith at gmail.com Sun Nov 8 18:14:48 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 9 Nov 2009 13:14:48 +1100 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> Message-ID: <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> OK.. thanks Mike. I assume I am using the Internal profile. I have defined user 2000 in the 'directory' using a context called family: switch_ivr.c:1367 Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family] This is an extract from sofia: sofia status profile internal ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.1.120 Ext-RTP-IP 124.xxx.xxx.xxx SIP-IP 192.168.1.120 Ext-SIP-IP 124.xxx.xxx.xxx URL sip:mod_sofia at 192.168.1.120:5060 BIND-URL sip:mod_sofia at 192.168.1.120:5060 HOLD-MUSIC silence OUTBOUND-PROXY N/A CODECS G726-32,G722,PCMU,PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 100 FAILED-CALLS-IN 25 CALLS-OUT 38 FAILED-CALLS-OUT 31 Registrations: ================================================================================================= Call-ID: 68534BBA9B461526 at 58.169.138.53 User: 2000 at 192.168.1.120 Contact: "user" Agent: dunno Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) Host: freeswitch IP: 58.xxx.xxx.xxx Port: 5060 Auth-User: 2000 Auth-Realm: markcs.dyndns.org MWI-Account: 2000 at 192.168.1.120 The internal.xml file has a lot in it, but I guess these are the important things for this profile: I will try to change auto-nat to be $${external_sip_ip} One question though: Any idea why I never see the TRYING or RINGING messages? Are tehse related to the RTP IP address or not? Without these I assume something is incorrect and I do not hear ringback.... Thanks! On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris wrote: > Your packet traces would disagree with the statements below. ?It is > sending your internal address in rtp, so its not set correctly on > whatever profile your using to call out, > > MIke > > On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: > >> Hi Mike, >> >> I should have put that in also. >> >> I do have external_rtp_ip set in my config. ?I have it set to my >> domain name: >> >> >> I should also mention that if I use flaphone.com (which registers with >> an external IP address), then I get audio. ?In sofia, I see my IP >> addresses: >> >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> ====================================================================== >> Name ? ? ? ? ? ? ? ? ? ?internal >> Domain Name ? ? ? ? ? ? N/A >> DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >> Pres Hosts >> Dialplan ? ? ? ? ? ? ? ?XML >> Context ? ? ? ? ? ? ? ? public >> Challenge Realm ? ? ? ? auto_from >> RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >> SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.x >> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >> HOLD-MUSIC ? ? ? ? ? ? ?silence >> OUTBOUND-PROXY ? ? ? ? ?N/A >> CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >> TEL-EVENT ? ? ? ? ? ? ? 101 >> DTMF-MODE ? ? ? ? ? ? ? rfc2833 >> CNG ? ? ? ? ? ? ? ? ? ? 13 >> SESSION-TO ? ? ? ? ? ? ?0 >> MAX-DIALOG ? ? ? ? ? ? ?0 >> NOMEDIA ? ? ? ? ? ? ? ? false >> LATE-NEG ? ? ? ? ? ? ? ?false >> PROXY-MEDIA ? ? ? ? ? ? false >> AGGRESSIVENAT ? ? ? ? ? true >> STUN-ENABLED ? ? ? ? ? ?true >> STUN-AUTO-DISABLE ? ? ? false >> >> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris >> wrote: >>> You don't have ext-rtp-ip set in your config. >>> >>> Mike >>> >>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: >>> >>>> Hi! >>>> >>>> I have FS natted and am connecting with an 'external' extension that >>>> is registered to FS. ?ie the extension 2000 is registered on the >>>> internet with a public IP through my router to FS (192.168.1.120 IP >>>> address). ?uPnP works and I see that the extension is registered >>>> successfully. >>>> >>>> The problem is that I do not get any audio >>>> >>>> When looking at the SIP trace, I see the INVITE but do not see a >>>> TRYING or RINGING message. ?The extension is actually ringing. ?I >>>> modified the RTP port range on the remote end to match the RTP ports >>>> of freeswitch. >>>> >>>> I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 >>>> >>>> If anyone has an idea what needs to be set to get audio, help >>>> appreciated. >>>> >>>> Thanks! >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sprice at gmail.com Sun Nov 8 18:26:20 2009 From: sprice at gmail.com (SP) Date: Sun, 8 Nov 2009 20:26:20 -0600 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information In-Reply-To: References: Message-ID: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> before playing with mod_sofia, did you try the sip_cid_type variable? http://wiki.freeswitch.org/wiki/Variable_sip_cid_type On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine wrote: > Hello, > > ??While?trying to display the called party name ?on SNOM phones I've found > that the field sent to the phone needs to be changed slightly in order to > make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects > Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and Cisco > work ok. Just wanted to let the developers know... > > ? And now a question: We have SNOM phones monitoring other extensions (BLF > feature). When a call comes in, the monitoring phones get notification, but > the name field (identity display) contains the calling extension number and > not its display name. Can this be fixed? > > ??????????????????????????????? Thanks! __Yehavi: > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Shannon From mcampbellsmith at gmail.com Sun Nov 8 18:32:04 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 9 Nov 2009 13:32:04 +1100 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> Message-ID: <33c87fa30911081832qdd7357ncfe016f10ba87160@mail.gmail.com> Hi again, Actually, changing the to means that I now see the IP address in the INVITE message: v=0 o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx s=FreeSWITCH c=IN IP4 124.xxx.xxx.xxx t=0 0 m=audio 21234 RTP/AVP 0 2 9 8 101 13 Why would this be? I thought auto-nat was meant to solve these issues? However, I still do not see the TRYING or RINGING messages.... ideas appreciated. Thanks! On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith wrote: > OK.. thanks Mike. > > I assume I am using the Internal profile. ? I have defined user 2000 > in the 'directory' using a context called family: ? switch_ivr.c:1367 > Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family] > > This is an extract from sofia: > > sofia status profile internal > ================================================================================================= > Name ? ? ? ? ? ? ? ? ? ?internal > Domain Name ? ? ? ? ? ? N/A > DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal > Pres Hosts > Dialplan ? ? ? ? ? ? ? ?XML > Context ? ? ? ? ? ? ? ? public > Challenge Realm ? ? ? ? auto_from > RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 > Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx > SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 > Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx > URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 > BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 > HOLD-MUSIC ? ? ? ? ? ? ?silence > OUTBOUND-PROXY ? ? ? ? ?N/A > CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA > TEL-EVENT ? ? ? ? ? ? ? 101 > DTMF-MODE ? ? ? ? ? ? ? rfc2833 > CNG ? ? ? ? ? ? ? ? ? ? 13 > SESSION-TO ? ? ? ? ? ? ?0 > MAX-DIALOG ? ? ? ? ? ? ?0 > NOMEDIA ? ? ? ? ? ? ? ? false > LATE-NEG ? ? ? ? ? ? ? ?false > PROXY-MEDIA ? ? ? ? ? ? false > AGGRESSIVENAT ? ? ? ? ? true > STUN-ENABLED ? ? ? ? ? ?true > STUN-AUTO-DISABLE ? ? ? false > CALLS-IN ? ? ? ? ? ? ? ?100 > FAILED-CALLS-IN ? ? ? ? 25 > CALLS-OUT ? ? ? ? ? ? ? 38 > FAILED-CALLS-OUT ? ? ? ?31 > > Registrations: > ================================================================================================= > Call-ID: ? ? ? ?68534BBA9B461526 at 58.169.138.53 > User: ? ? ? ? ? 2000 at 192.168.1.120 > Contact: ? ? ? ?"user" > Agent: ? ? ? ? ?dunno > Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) > Host: ? ? ? ? ? freeswitch > IP: ? ? ? ? ? ? 58.xxx.xxx.xxx > Port: ? ? ? ? ? 5060 > Auth-User: ? ? ?2000 > Auth-Realm: ? ? markcs.dyndns.org > MWI-Account: ? ?2000 at 192.168.1.120 > > The internal.xml file has a lot in it, but I guess these are the > important things for this profile: > > ? ? > ? ? > > ? ? > ? ? > > I will try to change auto-nat to be $${external_sip_ip} > > One question though: ?Any idea why I never see the TRYING or RINGING > messages? ? Are tehse related to the RTP IP address or not? ?Without > these I assume something is incorrect and I do not hear ringback.... > > Thanks! > > On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris wrote: >> Your packet traces would disagree with the statements below. ?It is >> sending your internal address in rtp, so its not set correctly on >> whatever profile your using to call out, >> >> MIke >> >> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: >> >>> Hi Mike, >>> >>> I should have put that in also. >>> >>> I do have external_rtp_ip set in my config. ?I have it set to my >>> domain name: >>> >>> >>> I should also mention that if I use flaphone.com (which registers with >>> an external IP address), then I get audio. ?In sofia, I see my IP >>> addresses: >>> >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> ====================================================================== >>> Name ? ? ? ? ? ? ? ? ? ?internal >>> Domain Name ? ? ? ? ? ? N/A >>> DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >>> Pres Hosts >>> Dialplan ? ? ? ? ? ? ? ?XML >>> Context ? ? ? ? ? ? ? ? public >>> Challenge Realm ? ? ? ? auto_from >>> RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >>> SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.x >>> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >>> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >>> HOLD-MUSIC ? ? ? ? ? ? ?silence >>> OUTBOUND-PROXY ? ? ? ? ?N/A >>> CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >>> TEL-EVENT ? ? ? ? ? ? ? 101 >>> DTMF-MODE ? ? ? ? ? ? ? rfc2833 >>> CNG ? ? ? ? ? ? ? ? ? ? 13 >>> SESSION-TO ? ? ? ? ? ? ?0 >>> MAX-DIALOG ? ? ? ? ? ? ?0 >>> NOMEDIA ? ? ? ? ? ? ? ? false >>> LATE-NEG ? ? ? ? ? ? ? ?false >>> PROXY-MEDIA ? ? ? ? ? ? false >>> AGGRESSIVENAT ? ? ? ? ? true >>> STUN-ENABLED ? ? ? ? ? ?true >>> STUN-AUTO-DISABLE ? ? ? false >>> >>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris >>> wrote: >>>> You don't have ext-rtp-ip set in your config. >>>> >>>> Mike >>>> >>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: >>>> >>>>> Hi! >>>>> >>>>> I have FS natted and am connecting with an 'external' extension that >>>>> is registered to FS. ?ie the extension 2000 is registered on the >>>>> internet with a public IP through my router to FS (192.168.1.120 IP >>>>> address). ?uPnP works and I see that the extension is registered >>>>> successfully. >>>>> >>>>> The problem is that I do not get any audio >>>>> >>>>> When looking at the SIP trace, I see the INVITE but do not see a >>>>> TRYING or RINGING message. ?The extension is actually ringing. ?I >>>>> modified the RTP port range on the remote end to match the RTP ports >>>>> of freeswitch. >>>>> >>>>> I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 >>>>> >>>>> If anyone has an idea what needs to be set to get audio, help >>>>> appreciated. >>>>> >>>>> Thanks! >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>>> users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From jmesquita at freeswitch.org Sun Nov 8 18:39:21 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 9 Nov 2009 00:39:21 -0200 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <33c87fa30911081832qdd7357ncfe016f10ba87160@mail.gmail.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> <33c87fa30911081832qdd7357ncfe016f10ba87160@mail.gmail.com> Message-ID: It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise, no go.... Have you changed the ext-sip-ip too? Regards, JM On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi again, > > Actually, changing the to > means that I > now see the IP address in the INVITE message: > > v=0 > o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx > s=FreeSWITCH > c=IN IP4 124.xxx.xxx.xxx > t=0 0 > m=audio 21234 RTP/AVP 0 2 9 8 101 13 > > Why would this be? I thought auto-nat was meant to solve these issues? > > However, I still do not see the TRYING or RINGING messages.... ideas > appreciated. > > Thanks! > > On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith > wrote: > > OK.. thanks Mike. > > > > I assume I am using the Internal profile. I have defined user 2000 > > in the 'directory' using a context called family: switch_ivr.c:1367 > > Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family] > > > > This is an extract from sofia: > > > > sofia status profile internal > > > ================================================================================================= > > Name internal > > Domain Name N/A > > DBName sofia_reg_internal > > Pres Hosts > > Dialplan XML > > Context public > > Challenge Realm auto_from > > RTP-IP 192.168.1.120 > > Ext-RTP-IP 124.xxx.xxx.xxx > > SIP-IP 192.168.1.120 > > Ext-SIP-IP 124.xxx.xxx.xxx > > URL sip:mod_sofia at 192.168.1.120:5060 > > BIND-URL sip:mod_sofia at 192.168.1.120:5060 > > HOLD-MUSIC silence > > OUTBOUND-PROXY N/A > > CODECS G726-32,G722,PCMU,PCMA > > TEL-EVENT 101 > > DTMF-MODE rfc2833 > > CNG 13 > > SESSION-TO 0 > > MAX-DIALOG 0 > > NOMEDIA false > > LATE-NEG false > > PROXY-MEDIA false > > AGGRESSIVENAT true > > STUN-ENABLED true > > STUN-AUTO-DISABLE false > > CALLS-IN 100 > > FAILED-CALLS-IN 25 > > CALLS-OUT 38 > > FAILED-CALLS-OUT 31 > > > > Registrations: > > > ================================================================================================= > > Call-ID: 68534BBA9B461526 at 58.169.138.53 > > User: 2000 at 192.168.1.120 > > Contact: "user" > > Agent: dunno > > Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) > > Host: freeswitch > > IP: 58.xxx.xxx.xxx > > Port: 5060 > > Auth-User: 2000 > > Auth-Realm: markcs.dyndns.org > > MWI-Account: 2000 at 192.168.1.120 > > > > The internal.xml file has a lot in it, but I guess these are the > > important things for this profile: > > > > > > > > > > > > > > > > I will try to change auto-nat to be $${external_sip_ip} > > > > One question though: Any idea why I never see the TRYING or RINGING > > messages? Are tehse related to the RTP IP address or not? Without > > these I assume something is incorrect and I do not hear ringback.... > > > > Thanks! > > > > On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris wrote: > >> Your packet traces would disagree with the statements below. It is > >> sending your internal address in rtp, so its not set correctly on > >> whatever profile your using to call out, > >> > >> MIke > >> > >> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: > >> > >>> Hi Mike, > >>> > >>> I should have put that in also. > >>> > >>> I do have external_rtp_ip set in my config. I have it set to my > >>> domain name: > >>> > >>> > >>> I should also mention that if I use flaphone.com (which registers with > >>> an external IP address), then I get audio. In sofia, I see my IP > >>> addresses: > >>> > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> ====================================================================== > >>> Name internal > >>> Domain Name N/A > >>> DBName sofia_reg_internal > >>> Pres Hosts > >>> Dialplan XML > >>> Context public > >>> Challenge Realm auto_from > >>> RTP-IP 192.168.1.120 > >>> Ext-RTP-IP 124.xxx.xxx.xxx > >>> SIP-IP 192.168.1.120 > >>> Ext-SIP-IP 124.xxx.xxx.x > >>> URL sip:mod_sofia at 192.168.1.120:5060 > >>> BIND-URL sip:mod_sofia at 192.168.1.120:5060 > >>> HOLD-MUSIC silence > >>> OUTBOUND-PROXY N/A > >>> CODECS G726-32,G722,PCMU,PCMA > >>> TEL-EVENT 101 > >>> DTMF-MODE rfc2833 > >>> CNG 13 > >>> SESSION-TO 0 > >>> MAX-DIALOG 0 > >>> NOMEDIA false > >>> LATE-NEG false > >>> PROXY-MEDIA false > >>> AGGRESSIVENAT true > >>> STUN-ENABLED true > >>> STUN-AUTO-DISABLE false > >>> > >>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris > >>> wrote: > >>>> You don't have ext-rtp-ip set in your config. > >>>> > >>>> Mike > >>>> > >>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: > >>>> > >>>>> Hi! > >>>>> > >>>>> I have FS natted and am connecting with an 'external' extension that > >>>>> is registered to FS. ie the extension 2000 is registered on the > >>>>> internet with a public IP through my router to FS (192.168.1.120 IP > >>>>> address). uPnP works and I see that the extension is registered > >>>>> successfully. > >>>>> > >>>>> The problem is that I do not get any audio > >>>>> > >>>>> When looking at the SIP trace, I see the INVITE but do not see a > >>>>> TRYING or RINGING message. The extension is actually ringing. I > >>>>> modified the RTP port range on the remote end to match the RTP ports > >>>>> of freeswitch. > >>>>> > >>>>> I have put a sip trace in the pastebin at > http://pastebin.freeswitch.org/11035 > >>>>> > >>>>> If anyone has an idea what needs to be set to get audio, help > >>>>> appreciated. > >>>>> > >>>>> Thanks! > >>>> > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>>> users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/fe572a26/attachment-0001.html From yehavi.bourvine at gmail.com Sun Nov 8 18:57:02 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 9 Nov 2009 04:57:02 +0200 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information In-Reply-To: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> References: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> Message-ID: I was not aware of this variable; I will take a look on it tomorrow. However, when looking in the code I did not find something which looks like "Remote-Party-ID'". Thanks! __Yehavi: 2009/11/9 SP > before playing with mod_sofia, did you try the sip_cid_type variable? > > http://wiki.freeswitch.org/wiki/Variable_sip_cid_type > > On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine > wrote: > > Hello, > > > > While trying to display the called party name on SNOM phones I've > found > > that the field sent to the phone needs to be changed slightly in order to > > make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects > > Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and > Cisco > > work ok. Just wanted to let the developers know... > > > > And now a question: We have SNOM phones monitoring other extensions > (BLF > > feature). When a call comes in, the monitoring phones get notification, > but > > the name field (identity display) contains the calling extension number > and > > not its display name. Can this be fixed? > > > > Thanks! __Yehavi: > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Shannon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/530e1fab/attachment.html From mcampbellsmith at gmail.com Sun Nov 8 19:40:17 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 9 Nov 2009 14:40:17 +1100 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> <33c87fa30911081832qdd7357ncfe016f10ba87160@mail.gmail.com> Message-ID: <33c87fa30911081940o4381f35dy6369b66384dbf90f@mail.gmail.com> Is there a way to determine if FS has detected nat? I am behind UPnP and I can see on the router the mappings for Freeswitch. 2009/11/9 Jo?o Mesquita : > It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise, > no go.... > > Have you changed the ext-sip-ip too? > > Regards, > > JM > > > On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith > wrote: >> >> Hi again, >> >> Actually, changing the to >> means that I >> now see the IP address in the INVITE message: >> >> ? v=0 >> ? o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx >> ? s=FreeSWITCH >> ? c=IN IP4 124.xxx.xxx.xxx >> ? t=0 0 >> ? m=audio 21234 RTP/AVP 0 2 9 8 101 13 >> >> Why would this be? ?I thought auto-nat was meant to solve these issues? >> >> However, I still do not see the TRYING or RINGING messages.... ?ideas >> appreciated. >> >> Thanks! >> >> On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith >> wrote: >> > OK.. thanks Mike. >> > >> > I assume I am using the Internal profile. ? I have defined user 2000 >> > in the 'directory' using a context called family: ? switch_ivr.c:1367 >> > Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family] >> > >> > This is an extract from sofia: >> > >> > sofia status profile internal >> > >> > ================================================================================================= >> > Name ? ? ? ? ? ? ? ? ? ?internal >> > Domain Name ? ? ? ? ? ? N/A >> > DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >> > Pres Hosts >> > Dialplan ? ? ? ? ? ? ? ?XML >> > Context ? ? ? ? ? ? ? ? public >> > Challenge Realm ? ? ? ? auto_from >> > RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> > Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >> > SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> > Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >> > URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >> > BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >> > HOLD-MUSIC ? ? ? ? ? ? ?silence >> > OUTBOUND-PROXY ? ? ? ? ?N/A >> > CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >> > TEL-EVENT ? ? ? ? ? ? ? 101 >> > DTMF-MODE ? ? ? ? ? ? ? rfc2833 >> > CNG ? ? ? ? ? ? ? ? ? ? 13 >> > SESSION-TO ? ? ? ? ? ? ?0 >> > MAX-DIALOG ? ? ? ? ? ? ?0 >> > NOMEDIA ? ? ? ? ? ? ? ? false >> > LATE-NEG ? ? ? ? ? ? ? ?false >> > PROXY-MEDIA ? ? ? ? ? ? false >> > AGGRESSIVENAT ? ? ? ? ? true >> > STUN-ENABLED ? ? ? ? ? ?true >> > STUN-AUTO-DISABLE ? ? ? false >> > CALLS-IN ? ? ? ? ? ? ? ?100 >> > FAILED-CALLS-IN ? ? ? ? 25 >> > CALLS-OUT ? ? ? ? ? ? ? 38 >> > FAILED-CALLS-OUT ? ? ? ?31 >> > >> > Registrations: >> > >> > ================================================================================================= >> > Call-ID: ? ? ? ?68534BBA9B461526 at 58.169.138.53 >> > User: ? ? ? ? ? 2000 at 192.168.1.120 >> > Contact: ? ? ? ?"user" >> > Agent: ? ? ? ? ?dunno >> > Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) >> > Host: ? ? ? ? ? freeswitch >> > IP: ? ? ? ? ? ? 58.xxx.xxx.xxx >> > Port: ? ? ? ? ? 5060 >> > Auth-User: ? ? ?2000 >> > Auth-Realm: ? ? markcs.dyndns.org >> > MWI-Account: ? ?2000 at 192.168.1.120 >> > >> > The internal.xml file has a lot in it, but I guess these are the >> > important things for this profile: >> > >> > ? ? >> > ? ? >> > >> > ? ? >> > ? ? >> > >> > I will try to change auto-nat to be $${external_sip_ip} >> > >> > One question though: ?Any idea why I never see the TRYING or RINGING >> > messages? ? Are tehse related to the RTP IP address or not? ?Without >> > these I assume something is incorrect and I do not hear ringback.... >> > >> > Thanks! >> > >> > On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris wrote: >> >> Your packet traces would disagree with the statements below. ?It is >> >> sending your internal address in rtp, so its not set correctly on >> >> whatever profile your using to call out, >> >> >> >> MIke >> >> >> >> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: >> >> >> >>> Hi Mike, >> >>> >> >>> I should have put that in also. >> >>> >> >>> I do have external_rtp_ip set in my config. ?I have it set to my >> >>> domain name: >> >>> >> >>> >> >>> I should also mention that if I use flaphone.com (which registers with >> >>> an external IP address), then I get audio. ?In sofia, I see my IP >> >>> addresses: >> >>> >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> ====================================================================== >> >>> Name ? ? ? ? ? ? ? ? ? ?internal >> >>> Domain Name ? ? ? ? ? ? N/A >> >>> DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >> >>> Pres Hosts >> >>> Dialplan ? ? ? ? ? ? ? ?XML >> >>> Context ? ? ? ? ? ? ? ? public >> >>> Challenge Realm ? ? ? ? auto_from >> >>> RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> >>> Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >> >>> SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> >>> Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.x >> >>> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >> >>> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >> >>> HOLD-MUSIC ? ? ? ? ? ? ?silence >> >>> OUTBOUND-PROXY ? ? ? ? ?N/A >> >>> CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >> >>> TEL-EVENT ? ? ? ? ? ? ? 101 >> >>> DTMF-MODE ? ? ? ? ? ? ? rfc2833 >> >>> CNG ? ? ? ? ? ? ? ? ? ? 13 >> >>> SESSION-TO ? ? ? ? ? ? ?0 >> >>> MAX-DIALOG ? ? ? ? ? ? ?0 >> >>> NOMEDIA ? ? ? ? ? ? ? ? false >> >>> LATE-NEG ? ? ? ? ? ? ? ?false >> >>> PROXY-MEDIA ? ? ? ? ? ? false >> >>> AGGRESSIVENAT ? ? ? ? ? true >> >>> STUN-ENABLED ? ? ? ? ? ?true >> >>> STUN-AUTO-DISABLE ? ? ? false >> >>> >> >>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris >> >>> wrote: >> >>>> You don't have ext-rtp-ip set in your config. >> >>>> >> >>>> Mike >> >>>> >> >>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: >> >>>> >> >>>>> Hi! >> >>>>> >> >>>>> I have FS natted and am connecting with an 'external' extension that >> >>>>> is registered to FS. ?ie the extension 2000 is registered on the >> >>>>> internet with a public IP through my router to FS (192.168.1.120 IP >> >>>>> address). ?uPnP works and I see that the extension is registered >> >>>>> successfully. >> >>>>> >> >>>>> The problem is that I do not get any audio >> >>>>> >> >>>>> When looking at the SIP trace, I see the INVITE but do not see a >> >>>>> TRYING or RINGING message. ?The extension is actually ringing. ?I >> >>>>> modified the RTP port range on the remote end to match the RTP ports >> >>>>> of freeswitch. >> >>>>> >> >>>>> I have put a sip trace in the pastebin at >> >>>>> http://pastebin.freeswitch.org/11035 >> >>>>> >> >>>>> If anyone has an idea what needs to be set to get audio, help >> >>>>> appreciated. >> >>>>> >> >>>>> Thanks! >> >>>> >> >>>> >> >>>> _______________________________________________ >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>>> users >> >>>> http://www.freeswitch.org >> >>>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>> users >> >>> http://www.freeswitch.org >> >> >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mcampbellsmith at gmail.com Sun Nov 8 20:18:14 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 9 Nov 2009 15:18:14 +1100 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <33c87fa30911081940o4381f35dy6369b66384dbf90f@mail.gmail.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> <33c87fa30911081832qdd7357ncfe016f10ba87160@mail.gmail.com> <33c87fa30911081940o4381f35dy6369b66384dbf90f@mail.gmail.com> Message-ID: <33c87fa30911082018r75bb2d9cvf9359846c4fb281f@mail.gmail.com> I think I've fixed it, but I had to change a few things... I had a host name set in vars.xml for external_rtp_ip and for external_sip_ip. Having the external_rtp_ip set to a hostname, sofia showed the RTP-IP 192.168.1.120 Ext-RTP-IP host:myhostname SIP-IP 192.168.1.120 Ext-SIP-IP 124.190.249.9 I think this caused some problems. Once this was changed back to stun, I now get RINGING messages and I get audio. I still have ext-rtp-ip and ext-sip-ip set to auto-nat in internal.xml. Could this be the cause or is there something else that caused this issue? I am using FreeSWITCH Version 1.0.trunk (15126) On Mon, Nov 9, 2009 at 2:40 PM, Mark Campbell-Smith wrote: > Is there a way to determine if FS has detected nat? ?I am behind UPnP > and I can see on the router the mappings for Freeswitch. > > 2009/11/9 Jo?o Mesquita : >> It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise, >> no go.... >> >> Have you changed the ext-sip-ip too? >> >> Regards, >> >> JM >> >> >> On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith >> wrote: >>> >>> Hi again, >>> >>> Actually, changing the to >>> means that I >>> now see the IP address in the INVITE message: >>> >>> ? v=0 >>> ? o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx >>> ? s=FreeSWITCH >>> ? c=IN IP4 124.xxx.xxx.xxx >>> ? t=0 0 >>> ? m=audio 21234 RTP/AVP 0 2 9 8 101 13 >>> >>> Why would this be? ?I thought auto-nat was meant to solve these issues? >>> >>> However, I still do not see the TRYING or RINGING messages.... ?ideas >>> appreciated. >>> >>> Thanks! >>> >>> On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith >>> wrote: >>> > OK.. thanks Mike. >>> > >>> > I assume I am using the Internal profile. ? I have defined user 2000 >>> > in the 'directory' using a context called family: ? switch_ivr.c:1367 >>> > Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family] >>> > >>> > This is an extract from sofia: >>> > >>> > sofia status profile internal >>> > >>> > ================================================================================================= >>> > Name ? ? ? ? ? ? ? ? ? ?internal >>> > Domain Name ? ? ? ? ? ? N/A >>> > DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >>> > Pres Hosts >>> > Dialplan ? ? ? ? ? ? ? ?XML >>> > Context ? ? ? ? ? ? ? ? public >>> > Challenge Realm ? ? ? ? auto_from >>> > RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> > Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >>> > SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> > Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >>> > URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >>> > BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >>> > HOLD-MUSIC ? ? ? ? ? ? ?silence >>> > OUTBOUND-PROXY ? ? ? ? ?N/A >>> > CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >>> > TEL-EVENT ? ? ? ? ? ? ? 101 >>> > DTMF-MODE ? ? ? ? ? ? ? rfc2833 >>> > CNG ? ? ? ? ? ? ? ? ? ? 13 >>> > SESSION-TO ? ? ? ? ? ? ?0 >>> > MAX-DIALOG ? ? ? ? ? ? ?0 >>> > NOMEDIA ? ? ? ? ? ? ? ? false >>> > LATE-NEG ? ? ? ? ? ? ? ?false >>> > PROXY-MEDIA ? ? ? ? ? ? false >>> > AGGRESSIVENAT ? ? ? ? ? true >>> > STUN-ENABLED ? ? ? ? ? ?true >>> > STUN-AUTO-DISABLE ? ? ? false >>> > CALLS-IN ? ? ? ? ? ? ? ?100 >>> > FAILED-CALLS-IN ? ? ? ? 25 >>> > CALLS-OUT ? ? ? ? ? ? ? 38 >>> > FAILED-CALLS-OUT ? ? ? ?31 >>> > >>> > Registrations: >>> > >>> > ================================================================================================= >>> > Call-ID: ? ? ? ?68534BBA9B461526 at 58.169.138.53 >>> > User: ? ? ? ? ? 2000 at 192.168.1.120 >>> > Contact: ? ? ? ?"user" >>> > Agent: ? ? ? ? ?dunno >>> > Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) >>> > Host: ? ? ? ? ? freeswitch >>> > IP: ? ? ? ? ? ? 58.xxx.xxx.xxx >>> > Port: ? ? ? ? ? 5060 >>> > Auth-User: ? ? ?2000 >>> > Auth-Realm: ? ? markcs.dyndns.org >>> > MWI-Account: ? ?2000 at 192.168.1.120 >>> > >>> > The internal.xml file has a lot in it, but I guess these are the >>> > important things for this profile: >>> > >>> > ? ? >>> > ? ? >>> > >>> > ? ? >>> > ? ? >>> > >>> > I will try to change auto-nat to be $${external_sip_ip} >>> > >>> > One question though: ?Any idea why I never see the TRYING or RINGING >>> > messages? ? Are tehse related to the RTP IP address or not? ?Without >>> > these I assume something is incorrect and I do not hear ringback.... >>> > >>> > Thanks! >>> > >>> > On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris wrote: >>> >> Your packet traces would disagree with the statements below. ?It is >>> >> sending your internal address in rtp, so its not set correctly on >>> >> whatever profile your using to call out, >>> >> >>> >> MIke >>> >> >>> >> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: >>> >> >>> >>> Hi Mike, >>> >>> >>> >>> I should have put that in also. >>> >>> >>> >>> I do have external_rtp_ip set in my config. ?I have it set to my >>> >>> domain name: >>> >>> >>> >>> >>> >>> I should also mention that if I use flaphone.com (which registers with >>> >>> an external IP address), then I get audio. ?In sofia, I see my IP >>> >>> addresses: >>> >>> >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> ====================================================================== >>> >>> Name ? ? ? ? ? ? ? ? ? ?internal >>> >>> Domain Name ? ? ? ? ? ? N/A >>> >>> DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >>> >>> Pres Hosts >>> >>> Dialplan ? ? ? ? ? ? ? ?XML >>> >>> Context ? ? ? ? ? ? ? ? public >>> >>> Challenge Realm ? ? ? ? auto_from >>> >>> RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> >>> Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >>> >>> SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> >>> Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.x >>> >>> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >>> >>> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >>> >>> HOLD-MUSIC ? ? ? ? ? ? ?silence >>> >>> OUTBOUND-PROXY ? ? ? ? ?N/A >>> >>> CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >>> >>> TEL-EVENT ? ? ? ? ? ? ? 101 >>> >>> DTMF-MODE ? ? ? ? ? ? ? rfc2833 >>> >>> CNG ? ? ? ? ? ? ? ? ? ? 13 >>> >>> SESSION-TO ? ? ? ? ? ? ?0 >>> >>> MAX-DIALOG ? ? ? ? ? ? ?0 >>> >>> NOMEDIA ? ? ? ? ? ? ? ? false >>> >>> LATE-NEG ? ? ? ? ? ? ? ?false >>> >>> PROXY-MEDIA ? ? ? ? ? ? false >>> >>> AGGRESSIVENAT ? ? ? ? ? true >>> >>> STUN-ENABLED ? ? ? ? ? ?true >>> >>> STUN-AUTO-DISABLE ? ? ? false >>> >>> >>> >>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris >>> >>> wrote: >>> >>>> You don't have ext-rtp-ip set in your config. >>> >>>> >>> >>>> Mike >>> >>>> >>> >>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: >>> >>>> >>> >>>>> Hi! >>> >>>>> >>> >>>>> I have FS natted and am connecting with an 'external' extension that >>> >>>>> is registered to FS. ?ie the extension 2000 is registered on the >>> >>>>> internet with a public IP through my router to FS (192.168.1.120 IP >>> >>>>> address). ?uPnP works and I see that the extension is registered >>> >>>>> successfully. >>> >>>>> >>> >>>>> The problem is that I do not get any audio >>> >>>>> >>> >>>>> When looking at the SIP trace, I see the INVITE but do not see a >>> >>>>> TRYING or RINGING message. ?The extension is actually ringing. ?I >>> >>>>> modified the RTP port range on the remote end to match the RTP ports >>> >>>>> of freeswitch. >>> >>>>> >>> >>>>> I have put a sip trace in the pastebin at >>> >>>>> http://pastebin.freeswitch.org/11035 >>> >>>>> >>> >>>>> If anyone has an idea what needs to be set to get audio, help >>> >>>>> appreciated. >>> >>>>> >>> >>>>> Thanks! >>> >>>> >>> >>>> >>> >>>> _______________________________________________ >>> >>>> FreeSWITCH-users mailing list >>> >>>> FreeSWITCH-users at lists.freeswitch.org >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> >>>> users >>> >>>> http://www.freeswitch.org >>> >>>> >>> >>> >>> >>> _______________________________________________ >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> >>> users >>> >>> http://www.freeswitch.org >>> >> >>> >> >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From shiyanov at gmail.com Sun Nov 8 23:04:10 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Mon, 9 Nov 2009 10:04:10 +0300 Subject: [Freeswitch-users] Dialpan: try.. finally analogs In-Reply-To: <191c3a030911061608w5be8af61y7bc10fe2d23dfc4a@mail.gmail.com> References: <191c3a030911061608w5be8af61y7bc10fe2d23dfc4a@mail.gmail.com> Message-ID: Closed. As (almost) usual the reason was me. Anthony's hint works perfectly: api uuid_transfer bridge:sofia/gateway// inline Sorry for bothering! On Sat, Nov 7, 2009 at 3:08 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > If you know the reason, why are you so puzzled by it? > I think you should not assume you understand what is happening unless you > really do. > > I think you need to provide an exact description of what you are doing so I > can explain to you where you are making the mistake. > > Make sure you are on latest SVN and reproduce this in a console log for us > and add an exact description of what you are doing in detail. > > > On Thu, Nov 5, 2009 at 11:44 AM, Artem Shiyanov wrote: > >> Hello! >> >> I have to deal with classic problem: "Leaking stream handle" in FS >> console. I also know the reason - firstly channel is sent to the extension >> with "playback" and later it is transfered to another extensions with >> "execute_extension" or, another trouble-case - channel is bridged to some >> addres. >> I do not ask (but I wish to) why FS doesn't close stream automatically >> when channel is gone. >> I ask whether it is possible to use some "try.. finally" construction in >> diaplan? If "yes" then I can simply stop playback in the "finally" block.. >> >> Any thoughs? >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/bda5ce2f/attachment.html From bruce at nani.ca Mon Nov 9 00:07:14 2009 From: bruce at nani.ca (Bruce Fletcher) Date: Mon, 9 Nov 2009 00:07:14 -0800 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> Message-ID: <2742E007-4C51-4E1A-96C6-B047F82174F2@nani.ca> The patch from the PortAudio site does get the library to build, but it still fails with the same assertion when I try to play MOH. The patch I'm talking about is this one: http://www.portaudio.com/trac/changeset/1418 If the same build problem applies to other 64 bit systems, it might be a good idea to incorporate this patch. It looks clean and reasonable to me, at least. I've managed to work around the problem entirely by building FreeSWITCH for i386, but I'll go ask the PortAudio folks what the status is of their 64 bit support. They are clearly assuming 32 bit long integers in some places, which is hopefully on a to-fix list somewhere. Thanks, - Bruce On 2009-11-08, at 12:25 PM, Michael Jerris wrote: > If you can figure out a clean way for us to do this with proper ifdefs > in tree in a way that will not break others that would be the most > preferred. > > Mike > > On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote: > >> OK, I'll ignore that MacPorts patch for now and try to find a better >> approach. >> >> I'll look into this further tonight, but this morning I found a more >> recent promising patch on the PortAudio site: >> >> http://www.portaudio.com/trac/changeset/1418 >> >> It seems to push some data types to 32 bit regardless of platform, >> which might work better than the MacPorts approach of migrating some >> data structures to 64 bit. At any rate, this patch being on the >> PortAudio site suggests it might be a more approved fix. >> >> I'll keep plugging at this in my free time and report any significant >> progress back to the list. >> >> Thanks, >> - Bruce >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From shouldbeq931 at googlemail.com Mon Nov 9 03:23:43 2009 From: shouldbeq931 at googlemail.com (shouldbe q931) Date: Mon, 9 Nov 2009 11:23:43 +0000 Subject: [Freeswitch-users] building on Fedora 12 Message-ID: <649eaa470911090323m543cca02td862834979d09949@mail.gmail.com> Hi, While I appreciate that Fedora 12 is still only in beta. Because I want to try FusionPBX and I've had no success getting pfSense to work in a single NIC environment, and FusionPBX needs PHP 5.3, and Fedora 12 appears to be the first distro with PHP5.3... Anyway, these are the steps that I've done yum install autoconf automake binutils bison curl-devel db4 db4-devel expat-devel flex gcc-c++ gettext gdb gdbm gdbm-devel gnutls-devel httpd kernel-devel libogg-devel libtiff libtiff-devel libtool libvorbis-devel make mkxauth mysql-server mysql mysql-devel ncurses ncurses-devel ncurses-libs openssl-devel perl-Apache2-SOAP perl-devel php php-common php-mysql php-pdo php-soap php-xmlrpc postfix python python-devel screen sqlite sqlite-devel sqlite2 sqlite2-devel subversion unixODBC-devel wget wireshark wireshark-gnome zlib zlib-devel mkdir /usr/src/freeswitch mkdir /usr/src/freeswitch/svn svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk /usr/src/freeswitch/svn cd /usr/src/freeswitch/svn ./bootstrap.sh ./configure make SVN reports Checked out external at revision 849. Checked out revision 15396. I then get the following error at the end -------------------------------------------------------------- Making all in soa Making all in tport LTCOMPILE tport_tls.lo cc1: warnings being treated as errors tport_tls.c: In function ?tls_init_context?: tport_tls.c:280: error: assignment discards qualifiers from pointer target type tport_tls.c:282: error: assignment discards qualifiers from pointer target type tport_tls.c: In function ?tls_post_connection_check?: tport_tls.c:527: error: assignment discards qualifiers from pointer target type make[9]: *** [tport_tls.lo] Error 1 make[8]: *** [all] Error 2 Making all in nta Making all in nth Making all in nea Making all in iptsec Making all in nua make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [//usr/src/freeswitch/svn/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 [root at conf-2 svn]# -------------------------------------------------------------- I then tried using the "quick and dirty" method from http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install, but get a similar failure -------------------------------------------------------------- Making all in nua make[9]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' make[10]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la make[10]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua' make[9]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[9]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua' make[8]: *** [all-recursive] Error 1 make[8]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua' Making all in packages make[8]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[8]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[7]: *** [all-recursive] Error 1 make[7]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[6]: *** [all] Error 2 make[6]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[5]: *** [/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[5]: Leaving directory `/usr/src/freeswitch.trunk/src/mod/endpoints/mod_sofia' make[4]: *** [mod_sofia-all] Error 1 make[4]: Leaving directory `/usr/src/freeswitch.trunk/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch.trunk/src' Making all in build make[3]: Entering directory `/usr/src/freeswitch.trunk/build' +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[3]: Leaving directory `/usr/src/freeswitch.trunk/build' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch.trunk' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch.trunk' make: *** [freeswitch] Error 2 [root at conf-2 src]# -------------------------------------------------------------- I'm still very much a noob when it comes to building, but I'd welcome any constructive suggestions on how to resolve this. Cheers Arne From lakindia89 at gmail.com Mon Nov 9 03:53:52 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 9 Nov 2009 17:23:52 +0530 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events Message-ID: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> Dear all, I did the below code, to callback a function when CHANNEL_EXECUTE_COMPLETE event comes. I executed the script for the 1st time and I got nothing. When I executed the script for the 2nd time, it ended with Sedmentation fault with core dumped. I was unable to attach the core dump file with this mail. Please specify how to send files to freeswitch user mailing list if need be. The freeswitch log is here: http://pastebin.freeswitch.org/11038 #!/usr/bin/perl use strict; use Data::Dumper; our $session; $session->answer(); my $events=new freeswitch::EventConsumer("CHANNEL_EXECUTE_COMPLETE"); $events->pop(1); $events->swig_e_callback_set("playvoice"); sub playvoice() { freeswitch::consoleLog("INFO","Call back function called\n"); } return 1; -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/858ab30b/attachment.html From edpimentl at gmail.com Mon Nov 9 04:05:53 2009 From: edpimentl at gmail.com (EdPimentl) Date: Mon, 9 Nov 2009 07:05:53 -0500 Subject: [Freeswitch-users] building on Fedora 12 In-Reply-To: <649eaa470911090323m543cca02td862834979d09949@mail.gmail.com> References: <649eaa470911090323m543cca02td862834979d09949@mail.gmail.com> Message-ID: <9dc4a1670911090405u56502857pcc647204dc1ffc4@mail.gmail.com> Any reason for not using uBuntu? Install Freeswitch + FusionPBX on Ubuntu step 1) add the fallowing lines to /etc/apt/ file. deb http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntuhardy main deb-src http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntuhardy main step 2) apt-get update step3) apt-get install freeswitch deps -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/4dfd7248/attachment.html From markmorreny at gmail.com Mon Nov 9 04:59:54 2009 From: markmorreny at gmail.com (mark morreny) Date: Mon, 9 Nov 2009 20:59:54 +0800 Subject: [Freeswitch-users] playback from hadoop Message-ID: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> Hi, Does anyone know how to playback based on files from hadoop storage. There is a libhdcp, and java api. Is there anyway to put together a sample middle piece to move files from hadoop to freeswitch using memory space, so there is no disk I/O? Any feedback or suggestion will be greatly appreciated. thx, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/bce7dde5/attachment.html From testeador01 at gmail.com Mon Nov 9 05:30:53 2009 From: testeador01 at gmail.com (Milena) Date: Mon, 9 Nov 2009 08:30:53 -0500 Subject: [Freeswitch-users] Valid Dial Strings In-Reply-To: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> Message-ID: Hello, When you post something on pastebin, please post the link to your post so everyone can find it, what is the link to it? Have a nice day :) 2009/11/7 Dave Stevenson > Hi Michael, > > thanks for the reply. I think that I have got to the bottom of how to allow > numbers to get to the VOIP gateway - at the moment, my dialplan just allows > any. > > The big problem is that the VOIP Gateway (Linksys 3102) rejects any calls > to > it from VOIP to the PSTN and I don't know why. > I have posted a dump to the pastebin, hopefully, the messages in there will > allow someone to see what the problem is and give me some pointers on how I > might fix it > > regards > Dave > > > > > ----- Original Message ----- > From: "Michael S Collins" > To: > Sent: Saturday, November 07, 2009 6:49 PM > Subject: Re: [Freeswitch-users] Valid Dial Strings > > > > > > On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" > > wrote: > > > >> Hi, > >> > >> can someone pointme to where the valid dialing strings are specified ? > >> > > For SIP dialstrings check here: > > http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings > > > > Also, if you send us examples of what you've tried we can help you > > figure out what's wrong. > > > >> I'm assuming that something, somewhere, tells FS that numbers are > >> invalid > >> before they get dialed ? > > > > Pastebin some debug logs of what's happening. Check out this page > > which has lots of useful information on how to collect information: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > > > It sounds like it's just a matter of figuring out how to configure > > your specific setup. Please report back with more information and > > we'll be happy to help. > > > > -MC > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/9870b09b/attachment-0001.html From shouldbeq931 at googlemail.com Mon Nov 9 05:48:26 2009 From: shouldbeq931 at googlemail.com (shouldbe q931) Date: Mon, 9 Nov 2009 13:48:26 +0000 Subject: [Freeswitch-users] building on Fedora 12 In-Reply-To: <9dc4a1670911090405u56502857pcc647204dc1ffc4@mail.gmail.com> References: <649eaa470911090323m543cca02td862834979d09949@mail.gmail.com> <9dc4a1670911090405u56502857pcc647204dc1ffc4@mail.gmail.com> Message-ID: <649eaa470911090548m9002e98l36def833e64b7c84@mail.gmail.com> Hi Ed, I installed Jaunty ( I don't have Hardy to hand) rather than /etc/apt, I presume you mean /etc/apt/sources.list after a "sudo apt-get update" I did a "sudo apt-get install freeswitch" I'm not sure what you meant by "deps" by your step 3 I then edited /etc/defaults/freeswitch and set false to true, saved the file and restarted, unfortunately freeswitch does not start, I had seen on the wiki that the debian packager was putting something in the wrong place, but I'm not sure where I would look for logs to show why it doesn't start. I will try and download a copy of hardy later on, and see if it has the same issues. Cheers Arne On Mon, Nov 9, 2009 at 12:05 PM, EdPimentl wrote: > Any reason for? not using uBuntu? > > Install Freeswitch + FusionPBX on Ubuntu > > step 1) add the fallowing?lines to /etc/apt/?file. > > deb > http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntu > hardy main > deb-src > http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntu > hardy main > > step 2) apt-get update > > step3) apt-get install freeswitch deps > > -E > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From stevendt at primrosebank.net Mon Nov 9 05:55:48 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 13:55:48 -0000 Subject: [Freeswitch-users] Valid Dial Strings References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> Message-ID: <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> Milena, thanks a lot for the reply - sorry, I'm new to this, but I'll remember that for next time. Actually, I found my way to the IRC site and the helpful chaps there got to the bottom of my problem. I had made an error copying the dialplan data from the "SPA3102 FreeSwitch HowTo" http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo I had read as i.e., substituted "(...)" for "{...}" Being unfamiliar with the FreeSwitch dialplans, I'd NEVER have found the problem without help, regards Dave ----- Original Message ----- From: Milena To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 1:30 PM Subject: Re: [Freeswitch-users] Valid Dial Strings Hello, When you post something on pastebin, please post the link to your post so everyone can find it, what is the link to it? Have a nice day :) 2009/11/7 Dave Stevenson Hi Michael, thanks for the reply. I think that I have got to the bottom of how to allow numbers to get to the VOIP gateway - at the moment, my dialplan just allows any. The big problem is that the VOIP Gateway (Linksys 3102) rejects any calls to it from VOIP to the PSTN and I don't know why. I have posted a dump to the pastebin, hopefully, the messages in there will allow someone to see what the problem is and give me some pointers on how I might fix it regards Dave ----- Original Message ----- From: "Michael S Collins" To: Sent: Saturday, November 07, 2009 6:49 PM Subject: Re: [Freeswitch-users] Valid Dial Strings > > On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" > wrote: > >> Hi, >> >> can someone pointme to where the valid dialing strings are specified ? >> > For SIP dialstrings check here: > http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings > > Also, if you send us examples of what you've tried we can help you > figure out what's wrong. > >> I'm assuming that something, somewhere, tells FS that numbers are >> invalid >> before they get dialed ? > > Pastebin some debug logs of what's happening. Check out this page > which has lots of useful information on how to collect information: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It sounds like it's just a matter of figuring out how to configure > your specific setup. Please report back with more information and > we'll be happy to help. > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/ac0e3bde/attachment.html From rupa at rupa.com Mon Nov 9 06:08:52 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 9 Nov 2009 06:08:52 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175B572244@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> <3C04B27FC880044F8FCD735D0D952FF7175B572244@EXMBXCLUS01.citservers.local> Message-ID: On Fri, Nov 6, 2009 at 7:59 AM, Ujjval Karihaloo wrote: > ?Any examples I can refer to for this? not that i know of > > Like for Channel vars and execute_application calls? Does this all need to be doen in dialplan.public.xml or also in other config files? most can be done in public.xml if you want to do it all in dialplan. > Sorry: I am still learning the Freeswitch world. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker > Sent: Thursday, November 05, 2009 8:56 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > This is true, BUT it is more flexible than it looks. > > http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E > > The caller controls can have a key execute a dialplan extension: > execute_application > You can set a channel var on the moderator prior to joining to the conf. > When the extenion is called, you can check the channel var for > moderator and act accordingly. > > Or you can send an event and monitor with an app over ESL and do > whatever you want there (probably using the same channel var trick for > knowing who is a mod or not). > > > On Thu, Nov 5, 2009 at 8:52 AM, Rob Forman wrote: >> Hi UK, >> >> ?From what I've done and read, the caller-controls (in >> conference.conf.xml) can be modified to almost anything you can think >> of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a >> caller control just for those with the moderator flag. ?So unless you >> want everyone able to mute/kick everyone then you can't do it. >> >> The wiki seems to indicate this as well: >> >> "Be aware that the caller-controls are applied across the entire >> conference. You cannot enter one member of the conference using caller- >> controls ABC and then enter a second member using caller-controls XYZ." >> >> http://wiki.freeswitch.org/wiki/Mod_conference >> >> >> I think this might be a limitation of mod_conference. ?Perhaps one of >> the pros can chime in if I'm off-base or there's some nifty way to >> accomplish this. >> >> Cheers, >> Rob >> >> On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: >> >>> Any ideas on the below...has anyone implemented the below: >>> >>> Once I have the Moderator and Participants logged on, how do I >>> invoke the moderator previlidges, LIk esay muting everyone/someone >>> or kicking someone out of the Conf and the like? >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, November 02, 2009 12:52 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Rob: >>> >>> ? Once I have the Moderator and Participants logged on, how do I >>> invoke the moderator previlidges, LIk esay muting everyone/someone >>> or kicking someone out of the Conf and the like? >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 9:34 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Hm, strange. ?I haven't seen that before. ?Can you pastebin your logs >>> at debug level? >>> >>> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >>> >>>> It's strange... a tcpdump tells me that there is no DTMF from my >>>> provider when using IVR, but when I call into a TN that goes >>>> directly into the Conference App, I see DTMF from the provider. >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman >>>> Sent: Friday, October 30, 2009 7:23 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> I've never had any problem with that. ?Is your logging at debug level >>>> so you can see the RECV DTFM in the log/fs_cli? ?Are you calling from >>>> a SIP phone on the pbx, or via a PSTN provider? ?Maybe your provider >>>> isn't passing them through. >>>> >>>> Make sure your logging is turned up then try something simpler, like >>>> calling the echo application, and see if DTFM comes through. >>>> >>>> Rob >>>> >>>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>>> >>>>> Rob: >>>>> >>>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>>> using the below configuration....so it basically timesout. >>>>> >>>>> UK >>>>> >>>>> >>>>> >>>>> -----Original Message----- >>>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>>> ] On Behalf Of Ujjval Karihaloo >>>>> Sent: Monday, October 26, 2009 9:21 AM >>>>> To: freeswitch-users at lists.freeswitch.org >>>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>>> >>>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>>> correct.. >>>>> =============== >>>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>>> with an IVR where the moderators are authenticated with an extra >>>>> pass- >>>>> code", which is what I did. ?I guess that's why I didn't understand >>>>> the point of the +pin. >>>>> ====================== >>>>> >>>>> I will try it out. >>>>> >>>>> Again thx a lot for your help. Will keep everyone posted. >>>>> >>>>> ________________________________________ >>>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>>> Sent: Friday, October 23, 2009 12:22 PM >>>>> To: freeswitch-users at lists.freeswitch.org >>>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>>> >>>>> I just re-tested with the pin in my dial plan: >>>>> >>>>> >>>>> >>>>> And it doesn't challenge me for the pin. ?I just drop right in. ?I >>>>> figured this is how it was intended, since the wiki says the pin is >>>>> set initially and only challenged in later attempts [by future >>>>> callers]: >>>>> >>>>> "The first time a conference name (confname) is used, it will be >>>>> created on demand, and the pin will be set to what ever is specified >>>>> at that time: the pin in the data string if specified, or if not, >>>>> the >>>>> "pin" setting in the conference profile, and if that is also >>>>> unspecified, then there is no pin protection. Any later attempt to >>>>> join the conference must specify the same pin number, if one existed >>>>> when it was created. " >>>>> >>>>> >>>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>>> with an IVR where the moderators are authenticated with an extra >>>>> pass- >>>>> code", which is what I did. ?I guess that's why I didn't understand >>>>> the point of the +pin. >>>>> >>>>> I'm sure there's a scenario where its used and useful, the wiki just >>>>> doesn't explain it. >>>>> >>>>> Rob >>>>> >>>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>>> >>>>>> Well first off you're not defining a pine here... >>>>>> >>>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>>> pin >>>>>> number] >>>>>> >>>>>> That might be why its not asking for a pin. >>>>>> >>>>>> /b >>>>>> >>>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>>> >>>>>>> ? >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From gshfreesw at gmail.com Mon Nov 9 06:44:16 2009 From: gshfreesw at gmail.com (Shameem Shiek) Date: Mon, 9 Nov 2009 09:44:16 -0500 Subject: [Freeswitch-users] SIP Provider with unlimited channels Message-ID: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> Dear Freeswitch Users, I am looking for a SIP Provider who can provide a DID with unlimited channels. Currently I am using junction networks but they have a high 2.9c/minute charge. I am looking for someone who has a flat rate for X minutes. Any advise would be much appreciated. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/8e762338/attachment.html From kristian.kielhofner at gmail.com Mon Nov 9 07:27:55 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 9 Nov 2009 10:27:55 -0500 Subject: [Freeswitch-users] SIP Provider with unlimited channels In-Reply-To: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> References: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> Message-ID: <2d9149cd0911090727j71de4c2bv9e5219ce37212037@mail.gmail.com> Beware of anyone that claims to offer "unlimited" channels. We're still fundamentally a TDM world and there is no such thing as unlimited. Depending on what you are looking for there are probably plenty of providers with a high enough limit to satisfy your actual needs. I just frown upon anyone claiming to offer "unlimited" channels because they either don't have an understanding of their true capacity (some tier X reseller who doesn't realize the upstream providers ALL have limits) or they don't care they are advertising falsely. Sure you don't really need "unlimited" channels. What are you looking for? 100? 1000? On Mon, Nov 9, 2009 at 9:44 AM, Shameem Shiek wrote: > Dear Freeswitch Users, > > ?I am looking for a SIP Provider who can provide? a DID with unlimited > channels. Currently I am using junction networks but they have a high > 2.9c/minute charge. I am looking for someone who has a flat rate for X > minutes. > > Any advise would be much appreciated. > > Thanks. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From kristian.kielhofner at gmail.com Mon Nov 9 07:30:53 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 9 Nov 2009 10:30:53 -0500 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information In-Reply-To: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> References: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> Message-ID: <2d9149cd0911090730j41c4738qe14fa9b14b9b2069@mail.gmail.com> This is for outbound calls, calling party name. The OP is talking about called party name, which is the neat feature of being able to update the display of the calling user with the name of the called user (instead of just displaying their numeric extension for the duration of the call). On Sun, Nov 8, 2009 at 9:26 PM, SP wrote: > before playing with mod_sofia, did you try the sip_cid_type variable? > > http://wiki.freeswitch.org/wiki/Variable_sip_cid_type > > On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine wrote: >> Hello, >> >> ??While?trying to display the called party name ?on SNOM phones I've found >> that the field sent to the phone needs to be changed slightly in order to >> make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects >> Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and Cisco >> work ok. Just wanted to let the developers know... >> >> ? And now a question: We have SNOM phones monitoring other extensions (BLF >> feature). When a call comes in, the monitoring phones get notification, but >> the name field (identity display) contains the calling extension number and >> not its display name. Can this be fixed? >> >> ??????????????????????????????? Thanks! __Yehavi: >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Shannon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Mon Nov 9 07:45:49 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Nov 2009 09:45:49 -0600 Subject: [Freeswitch-users] SIP Provider with unlimited channels In-Reply-To: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> References: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> Message-ID: <614163A1-E814-42CB-B953-218FB55D7F63@freeswitch.org> Have you tried Bandwidth.com or iCall? /b On Nov 9, 2009, at 8:44 AM, Shameem Shiek wrote: > Dear Freeswitch Users, > > I am looking for a SIP Provider who can provide a DID with > unlimited channels. Currently I am using junction networks but they > have a high 2.9c/minute charge. I am looking for someone who has a > flat rate for X minutes. > > Any advise would be much appreciated. > > Thanks. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Nov 9 08:01:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 10:01:00 -0600 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information In-Reply-To: <2d9149cd0911090730j41c4738qe14fa9b14b9b2069@mail.gmail.com> References: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> <2d9149cd0911090730j41c4738qe14fa9b14b9b2069@mail.gmail.com> Message-ID: <191c3a030911090801k79b42baeyf31059d67d41fcd@mail.gmail.com> If the patch is not received today it will not make it into 1.0.5 On Mon, Nov 9, 2009 at 9:30 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > This is for outbound calls, calling party name. The OP is talking > about called party name, which is the neat feature of being able to > update the display of the calling user with the name of the called > user (instead of just displaying their numeric extension for the > duration of the call). > > On Sun, Nov 8, 2009 at 9:26 PM, SP wrote: > > before playing with mod_sofia, did you try the sip_cid_type variable? > > > > http://wiki.freeswitch.org/wiki/Variable_sip_cid_type > > > > On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine > wrote: > >> Hello, > >> > >> While trying to display the called party name on SNOM phones I've > found > >> that the field sent to the phone needs to be changed slightly in order > to > >> make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects > >> Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and > Cisco > >> work ok. Just wanted to let the developers know... > >> > >> And now a question: We have SNOM phones monitoring other extensions > (BLF > >> feature). When a call comes in, the monitoring phones get notification, > but > >> the name field (identity display) contains the calling extension number > and > >> not its display name. Can this be fixed? > >> > >> Thanks! __Yehavi: > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > > > -- > > Shannon > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/63790bfb/attachment.html From rob4manhere at gmail.com Mon Nov 9 08:02:07 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Mon, 9 Nov 2009 10:02:07 -0600 Subject: [Freeswitch-users] SIP Provider with unlimited channels In-Reply-To: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> References: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> Message-ID: <613F14B9-1AF1-4FEE-A206-C3FCECE4DE33@gmail.com> I agree there is no such thing as unlimited. The three ways most SIP providers will structure pricing is 1) per minute (ie $0.02/minute), 2) per channel (ie $15/month) or 3) "unlimited" with a channel limit (ie $7/month for any amount of minutes but after two simultaneous channels its ring busy). If you don't want to be limited by channels, then it sounds like your only choice is per minute. Some providers will say their per-minute is unlimited channels but you should ask them what there hard limit is (because there is one) and if it can be increased should you need more capacity. I've used iCall and am pretty happy with their service and pricing choices. I think their default channel limit on per-minute billing is 100. On Nov 9, 2009, at 8:44 AM, Shameem Shiek wrote: > Dear Freeswitch Users, > > I am looking for a SIP Provider who can provide a DID with > unlimited channels. Currently I am using junction networks but they > have a high 2.9c/minute charge. I am looking for someone who has a > flat rate for X minutes. > > Any advise would be much appreciated. > > Thanks. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shouldbeq931 at googlemail.com Mon Nov 9 08:08:33 2009 From: shouldbeq931 at googlemail.com (shouldbe q931) Date: Mon, 9 Nov 2009 16:08:33 +0000 Subject: [Freeswitch-users] building on Fedora 12 In-Reply-To: <649eaa470911090548m9002e98l36def833e64b7c84@mail.gmail.com> References: <649eaa470911090323m543cca02td862834979d09949@mail.gmail.com> <9dc4a1670911090405u56502857pcc647204dc1ffc4@mail.gmail.com> <649eaa470911090548m9002e98l36def833e64b7c84@mail.gmail.com> Message-ID: <649eaa470911090808g2d2e2b97nccf627533a49d2c3@mail.gmail.com> Hi Ed, I've just finished installing Hardy, and following the same steps again, freeswitch is not running. Any suggestions ? On Mon, Nov 9, 2009 at 1:48 PM, shouldbe q931 wrote: > Hi Ed, > > I installed Jaunty ( I don't have Hardy to hand) > > rather than /etc/apt, I presume you mean /etc/apt/sources.list > > after a "sudo apt-get update" I did a "sudo apt-get install > freeswitch" I'm not sure what you meant by "deps" by your step 3 > > I then edited /etc/defaults/freeswitch and set false to true, saved > the file and restarted, unfortunately freeswitch does not start, I had > seen on the wiki that the debian packager was putting something in the > wrong place, but I'm not sure where I would look for logs to show why > it doesn't start. > > I will try and download a copy of hardy later on, and see if it has > the same issues. > > Cheers > > Arne > > > On Mon, Nov 9, 2009 at 12:05 PM, EdPimentl wrote: >> Any reason for? not using uBuntu? >> >> Install Freeswitch + FusionPBX on Ubuntu >> >> step 1) add the fallowing?lines to /etc/apt/?file. >> >> deb >> http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntu >> hardy main >> deb-src >> http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntu >> hardy main >> >> step 2) apt-get update >> >> step3) apt-get install freeswitch deps >> >> -E >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From rob4manhere at gmail.com Mon Nov 9 08:19:27 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Mon, 9 Nov 2009 10:19:27 -0600 Subject: [Freeswitch-users] javascript parameter In-Reply-To: <200911090034059066890@gmail.com> References: <200911090034059066890@gmail.com> Message-ID: <0323900B-D84A-4872-8690-728D62C74BC7@gmail.com> You can check the numbers of arguments passed with argc, and access them via argv[0], argv[1], etc. Its hinted at on the main Javascript wiki page, and also detailed in the FAQ. http://wiki.freeswitch.org/wiki/Javascript_FAQ On Nov 8, 2009, at 10:34 AM, god.nirvana wrote: > hi all: > how can i get the value of the myArg1 myArg2 in test.js. > like this originate sofia/example/1000 at somewhere.com > '&javascript(test.js myArg1 myArg2)' > thanks! > > 2009-11-09 > god.nirvana > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/1117c449/attachment.html From anthony.minessale at gmail.com Mon Nov 9 08:34:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 10:34:58 -0600 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events In-Reply-To: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> References: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> Message-ID: <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> 1) install gdb 2) run support_d/fscore_db in the tree from the working directory of the core. 3) if you are not on svn trunk, "make current" and start over. On Mon, Nov 9, 2009 at 5:53 AM, lakshmanan ganapathy wrote: > Dear all, > I did the below code, to callback a function when CHANNEL_EXECUTE_COMPLETE > event comes. > I executed the script for the 1st time and I got nothing. > When I executed the script for the 2nd time, it ended with Sedmentation > fault with core dumped. > > I was unable to attach the core dump file with this mail. > Please specify how to send files to freeswitch user mailing list if need > be. > > The freeswitch log is here: > http://pastebin.freeswitch.org/11038 > > #!/usr/bin/perl > use strict; > use Data::Dumper; > our $session; > $session->answer(); > my $events=new freeswitch::EventConsumer("CHANNEL_EXECUTE_COMPLETE"); > $events->pop(1); > $events->swig_e_callback_set("playvoice"); > sub playvoice() > { > freeswitch::consoleLog("INFO","Call back function called\n"); > } > return 1; > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/1561093e/attachment.html From anthony.minessale at gmail.com Mon Nov 9 08:50:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 10:50:40 -0600 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <2742E007-4C51-4E1A-96C6-B047F82174F2@nani.ca> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> <2742E007-4C51-4E1A-96C6-B047F82174F2@nani.ca> Message-ID: <191c3a030911090850j4d9e7972m15fa4661a2da7926@mail.gmail.com> maybe we should write a new audio abstraction lib =D On Mon, Nov 9, 2009 at 2:07 AM, Bruce Fletcher wrote: > The patch from the PortAudio site does get the library to build, but > it still fails with the same assertion when I try to play MOH. The > patch I'm talking about is this one: > > http://www.portaudio.com/trac/changeset/1418 > > If the same build problem applies to other 64 bit systems, it might be > a good idea to incorporate this patch. It looks clean and reasonable > to me, at least. > > I've managed to work around the problem entirely by building > FreeSWITCH for i386, but I'll go ask the PortAudio folks what the > status is of their 64 bit support. They are clearly assuming 32 bit > long integers in some places, which is hopefully on a to-fix list > somewhere. > > Thanks, > - Bruce > > > On 2009-11-08, at 12:25 PM, Michael Jerris wrote: > > > If you can figure out a clean way for us to do this with proper ifdefs > > in tree in a way that will not break others that would be the most > > preferred. > > > > Mike > > > > On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote: > > > >> OK, I'll ignore that MacPorts patch for now and try to find a better > >> approach. > >> > >> I'll look into this further tonight, but this morning I found a more > >> recent promising patch on the PortAudio site: > >> > >> http://www.portaudio.com/trac/changeset/1418 > >> > >> It seems to push some data types to 32 bit regardless of platform, > >> which might work better than the MacPorts approach of migrating some > >> data structures to 64 bit. At any rate, this patch being on the > >> PortAudio site suggests it might be a more approved fix. > >> > >> I'll keep plugging at this in my free time and report any significant > >> progress back to the list. > >> > >> Thanks, > >> - Bruce > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/6b131013/attachment.html From codecomplete at free.fr Mon Nov 9 09:01:45 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 9 Nov 2009 09:01:45 -0800 (PST) Subject: [Freeswitch-users] cd-sounds vs. sounds? Message-ID: <26269842.post@talk.nabble.com> Hello I successfully installed FreeSwitch from SVN, and am now prompted to install the sound files. Am I correct in understanding that "sounds" are POTS-grade files (8KHz?) while "cd-sounds" are closer to VoIP-grade (16KHz?), and "hd-sounds" and "uhd-sounds" are for Skype-grade sound files? In that case, if I only need to play sound files to POTS callers, I only need the "sounds" files? Thank you. -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26269842.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Nov 9 09:34:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 09:34:17 -0800 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <26269842.post@talk.nabble.com> References: <26269842.post@talk.nabble.com> Message-ID: <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> On Mon, Nov 9, 2009 at 9:01 AM, Fred-145 wrote: > > Hello > > I successfully installed FreeSwitch from SVN, and am now prompted to > install > the sound files. Am I correct in understanding that "sounds" are POTS-grade > files (8KHz?) while "cd-sounds" are closer to VoIP-grade (16KHz?), and > "hd-sounds" and "uhd-sounds" are for Skype-grade sound files? > > In that case, if I only need to play sound files to POTS callers, I only > need the "sounds" files? > I recommend you just do this: make cd-sounds-install && make cd-moh-install This will install all the sounds (8k, 16k, 32k, 48k) and thus FS can play whatever sampling rate is necessary. I promise you that this is the easiest way to go. Others have tried to save a few megabytes of disk space by not installing the higher quality sounds and music and have ended up wasting hours debugging "issues with playing sound files." For the record, the 8k files are just "sounds" and then you have hd (16k), uhd (32k), and cd (48k). -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/b1318cf7/attachment.html From msc at freeswitch.org Mon Nov 9 09:37:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 09:37:46 -0800 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo Wiki - HELP Please! In-Reply-To: <7B9E5C0A81154EDB8B2F2979EAF0BA17@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> <2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> <1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com> <665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> <87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com> <97FBB4B6002848BCA4F2D89F13626754@bp1.ad.bp.com> <41A5CF92E4E94547BCE301E0F5A5B79B@bp1.ad.bp.com> <7B9E5C0A81154EDB8B2F2979EAF0BA17@bp1.ad.bp.com> Message-ID: <87f2f3b90911090937t39d74fc7h3f887bac28c9e1b5@mail.gmail.com> Two quick questions: which version of FreeSWITCH are you running? If you're not on the latest then I recommend getting SVN trunk or at the very least the latest prerelease. Second, can you capture the debug information and use pastebin? It makes it much easier for everyone to help you. http://pastebin.freeswitch.org. Also, this page might be helpful: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has handy tips on collecting information and asking the community for assistance. -MC On Sat, Nov 7, 2009 at 6:42 AM, Dave Stevenson wrote: > Follow up to previous post..... regarding making outgoing calls. > > I ***think*** that I have configured a dialplan that allows the user to > dial out but the requests seem to be getting rejected by the SPA3102. > > I can dial 0 and the FreeSwitch attendant will connect to the PSTN line > (FreeSwitch reports that the call has been answered). > Similarly, I can dial 1000 - the SPA3102 extension number with similar > results. > > However, if a try to dial an external number, the gateway rejects the call. > > I have captured some of the debug log but the info in there is way over my > head, can anyone help me understand what it's telling me please ? > > regards > Dave > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/78acf0ba/attachment-0001.html From ddechev at nutel.cc Mon Nov 9 02:10:26 2009 From: ddechev at nutel.cc (Dimitar Dechev) Date: Mon, 9 Nov 2009 12:10:26 +0200 Subject: [Freeswitch-users] Monitoring via SNMP Message-ID: <001001ca6124$dc9268e0$95b73aa0$@cc> Dear All, I couldn't find much information about how to monitor Freeswitch via SNMP like how many calls/legs I have, how many CAPs, and etc. One of the thing I do currently is to make simple bash script which in general runs "fs_cli -x 'show calls count'" or some other command and call that script via snmpd.conf. I would appreciate if somebody tell me if they is snmp module in Freeswitch, or provide with link/method from where I can read some information. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/b6c4f2fa/attachment.html From dharding at nucleus.com Mon Nov 9 04:10:56 2009 From: dharding at nucleus.com (Dana Harding) Date: Mon, 9 Nov 2009 05:10:56 -0700 Subject: [Freeswitch-users] suggestions for hardware. References: <4AF4AF73.8070804@tx.rr.com> Message-ID: <11A13463A6CB4AC196401DA7AE7E6F34@danahome> > I am looking on advice on how to set up a simple office PBX, 20 phones > and 4 outside lines.with 2 or 3 "operator" phones and the rest will be > extensions. > > 4 spa3000's to handle the outside lines. > 2-3 polycom 601 phones with expansion modules (Operator phones) > 18 polycom 330 or other phones for desks. > 2-24 port cisco POE switches > 1 pfSense server. The last time I tried using spa3000's for PSTN connections, I had a lot of difficulty tuning the settings to get rid of some echo the users were hearing. I don't know what the local loop length was, and the wiring in that neighbourhood is pretty old - both probably had a strong influence on what I was seeing. My determination at that time (~3 years ago) was that they were good for home connections, but not suitable for business use. I ultimately went with a Sangoma card (A200 +HWEC module), it killed the echo and sounded great. YMMV - Firmware and hardware has probably changed quite a bit in 3 years, and your loop characteristics might be better. My inclination would be to buy one, and see how well it works at your site. > System Design. > > Extension Numbers 2xx > Outside line access 1xxxxxxxxxx > groups 3xx > auto-attendent ??? Depending on how you want your system to work - auto-attendant is a good way to go if you won't have DIDs for all or most individuals. Our receptionist's (and our client's) time was being chewed up by all calls being forced to go through her before we implemented an auto-attendant. From shouldbeq931 at googlemail.com Mon Nov 9 09:48:16 2009 From: shouldbeq931 at googlemail.com (shouldbe q931) Date: Mon, 9 Nov 2009 17:48:16 +0000 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> Message-ID: <649eaa470911090948v34ee9239l8eb98ba964d42ad7@mail.gmail.com> While I'm very happy to hear this, the wiki has in more than one place suggestions to install multiple "sound and moh" 'sets'... On Mon, Nov 9, 2009 at 5:34 PM, Michael Collins wrote: > > > On Mon, Nov 9, 2009 at 9:01 AM, Fred-145 wrote: >> >> Hello >> >> I successfully installed FreeSwitch from SVN, and am now prompted to >> install >> the sound files. Am I correct in understanding that "sounds" are >> POTS-grade >> files (8KHz?) while "cd-sounds" are closer to VoIP-grade (16KHz?), and >> "hd-sounds" and "uhd-sounds" are for Skype-grade sound files? >> >> In that case, if I only need to play sound files to POTS callers, I only >> need the "sounds" files? > > I recommend you just do this: > make cd-sounds-install && make cd-moh-install > > This will install all the sounds (8k, 16k, 32k, 48k) and thus FS can play > whatever sampling rate is necessary. I promise you that this is the easiest > way to go. Others have tried to save a few megabytes of disk space by not > installing the higher quality sounds and music and have ended up wasting > hours debugging "issues with playing sound files." > > For the record, the 8k files are just "sounds" and then you have hd (16k), > uhd (32k), and cd (48k). > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Nov 9 10:13:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 10:13:47 -0800 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <649eaa470911090948v34ee9239l8eb98ba964d42ad7@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> <649eaa470911090948v34ee9239l8eb98ba964d42ad7@mail.gmail.com> Message-ID: <87f2f3b90911091013i1f74fa87s2b9f79112e67bc48@mail.gmail.com> On Mon, Nov 9, 2009 at 9:48 AM, shouldbe q931 wrote: > While I'm very happy to hear this, the wiki has in more than one place > suggestions to install multiple "sound and moh" 'sets'... > > Link(s) please? I'll take care of the wiki. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/85e066e1/attachment.html From shouldbeq931 at googlemail.com Mon Nov 9 10:28:16 2009 From: shouldbeq931 at googlemail.com (shouldbe q931) Date: Mon, 9 Nov 2009 18:28:16 +0000 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <87f2f3b90911091013i1f74fa87s2b9f79112e67bc48@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> <649eaa470911090948v34ee9239l8eb98ba964d42ad7@mail.gmail.com> <87f2f3b90911091013i1f74fa87s2b9f79112e67bc48@mail.gmail.com> Message-ID: <649eaa470911091028s202aad27x351d09b0f6925b0f@mail.gmail.com> I was sure I'd seen more, but http://wiki.freeswitch.org/wiki/Installation_Guide search for "There are also higher bitrate sounds available for download and installation with:" Cheers Arne On Mon, Nov 9, 2009 at 6:13 PM, Michael Collins wrote: > > > On Mon, Nov 9, 2009 at 9:48 AM, shouldbe q931 > wrote: >> >> While I'm very happy to hear this, the wiki has in more than one place >> suggestions to install multiple "sound and moh" 'sets'... >> > Link(s) please? I'll take care of the wiki. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Nov 9 10:32:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 10:32:03 -0800 Subject: [Freeswitch-users] Monitoring via SNMP In-Reply-To: <001001ca6124$dc9268e0$95b73aa0$@cc> References: <001001ca6124$dc9268e0$95b73aa0$@cc> Message-ID: <87f2f3b90911091032h423890a9oea02c23674e2f3f4@mail.gmail.com> 2009/11/9 Dimitar Dechev > Dear All, > > > > I couldn?t find much information about how to monitor Freeswitch via SNMP > like how many calls/legs I have, how many CAPs, and etc. One of the thing I > do currently is to make simple bash script which in general runs ?fs_cli -x > ?show calls count?? or some other command and call that script via > snmpd.conf. > > > > I would appreciate if somebody tell me if they is snmp module in > Freeswitch, or provide with link/method from where I can read some > information. > > > > Thanks! > Sorry, no SNMP. However, there is something with a little more power: the event socket. You can use the ESL (event socket library) to write your own monitor script. More info: http://wiki.freeswitch.org/wiki/Event_Socket Also, if you really wanted to, you could do the "fs_cli -x 'foo bar'" stuff but there are more elegant and powerful ways of handing this. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/41e5692d/attachment.html From msc at freeswitch.org Mon Nov 9 10:32:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 10:32:32 -0800 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <649eaa470911091028s202aad27x351d09b0f6925b0f@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> <649eaa470911090948v34ee9239l8eb98ba964d42ad7@mail.gmail.com> <87f2f3b90911091013i1f74fa87s2b9f79112e67bc48@mail.gmail.com> <649eaa470911091028s202aad27x351d09b0f6925b0f@mail.gmail.com> Message-ID: <87f2f3b90911091032j4fbec27fi8d1e2dc3aa212d45@mail.gmail.com> On Mon, Nov 9, 2009 at 10:28 AM, shouldbe q931 wrote: > I was sure I'd seen more, but > http://wiki.freeswitch.org/wiki/Installation_Guide search for "There > are also higher bitrate sounds available for download and installation > with:" > > Cheers > > Arne > > Thanks! I'll clean that up a bit. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/7ed1e993/attachment.html From codecomplete at free.fr Mon Nov 9 10:39:45 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 9 Nov 2009 10:39:45 -0800 (PST) Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> Message-ID: <26271417.post@talk.nabble.com> mercutioviz wrote: > > I recommend you just do this: make cd-sounds-install && make > cd-moh-install > Will do. Thanks. -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26271417.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jmesquita at freeswitch.org Mon Nov 9 10:43:36 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 9 Nov 2009 16:43:36 -0200 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <191c3a030911090850j4d9e7972m15fa4661a2da7926@mail.gmail.com> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> <2742E007-4C51-4E1A-96C6-B047F82174F2@nani.ca> <191c3a030911090850j4d9e7972m15fa4661a2da7926@mail.gmail.com> Message-ID: Or write one for Mac specifically since PA is fine for all the rest (I think)? JM On Mon, Nov 9, 2009 at 2:50 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > maybe we should write a new audio abstraction lib =D > > > On Mon, Nov 9, 2009 at 2:07 AM, Bruce Fletcher wrote: > >> The patch from the PortAudio site does get the library to build, but >> it still fails with the same assertion when I try to play MOH. The >> patch I'm talking about is this one: >> >> http://www.portaudio.com/trac/changeset/1418 >> >> If the same build problem applies to other 64 bit systems, it might be >> a good idea to incorporate this patch. It looks clean and reasonable >> to me, at least. >> >> I've managed to work around the problem entirely by building >> FreeSWITCH for i386, but I'll go ask the PortAudio folks what the >> status is of their 64 bit support. They are clearly assuming 32 bit >> long integers in some places, which is hopefully on a to-fix list >> somewhere. >> >> Thanks, >> - Bruce >> >> >> On 2009-11-08, at 12:25 PM, Michael Jerris wrote: >> >> > If you can figure out a clean way for us to do this with proper ifdefs >> > in tree in a way that will not break others that would be the most >> > preferred. >> > >> > Mike >> > >> > On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote: >> > >> >> OK, I'll ignore that MacPorts patch for now and try to find a better >> >> approach. >> >> >> >> I'll look into this further tonight, but this morning I found a more >> >> recent promising patch on the PortAudio site: >> >> >> >> http://www.portaudio.com/trac/changeset/1418 >> >> >> >> It seems to push some data types to 32 bit regardless of platform, >> >> which might work better than the MacPorts approach of migrating some >> >> data structures to 64 bit. At any rate, this patch being on the >> >> PortAudio site suggests it might be a more approved fix. >> >> >> >> I'll keep plugging at this in my free time and report any significant >> >> progress back to the list. >> >> >> >> Thanks, >> >> - Bruce >> >> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/b7faf0f2/attachment-0001.html From stevendt at primrosebank.net Mon Nov 9 10:44:27 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 18:44:27 -0000 Subject: [Freeswitch-users] Cordless VOIP Phones References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> Message-ID: <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Hi, has anyone any good results to share with using cordless phones for VOIP with FreeSwitch ? I have seen a few around that appear to operate with wireless networks and make SIP connections to VOIP PBXs. I have seen various models from Engenius, Prestige, DORO and Siemens as well as Snom. regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/4db387aa/attachment.html From jmesquita at freeswitch.org Mon Nov 9 10:50:02 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 9 Nov 2009 16:50:02 -0200 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: I have Siemens A58IP and Snom M3. Both work very well with pros and cons. Nonetheless, both lack HD .... JM On Mon, Nov 9, 2009 at 4:44 PM, Dave Stevenson wrote: > Hi, > > has anyone any good results to share with using cordless phones for VOIP > with FreeSwitch ? > > I have seen a few around that appear to operate with wireless networks and > make SIP connections to VOIP PBXs. > > I have seen various models from Engenius, Prestige, DORO and Siemens as > well as Snom. > > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/1eb79f3a/attachment.html From djbinter at yahoo.com Mon Nov 9 11:05:43 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 9 Nov 2009 11:05:43 -0800 (PST) Subject: [Freeswitch-users] Another Concurrent Calls Monitor Question Message-ID: <158503.71880.qm@web37504.mail.mud.yahoo.com> I am also curious whether you can recommend how I can get the info if I want to see concurrent calls by account code. Let's say if I am running FS as SBC and I want to monitor concurrent calls per customer. I've looked at the HEARTBEAT, but it only gives me overall session-count. How safe is it if I have a cron job running every 5 minutes, and get the data from core.db in the calls tables. For instance, if I issue the following query: select count(*) from calls where substr(callee_chan_name,27,15)='$gw_ip_addr'; I don't want to query directly from core.db, so it would be great if I can use something from event socket to monitor per customer (account code) or ip address. Thank you. From codecomplete at free.fr Mon Nov 9 11:16:27 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 9 Nov 2009 11:16:27 -0800 (PST) Subject: [Freeswitch-users] Right way to start FS on CentOS at boot-time? Message-ID: <26272066.post@talk.nabble.com> Hello For those of you running FS on CentOS (5.4) who compiled FS from SVN, I'd like to make sure I'm doing it right to have FS start automatically at boot-time: 1. cp /usr/src/freeswitch/build/freeswitch.init.redhat /etc/init.d/freeswitch 2. vi /etc/init.d/freeswitch: PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} 3. chmod 755 /etc/init.d/freeswitch 4. chkconfig --level 345 freeswitch on 5. chkconfig --list freeswitch 6. (why needed in addition to chkconfig?) ln -s /etc/init.d/freeswitch /usr/sbin/rcfreeswitch To manually start the server: Launch the server through the rc.d script: /etc/init.d/freeswitch start Is the above correct? Thank you. -- View this message in context: http://old.nabble.com/Right-way-to-start-FS-on-CentOS-at-boot-time--tp26272066p26272066.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From bruce at nani.ca Mon Nov 9 11:23:25 2009 From: bruce at nani.ca (Bruce Fletcher) Date: Mon, 9 Nov 2009 11:23:25 -0800 Subject: [Freeswitch-users] 64 bit PortAudio status In-Reply-To: References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> <2742E007-4C51-4E1A-96C6-B047F82174F2@nani.ca> <191c3a030911090850j4d9e7972m15fa4661a2da7926@mail.gmail.com> Message-ID: I just want to clarify the status of PortAudio on 64 bit architectures. There is a compile-time problem in pa_dither.c (and .h) that comes from the code not being 64 bit ready. This problem has been patched cleanly here: http://www.portaudio.com/trac/changeset/1418 I think this patch should go into FreeSWITCH because it makes PortAudio compilable and minimally useful on 64 bit platforms. There is a separate 64 bit issue involving PortAudio's Mac interface which shows up at runtime in the function ringBufferIOProc(). Someone marked it as a to-fix issue for 64 bit compiles with the following assertion: assert( sizeof( UInt32 ) == sizeof( long ) ); There is a discussion that just started this weekend about fixing this on the PortAudio mailing list here: http://music.columbia.edu/pipermail/portaudio/2009-November/009581.html It sounds like someone there may fix up this problem soon. I am going to keep tracking that and see if I can help it along. In any event, this problem is unrelated to the pa_dither problem. I just wanted to clarify this so the existing PortAudio patch for pa_dither could possibly be included before 1.05 is released. Thanks, - Bruce From andrew at hijacked.us Mon Nov 9 11:29:05 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 9 Nov 2009 14:29:05 -0500 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> Message-ID: <20091109192904.GI9418@hijacked.us> On Mon, Nov 09, 2009 at 08:59:54PM +0800, mark morreny wrote: > Hi, > > Does anyone know how to playback based on files from hadoop storage. > > There is a libhdcp, and java api. Is there anyway to put together a sample > middle piece to move files from hadoop to freeswitch using memory space, so > there is no disk I/O? > > Any feedback or suggestion will be greatly appreciated. > mod_shell_stream might work, if you can just spit out the raw audio to the shell. Or write another stream module that works with libhdcp. Andrew From stevendt at primrosebank.net Mon Nov 9 11:38:03 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 19:38:03 -0000 Subject: [Freeswitch-users] Cordless VOIP Phones References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: Joao, thanks for the note. The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ? Interesting about the HD, but do you notice the difference and find that you're disappointed with the quality of their sounds ? regards Dave ----- Original Message ----- From: Jo?o Mesquita To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 6:50 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones I have Siemens A58IP and Snom M3. Both work very well with pros and cons. Nonetheless, both lack HD .... JM On Mon, Nov 9, 2009 at 4:44 PM, Dave Stevenson wrote: Hi, has anyone any good results to share with using cordless phones for VOIP with FreeSwitch ? I have seen a few around that appear to operate with wireless networks and make SIP connections to VOIP PBXs. I have seen various models from Engenius, Prestige, DORO and Siemens as well as Snom. regards Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/34e10e68/attachment.html From brian at freeswitch.org Mon Nov 9 12:05:07 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Nov 2009 14:05:07 -0600 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote: > Joao, > > thanks for the note. The Snom M3 is one of the ones that I was > looking at - I would be interested in the "Pro's & Cons" ? RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN > > Interesting about the HD, but do you notice the difference and find > that you're disappointed with the quality of their sounds ? Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements. > > regards > Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/98c02fb6/attachment-0001.html From stevendt at primrosebank.net Mon Nov 9 12:15:45 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 20:15:45 -0000 Subject: [Freeswitch-users] Cordless VOIP Phones References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> Message-ID: <4E593A21A8B9463FB86AAC3B5ED5A941@bp1.ad.bp.com> Hi, thanks Brian, that's interesting. I had a comment "off list" which suggested the same thing. It did not quite fit with my aspiration for an all VOIP solution, but it sounds like the technology is not quite there yet for hands-free. That's great feedback before I spend some cash on a hands-free VOIP phone regards Dave ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 8:05 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote: Joao, thanks for the note. The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ? RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN Interesting about the HD, but do you notice the difference and find that you're disappointed with the quality of their sounds ? Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements. regards Dave ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/0804c928/attachment.html From rupa at rupa.com Mon Nov 9 12:19:42 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 9 Nov 2009 12:19:42 -0800 Subject: [Freeswitch-users] Another Concurrent Calls Monitor Question In-Reply-To: <158503.71880.qm@web37504.mail.mud.yahoo.com> References: <158503.71880.qm@web37504.mail.mud.yahoo.com> Message-ID: Use mod limit to do this. You can choose to use it in count only mode if you want (no limit). On Mon, Nov 9, 2009 at 11:05 AM, DJB wrote: > I am also curious whether you can recommend how I can get the info if I want to see concurrent calls by account code. ?Let's say if I am running FS as SBC and I want to monitor concurrent calls per customer. ?I've looked at the HEARTBEAT, but it only gives me overall session-count. > > How safe is it if I have a cron job running every 5 minutes, and get the data from core.db in the calls tables. ?For instance, if I issue the following query: > select count(*) from calls where substr(callee_chan_name,27,15)='$gw_ip_addr'; > > I don't want to query directly from core.db, so it would be great if I can use something from event socket to monitor per customer (account code) or ip address. > > Thank you. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From stevendt at primrosebank.net Mon Nov 9 12:22:15 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 20:22:15 -0000 Subject: [Freeswitch-users] RegEx Help References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> I **think** that the following will match any three character strings from 1xx to 399 I want to exclude 100 though, can anyone help me with the required RegEx please ? ^([1-3][0-9][0-9])$ I could (I think) do ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ But it does not "feel" elegant - is there a better way ? regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/e179c576/attachment.html From rupa at rupa.com Mon Nov 9 12:23:05 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 9 Nov 2009 12:23:05 -0800 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> References: <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> Message-ID: I agree about the M3. I have the handset and it is not ergonomic at all. I also have a Siemens A580-IP. It does do G722 but has a few bugs related to G722 that I normally run it with G711 only. There is a quality difference between G722 and G711 when talking among the A580 handsets or the handset and my Polycom 450. Of the DECT handsets, I'd recommend the Siemens. It is also very inexpensive. Beware it has a 3 concurrent calls limit per base-station. On Mon, Nov 9, 2009 at 12:05 PM, Brian West wrote: > > On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote: > > Joao, > > thanks for the note. The Snom M3 is one of the ones that I was looking at - > I would be interested in the "Pro's & Cons" ? > > RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN > > > Interesting about the HD, but do you?notice the difference and find that > you're disappointed with the quality of their sounds ? > > Get an ATA with a Dect handset it works much better... the Snom M3 and the > Aastra are one in the same and they both do not live up to the quality or > usability requirements. > > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From hads at nice.net.nz Mon Nov 9 12:26:41 2009 From: hads at nice.net.nz (Hadley Rich) Date: Tue, 10 Nov 2009 09:26:41 +1300 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> Message-ID: <1257798401.10738.18.camel@sodium> On Mon, 2009-11-09 at 14:05 -0600, Brian West wrote: > Get an ATA with a Dect handset it works much better... the Snom M3 and > the Aastra are one in the same and they both do not live up to the > quality or usability requirements. That said, they are better than what else is around. I'd call them average. Nothing to write home about but you don't need to run away from them. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier From stevecrozz at gmail.com Mon Nov 9 12:37:15 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 9 Nov 2009 12:37:15 -0800 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> Message-ID: <11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> Would something like this work for you? --Stephen On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson wrote: > I **think** that the following will match any three character strings > from 1xx to 399 > > I want to exclude 100 though, can anyone help me with the required RegEx > please ? > > > ^([1-3][0-9][0-9])$ > > I could (I think) do > > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ > > But it does not "feel" elegant - is there a better way ? > > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/b97dab90/attachment.html From msc at freeswitch.org Mon Nov 9 12:38:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 12:38:17 -0800 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> Message-ID: <87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson wrote: > I **think** that the following will match any three character strings > from 1xx to 399 > > I want to exclude 100 though, can anyone help me with the required RegEx > please ? > > > ^([1-3][0-9][0-9])$ > > I could (I think) do > > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ > > But it does not "feel" elegant - is there a better way ? > > expression="^[1-3]\d\d$" will match 100 to 399 -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/a48f1d30/attachment-0001.html From stevendt at primrosebank.net Mon Nov 9 12:55:30 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 20:55:30 -0000 Subject: [Freeswitch-users] RegEx Help References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> Message-ID: <6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com> Hi Stephen, thanks for the reply. I'm not sure , does the code below handle all number from 101 to 399 ? It would rely on the 100 code being picked up by the dialplan before the other extensions were processed so the order of the code in the dialplan is significant. Is that how people normally write their code, i.e., the extension processing is position dependant in the file ? regards Dave ----- Original Message ----- From: Stephen Crosby To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 8:37 PM Subject: Re: [Freeswitch-users] RegEx Help Would something like this work for you? --Stephen On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson wrote: I **think** that the following will match any three character strings from 1xx to 399 I want to exclude 100 though, can anyone help me with the required RegEx please ? ^([1-3][0-9][0-9])$ I could (I think) do ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ But it does not "feel" elegant - is there a better way ? regards Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/a3899507/attachment.html From stevendt at primrosebank.net Mon Nov 9 12:55:34 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 20:55:34 -0000 Subject: [Freeswitch-users] RegEx Help References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> Message-ID: <56D7DCE6AE1844A3B47AE8E351733B9C@bp1.ad.bp.com> Thanks Michael, but I want to exclude 100 ? regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 8:38 PM Subject: Re: [Freeswitch-users] RegEx Help On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson wrote: I **think** that the following will match any three character strings from 1xx to 399 I want to exclude 100 though, can anyone help me with the required RegEx please ? ^([1-3][0-9][0-9])$ I could (I think) do ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ But it does not "feel" elegant - is there a better way ? expression="^[1-3]\d\d$" will match 100 to 399 -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/9d936bfc/attachment.html From federico.omoto at gmail.com Mon Nov 9 11:50:14 2009 From: federico.omoto at gmail.com (Fede) Date: Mon, 9 Nov 2009 17:50:14 -0200 Subject: [Freeswitch-users] Doddle Web SIP phone Message-ID: <8b4221f20911091150k7f3d01eem3c5eae845158c050@mail.gmail.com> Hi! I'm trying the Doodle web SIP phone but for some reason I'm unable to register to my FreeSWITCH server. I've tried with other servers and it works ok. Did someone tried this web phone with FreeSWITCH? Any tips why it doesn't authenticate? Thank you! Federico Omoto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/cf7aa011/attachment.html From frank at carmickle.com Mon Nov 9 13:03:43 2009 From: frank at carmickle.com (Frank Carmickle) Date: Mon, 9 Nov 2009 16:03:43 -0500 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> Message-ID: <20091109210343.GH11697@base.carmickle.com> On Mon, Nov 09, Michael Collins wrote: > On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson > wrote: > > > I **think** that the following will match any three character strings > > from 1xx to 399 > > > > I want to exclude 100 though, can anyone help me with the required RegEx > > please ? > > > > > > ^([1-3][0-9][0-9])$ > > > > I could (I think) do > > > > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ You mean (^1[0-9][1-9]$|^[2-3]\d\d$) --FC From stevendt at primrosebank.net Mon Nov 9 13:05:40 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 21:05:40 -0000 Subject: [Freeswitch-users] Cordless VOIP Phones References: <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> Message-ID: <90CFC4A7AD444A9CA63B3FAF00744E5C@bp1.ad.bp.com> Hi Rupa, thanks for the tip. I've had a look for the A580-PI - as you say, quite inexpensive and probably worth taking a chance on one. regards Dave ----- Original Message ----- From: "Rupa Schomaker" To: Sent: Monday, November 09, 2009 8:23 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones I agree about the M3. I have the handset and it is not ergonomic at all. I also have a Siemens A580-IP. It does do G722 but has a few bugs related to G722 that I normally run it with G711 only. There is a quality difference between G722 and G711 when talking among the A580 handsets or the handset and my Polycom 450. Of the DECT handsets, I'd recommend the Siemens. It is also very inexpensive. Beware it has a 3 concurrent calls limit per base-station. On Mon, Nov 9, 2009 at 12:05 PM, Brian West wrote: > > On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote: > > Joao, > > thanks for the note. The Snom M3 is one of the ones that I was looking > at - > I would be interested in the "Pro's & Cons" ? > > RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN > > > Interesting about the HD, but do you notice the difference and find that > you're disappointed with the quality of their sounds ? > > Get an ATA with a Dect handset it works much better... the Snom M3 and the > Aastra are one in the same and they both do not live up to the quality or > usability requirements. > > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From stevendt at primrosebank.net Mon Nov 9 13:09:15 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 21:09:15 -0000 Subject: [Freeswitch-users] RegEx Help References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com><87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> <20091109210343.GH11697@base.carmickle.com> Message-ID: <9D87DD1FCD5A45518D76F9A70F001F96@bp1.ad.bp.com> Hi Frank Yup ! That's what I mean :-) thanks a lot, regards Dave ----- Original Message ----- From: "Frank Carmickle" To: Sent: Monday, November 09, 2009 9:03 PM Subject: Re: [Freeswitch-users] RegEx Help > On Mon, Nov 09, Michael Collins wrote: >> On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson >> wrote: >> >> > I **think** that the following will match any three character strings >> > from 1xx to 399 >> > >> > I want to exclude 100 though, can anyone help me with the required >> > RegEx >> > please ? >> > >> > >> > ^([1-3][0-9][0-9])$ >> > >> > I could (I think) do >> > >> > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ > > You mean > > (^1[0-9][1-9]$|^[2-3]\d\d$) > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Russell.Mosemann at cune.org Mon Nov 9 13:10:25 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Mon, 9 Nov 2009 21:10:25 -0000 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> Message-ID: <20091109211025.ACC7B2E3DD4@mail.cune.org> Dave Stevenson said: > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ Another possibility. ^(1(0[1-9]|[1-9]\d)|[2-3]\d{2}) -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From stevecrozz at gmail.com Mon Nov 9 13:15:56 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 9 Nov 2009 13:15:56 -0800 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <20091109210343.GH11697@base.carmickle.com> References: <4AF4AF73.8070804@tx.rr.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> <20091109210343.GH11697@base.carmickle.com> Message-ID: <11990ade0911091315l6db2eb04v4d1a472d2b9b8b10@mail.gmail.com> Dave, I think extensions are processed in order although I can't quickly find any documentation that says this, why don't you try it and see, it would take only a moment to find out for sure. --Stephen On Mon, Nov 9, 2009 at 1:03 PM, Frank Carmickle wrote: > On Mon, Nov 09, Michael Collins wrote: > > On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson > > wrote: > > > > > I **think** that the following will match any three character strings > > > from 1xx to 399 > > > > > > I want to exclude 100 though, can anyone help me with the required > RegEx > > > please ? > > > > > > > > > ^([1-3][0-9][0-9])$ > > > > > > I could (I think) do > > > > > > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ > > You mean > > (^1[0-9][1-9]$|^[2-3]\d\d$) > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/d85d1346/attachment.html From msc at freeswitch.org Mon Nov 9 13:15:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 13:15:56 -0800 Subject: [Freeswitch-users] Doddle Web SIP phone In-Reply-To: <8b4221f20911091150k7f3d01eem3c5eae845158c050@mail.gmail.com> References: <8b4221f20911091150k7f3d01eem3c5eae845158c050@mail.gmail.com> Message-ID: <87f2f3b90911091315o722ae1c4xdc6728e251f4f7b2@mail.gmail.com> On Mon, Nov 9, 2009 at 11:50 AM, Fede wrote: > Hi! > > I'm trying the Doodle web SIP phone but for some reason I'm unable to > register to my FreeSWITCH server. I've tried with other servers and it works > ok. > Did someone tried this web phone with FreeSWITCH? Any tips why it doesn't > authenticate? > Can you capture the debug log from the command line? It would also be good to have a SIP trace. More information on gathering info and putting it in pastebin can be found here: http://wiki.freeswitch.org/wiki/Reporting_Bugs Also, be sure that you are using the latest version of FreeSWITCH, preferably SVN trunk. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/dc67cecd/attachment.html From msc at freeswitch.org Mon Nov 9 13:20:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 13:20:59 -0800 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <20091109211025.ACC7B2E3DD4@mail.cune.org> References: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <20091109211025.ACC7B2E3DD4@mail.cune.org> Message-ID: <87f2f3b90911091320g79a6da8eo48b2b572322f8eb2@mail.gmail.com> On Mon, Nov 9, 2009 at 1:10 PM, wrote: > Dave Stevenson said: > > > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ > > Another possibility. > > ^(1(0[1-9]|[1-9]\d)|[2-3]\d{2}) > > Yep this is the one. I'm sorry I didn't read the OP correctly the first time. Skipping 100 and matching 101 is the tricky part, obviously. This regex should fit the bill. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/bd29c30b/attachment.html From mattdfong at gmail.com Mon Nov 9 13:41:01 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 10 Nov 2009 04:41:01 +0700 Subject: [Freeswitch-users] Doddle Web SIP phone In-Reply-To: <87f2f3b90911091315o722ae1c4xdc6728e251f4f7b2@mail.gmail.com> References: <8b4221f20911091150k7f3d01eem3c5eae845158c050@mail.gmail.com> <87f2f3b90911091315o722ae1c4xdc6728e251f4f7b2@mail.gmail.com> Message-ID: <4256bf830911091341w1bfb2dafx818dd4d4f18248ec@mail.gmail.com> I just tried the webphone with my freeswitch server and it worked fine, making a call to my echo test w/o any issues...so it's probably a configuration issue with freeswitch. --matt http://www.hellohunter.com On Tue, Nov 10, 2009 at 4:15 AM, Michael Collins wrote: > > > On Mon, Nov 9, 2009 at 11:50 AM, Fede wrote: > >> Hi! >> >> I'm trying the Doodle web SIP phone but for some reason I'm unable to >> register to my FreeSWITCH server. I've tried with other servers and it works >> ok. >> Did someone tried this web phone with FreeSWITCH? Any tips why it doesn't >> authenticate? >> > > Can you capture the debug log from the command line? It would also be good > to have a SIP trace. More information on gathering info and putting it in > pastebin can be found here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Also, be sure that you are using the latest version of FreeSWITCH, > preferably SVN trunk. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/ed0488ee/attachment.html From federico.omoto at gmail.com Mon Nov 9 13:46:30 2009 From: federico.omoto at gmail.com (Fede) Date: Mon, 9 Nov 2009 19:46:30 -0200 Subject: [Freeswitch-users] Doddle Web SIP phone In-Reply-To: <87f2f3b90911091315o722ae1c4xdc6728e251f4f7b2@mail.gmail.com> References: <8b4221f20911091150k7f3d01eem3c5eae845158c050@mail.gmail.com> <87f2f3b90911091315o722ae1c4xdc6728e251f4f7b2@mail.gmail.com> Message-ID: <8b4221f20911091346y6181f8b1o6e6c4d88ca81eb94@mail.gmail.com> Hi Michael! Thank you for your quicky answer. I'm using FreeSWITCH 1.0.5 pre5. The debug log from the command line plus the SIP trace are at: http://pastebin.freeswitch.org/11043 The Doddle web phone is at: http://www.doddlephone.com You can test this account at my FreeSWITCH server at: -server: 216.75.60.102 -username: doddle -password: doddle Thank you! Federico Omoto On Mon, Nov 9, 2009 at 7:15 PM, Michael Collins wrote: > > > On Mon, Nov 9, 2009 at 11:50 AM, Fede wrote: > >> Hi! >> >> I'm trying the Doodle web SIP phone but for some reason I'm unable to >> register to my FreeSWITCH server. I've tried with other servers and it works >> ok. >> Did someone tried this web phone with FreeSWITCH? Any tips why it doesn't >> authenticate? >> > > Can you capture the debug log from the command line? It would also be good > to have a SIP trace. More information on gathering info and putting it in > pastebin can be found here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Also, be sure that you are using the latest version of FreeSWITCH, > preferably SVN trunk. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/a8eeec25/attachment.html From JCasale at activenetwerx.com Mon Nov 9 14:46:18 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 9 Nov 2009 22:46:18 +0000 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: >The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ? Worst POS I have ever used, from a sound quality to ergonomics pov, tech support was as bad... I have Aastra 480i CT's which work well. jlc From stevendt at primrosebank.net Mon Nov 9 14:56:10 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 22:56:10 -0000 Subject: [Freeswitch-users] DIalplan logic References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com><11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> <6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com> Message-ID: <77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox set the timeout for the call Added these lines - but don't know why - they are in the default extension code ????? then go to voice mail on 100 giving The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/78c113b9/attachment-0001.html From anthony.minessale at gmail.com Mon Nov 9 15:01:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 17:01:13 -0600 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> asstra has one issue where if you look at them wrong they start telling the server that the media ip is 0.0.0.0 which we have never identified but they indeed seem to work better than snom m3 On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale wrote: > >The Snom M3 is one of the ones that I was looking at - I would be > interested in the "Pro's & Cons" ? > > Worst POS I have ever used, from a sound quality to ergonomics pov, tech > support was as bad... > > I have Aastra 480i CT's which work well. > > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/9fca3e43/attachment.html From stevendt at primrosebank.net Mon Nov 9 15:01:13 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 23:01:13 -0000 Subject: [Freeswitch-users] Cordless VOIP Phones References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: <908AF2F7F29D44CEA03C12A60C6CB298@bp1.ad.bp.com> Thanks - pretty unambiguous reply ! I won't go down that route then :-) ----- Original Message ----- From: "Joseph L. Casale" To: Sent: Monday, November 09, 2009 10:46 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones > >The Snom M3 is one of the ones that I was looking at - I would be > >interested in the "Pro's & Cons" ? > > Worst POS I have ever used, from a sound quality to ergonomics pov, tech > support was as bad... > > I have Aastra 480i CT's which work well. > > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Mon Nov 9 15:04:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 17:04:13 -0600 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <87f2f3b90911091320g79a6da8eo48b2b572322f8eb2@mail.gmail.com> References: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <20091109211025.ACC7B2E3DD4@mail.cune.org> <87f2f3b90911091320g79a6da8eo48b2b572322f8eb2@mail.gmail.com> Message-ID: <191c3a030911091504s675f8e79vafffea36130e20c5@mail.gmail.com> If the global var "auto_hunt" is "true" the xml dialplan will try to find an extension where the "name" param matches the destination number. This is not the default The default is to try them in order from top to bottom. On Mon, Nov 9, 2009 at 3:20 PM, Michael Collins wrote: > > > On Mon, Nov 9, 2009 at 1:10 PM, wrote: > >> Dave Stevenson said: >> >> > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ >> >> Another possibility. >> >> ^(1(0[1-9]|[1-9]\d)|[2-3]\d{2}) >> >> > Yep this is the one. I'm sorry I didn't read the OP correctly the first > time. Skipping 100 and matching 101 is the tricky part, obviously. This > regex should fit the bill. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/77405a75/attachment.html From jmesquita at freeswitch.org Mon Nov 9 15:16:37 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 9 Nov 2009 21:16:37 -0200 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> References: <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> Message-ID: Beat me with a dead cat all you want but I rather the snom m3 than the Siemens A580IP.... Siemens has very low volume which makes its call quality suck despite of being ergonomic and all... That gigaset application sucks and the base station is slow as hell... Maybe I have a bad unit? The snom m3 has its downsides, but all and all, I am happy with the phone if you consider its price tag here in South America where a Polycom can easily cost over 200USD the cheapest unit. Regards, JM On Mon, Nov 9, 2009 at 9:01 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > asstra has one issue where if you look at them wrong they start telling the > server that the media ip is 0.0.0.0 which we have never identified but they > indeed seem to work better than snom m3 > > > > On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale < > JCasale at activenetwerx.com> wrote: > >> >The Snom M3 is one of the ones that I was looking at - I would be >> interested in the "Pro's & Cons" ? >> >> Worst POS I have ever used, from a sound quality to ergonomics pov, tech >> support was as bad... >> >> I have Aastra 480i CT's which work well. >> >> jlc >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/88edc5fd/attachment.html From msc at freeswitch.org Mon Nov 9 15:26:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 15:26:25 -0800 Subject: [Freeswitch-users] DIalplan logic In-Reply-To: <77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> References: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> <6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com> <77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> Message-ID: <87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com> See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: > Hi Guys, > > OK, with the RegEx help that you gave me, I have separated out the > processing of extension 100 from 101 to 399 as I wanted. > > I have created a group (100) which contains a number of phones - 101 to 105 > at the moment. > > When the PSTN line rings, I want all the extensions in the group to ring - > that's the easy bit (I think - it's a copy of extension 2000 code) > > > > > > > > That's fine and the nominated phones all ring. > > I'm struggling to get it to do what I want when some doesn't pick up > though. > > All extensions ring as required, but their own dialplan entries (copies of > the 1001 to 1005 code in the default dialplan) don't answer the call. That's > fine, as you would not want every extension's voice mail to kick in. > > What I want to happen is for extension 100's voice mail to kick in after a > time delay. > > So, get the dialed exetension number so that I can point at the right > mailbox > > > set the timeout for the call > > > Added these lines - but don't know why - they are in the default extension > code ????? > > > > > then go to voice mail on 100 > > > > > giving > > > The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. > > > > > > > > > > I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" > > > > > The voicemail kicks in, and prompts are correct (although the extension > name is not spoken) but the wav file is saved in the 1001 directory not 100 > and neither extension 100 or 1001 think that have any voice mail messages. > > Can someone help please ? > > Where am i going wrong ? > > Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/3389c076/attachment-0001.html From srinivas.ksvreddy at gmail.com Mon Nov 9 15:36:14 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Mon, 9 Nov 2009 18:36:14 -0500 Subject: [Freeswitch-users] Request: Notify sip messages from Freeswitch to UserAgent Message-ID: Hi, >From Freeswitch there is continuously Request: Notify (Messages-waiting) requests are comming, i didnt subscribe from Freeswith and pjsip(ua). any body know how to stop those requests from Freeswitch. Thanks-- Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/6254aa14/attachment.html From stevendt at primrosebank.net Mon Nov 9 15:52:44 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 23:52:44 -0000 Subject: [Freeswitch-users] DIalplan logic References: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com><11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com><6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com><77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> <87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com> Message-ID: Michael, thanks a lot - it's fixed...... you spotted exactly what the problem was ! regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 11:26 PM Subject: Re: [Freeswitch-users] DIalplan logic See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox set the timeout for the call Added these lines - but don't know why - they are in the default extension code ????? then go to voice mail on 100 giving The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/61cbd198/attachment.html From brian at freeswitch.org Mon Nov 9 16:03:14 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Nov 2009 18:03:14 -0600 Subject: [Freeswitch-users] Request: Notify sip messages from Freeswitch to UserAgent In-Reply-To: References: Message-ID: <3EBF2A1F-122E-4116-B0AE-989C9268B1D5@freeswitch.org> gratuitous notifies are what they are called and I think their is a patch on jira with that functionality. I would have to dig thru jira to double check... I think Moc wrote the patch. /b On Nov 9, 2009, at 5:36 PM, srinivasula reddy wrote: > Hi, > > From Freeswitch there is continuously Request: Notify (Messages- > waiting) requests are comming, i didnt subscribe from Freeswith and > pjsip(ua). > any body know how to stop those requests from Freeswitch. > > Thanks-- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Nov 9 16:09:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 18:09:07 -0600 Subject: [Freeswitch-users] Request: Notify sip messages from Freeswitch to UserAgent In-Reply-To: References: Message-ID: <191c3a030911091609u4e809fe6r5858ec3d1f917adf@mail.gmail.com> Add that to your sofia profile. You must be new to SIP, you will soon learn that almost every SIP device just stupidly expects you to send this and never does it the correct way by subscribing to it which is why this option is the default. On Mon, Nov 9, 2009 at 5:36 PM, srinivasula reddy < srinivas.ksvreddy at gmail.com> wrote: > Hi, > > From Freeswitch there is continuously Request: Notify (Messages-waiting) > requests are comming, i didnt subscribe from Freeswith and pjsip(ua). > any body know how to stop those requests from Freeswitch. > > Thanks-- > Srinivasula Reddy K > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/031a5e96/attachment.html From stevendt at primrosebank.net Mon Nov 9 16:18:15 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 10 Nov 2009 00:18:15 -0000 Subject: [Freeswitch-users] DIalplan logic References: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com><11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com><6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com><77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com><87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com> Message-ID: <054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> Well, I thought it was fixed - it is more or less working, with one more stumbling block. I have just posted a dump to the pastebin - from Dave (stevendt) The voice mail works - but too well. If the call is answered by a someone at this end - everything is fine until the user hangs up, then the remote party gets the voicemail messages.# Is there something else wrong with the dialplan logic below ? regards Dave ----- Original Message ----- From: Dave Stevenson To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 11:52 PM Subject: Re: [Freeswitch-users] DIalplan logic Michael, thanks a lot - it's fixed...... you spotted exactly what the problem was ! regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 11:26 PM Subject: Re: [Freeswitch-users] DIalplan logic See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox set the timeout for the call Added these lines - but don't know why - they are in the default extension code ????? then go to voice mail on 100 giving The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/a4515cf4/attachment-0001.html From mike at jerris.com Mon Nov 9 16:19:10 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Nov 2009 19:19:10 -0500 Subject: [Freeswitch-users] Request: Notify sip messages from Freeswitch to UserAgent In-Reply-To: References: Message-ID: I have asked you before to please not cross post to both mailing lists. Please refrain from this in the future. Mike On Nov 9, 2009, at 6:36 PM, srinivasula reddy wrote: > Hi, > > From Freeswitch there is continuously Request: Notify (Messages- > waiting) requests are comming, i didnt subscribe from Freeswith and > pjsip(ua). > any body know how to stop those requests from Freeswitch. > > Thanks-- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Nov 9 16:29:24 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 18:29:24 -0600 Subject: [Freeswitch-users] DIalplan logic In-Reply-To: <054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> References: <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> <6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com> <77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> <87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com> <054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> Message-ID: <191c3a030911091629g584b4519nb8e555bbd38ff7b3@mail.gmail.com> You set both hangup_after_bridge and continue_on_fail after you already called bridge. Try setting it *before* Seems to be a running theme here that things will be parsed in a linear fashion that you may want to take note of. On Mon, Nov 9, 2009 at 6:18 PM, Dave Stevenson wrote: > Well, > > I thought it was fixed - it is more or less working, with one more > stumbling block. > > I have just posted a dump to the pastebin - from Dave (stevendt) > > The voice mail works - but too well. > > If the call is answered by a someone at this end - everything is fine until > the user hangs up, then the remote party gets the voicemail messages.# > > Is there something else wrong with the dialplan logic below ? > > regards > Dave > > > > ----- Original Message ----- > *From:* Dave Stevenson > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, November 09, 2009 11:52 PM > *Subject:* Re: [Freeswitch-users] DIalplan logic > > Michael, > > thanks a lot - it's fixed...... > > > you spotted exactly what the problem was ! > > > > > > > regards > Dave > > > > > ----- Original Message ----- > *From:* Michael Collins > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, November 09, 2009 11:26 PM > *Subject:* Re: [Freeswitch-users] DIalplan logic > > See comment inline > > On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: > >> Hi Guys, >> >> OK, with the RegEx help that you gave me, I have separated out the >> processing of extension 100 from 101 to 399 as I wanted. >> >> I have created a group (100) which contains a number of phones - 101 to >> 105 at the moment. >> >> When the PSTN line rings, I want all the extensions in the group to ring - >> that's the easy bit (I think - it's a copy of extension 2000 code) >> >> >> >> > >> >> >> That's fine and the nominated phones all ring. >> >> I'm struggling to get it to do what I want when some doesn't pick up >> though. >> >> All extensions ring as required, but their own dialplan entries (copies of >> the 1001 to 1005 code in the default dialplan) don't answer the call. That's >> fine, as you would not want every extension's voice mail to kick in. >> >> What I want to happen is for extension 100's voice mail to kick in after a >> time delay. >> >> So, get the dialed exetension number so that I can point at the right >> mailbox >> >> >> set the timeout for the call >> >> >> Added these lines - but don't know why - they are in the default extension >> code ????? >> >> >> >> >> then go to voice mail on 100 >> >> >> >> >> giving >> >> >> > > The following line needs to have 100 in parens like this: "^(100)$" because > that's how you get $1 to be populated. > >> >> >> >> > >> >> >> >> >> > > I think this might be a typo? Shouldn't this next line be ... data="default > ${domain_name} ${dialed_extension}" > >> >> >> >> >> The voicemail kicks in, and prompts are correct (although the extension >> name is not spoken) but the wav file is saved in the 1001 directory not 100 >> and neither extension 100 or 1001 think that have any voice mail messages. >> >> Can someone help please ? >> >> Where am i going wrong ? >> >> > > Make those changes, reloadxml, and then try again. Be sure to capture a > debug log if it doesn't work and put that log in pastebin.freeswitch.org. > -MC > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/5a7a0b28/attachment.html From msc at freeswitch.org Mon Nov 9 16:32:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 16:32:14 -0800 Subject: [Freeswitch-users] DIalplan logic In-Reply-To: <054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> References: <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> <6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com> <77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> <87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com> <054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> Message-ID: <87f2f3b90911091632v66c5678dg3c3120115bda63dc@mail.gmail.com> Oops, you've got some lines that are in the wrong place: Those lines need to come prior to the bridge call or they'll never be applied. :) -MC On Mon, Nov 9, 2009 at 4:18 PM, Dave Stevenson wrote: > Well, > > I thought it was fixed - it is more or less working, with one more > stumbling block. > > I have just posted a dump to the pastebin - from Dave (stevendt) > > The voice mail works - but too well. > > If the call is answered by a someone at this end - everything is fine until > the user hangs up, then the remote party gets the voicemail messages.# > > Is there something else wrong with the dialplan logic below ? > > regards > Dave > > > > ----- Original Message ----- > *From:* Dave Stevenson > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, November 09, 2009 11:52 PM > *Subject:* Re: [Freeswitch-users] DIalplan logic > > Michael, > > thanks a lot - it's fixed...... > > > you spotted exactly what the problem was ! > > > > > > > regards > Dave > > > > > ----- Original Message ----- > *From:* Michael Collins > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, November 09, 2009 11:26 PM > *Subject:* Re: [Freeswitch-users] DIalplan logic > > See comment inline > > On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: > >> Hi Guys, >> >> OK, with the RegEx help that you gave me, I have separated out the >> processing of extension 100 from 101 to 399 as I wanted. >> >> I have created a group (100) which contains a number of phones - 101 to >> 105 at the moment. >> >> When the PSTN line rings, I want all the extensions in the group to ring - >> that's the easy bit (I think - it's a copy of extension 2000 code) >> >> >> >> > >> >> >> That's fine and the nominated phones all ring. >> >> I'm struggling to get it to do what I want when some doesn't pick up >> though. >> >> All extensions ring as required, but their own dialplan entries (copies of >> the 1001 to 1005 code in the default dialplan) don't answer the call. That's >> fine, as you would not want every extension's voice mail to kick in. >> >> What I want to happen is for extension 100's voice mail to kick in after a >> time delay. >> >> So, get the dialed exetension number so that I can point at the right >> mailbox >> >> >> set the timeout for the call >> >> >> Added these lines - but don't know why - they are in the default extension >> code ????? >> >> >> >> >> then go to voice mail on 100 >> >> >> >> >> giving >> >> >> > > The following line needs to have 100 in parens like this: "^(100)$" because > that's how you get $1 to be populated. > >> >> >> >> > >> >> >> >> >> > > I think this might be a typo? Shouldn't this next line be ... data="default > ${domain_name} ${dialed_extension}" > >> >> >> >> >> The voicemail kicks in, and prompts are correct (although the extension >> name is not spoken) but the wav file is saved in the 1001 directory not 100 >> and neither extension 100 or 1001 think that have any voice mail messages. >> >> Can someone help please ? >> >> Where am i going wrong ? >> >> > > Make those changes, reloadxml, and then try again. Be sure to capture a > debug log if it doesn't work and put that log in pastebin.freeswitch.org. > -MC > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/513dd73c/attachment-0001.html From stevendt at primrosebank.net Mon Nov 9 16:42:14 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 10 Nov 2009 00:42:14 -0000 Subject: [Freeswitch-users] DIalplan logic References: <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com><11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com><6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com><77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com><87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com><054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> <191c3a030911091629g584b4519nb8e555bbd38ff7b3@mail.gmail.com> Message-ID: <160223E6251F4D24B6D9A50362204BA3@bp1.ad.bp.com> Thanks Anthony that did the trick ! Excuse my ignorance - this is all new to me . . . It would help if I knew what I was doing, as I commented below, I copied the from the code for the default extensions (in the wrong order though obviously), without understanding what they were there for ! I've just stumbled across the Wiki page http://wiki.freeswitch.org/wiki/Extension_Status_Example - hopefully, I understand now regards Dave ----- Original Message ----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 10, 2009 12:29 AM Subject: Re: [Freeswitch-users] DIalplan logic You set both hangup_after_bridge and continue_on_fail after you already called bridge. Try setting it *before* Seems to be a running theme here that things will be parsed in a linear fashion that you may want to take note of. On Mon, Nov 9, 2009 at 6:18 PM, Dave Stevenson wrote: Well, I thought it was fixed - it is more or less working, with one more stumbling block. I have just posted a dump to the pastebin - from Dave (stevendt) The voice mail works - but too well. If the call is answered by a someone at this end - everything is fine until the user hangs up, then the remote party gets the voicemail messages.# Is there something else wrong with the dialplan logic below ? regards Dave ----- Original Message ----- From: Dave Stevenson To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 11:52 PM Subject: Re: [Freeswitch-users] DIalplan logic Michael, thanks a lot - it's fixed...... you spotted exactly what the problem was ! regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 11:26 PM Subject: Re: [Freeswitch-users] DIalplan logic See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox set the timeout for the call Added these lines - but don't know why - they are in the default extension code ????? then go to voice mail on 100 giving The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC ------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/63e062d0/attachment-0001.html From gshfreesw at gmail.com Mon Nov 9 18:54:15 2009 From: gshfreesw at gmail.com (Shameem Shiek) Date: Mon, 9 Nov 2009 21:54:15 -0500 Subject: [Freeswitch-users] Mod Voicemail, Is it just for registered users? Message-ID: <5070fcbd0911091854t3d359deam6170d27fa426d9ab@mail.gmail.com> Dear Freeswitch users, I am building an app where the extensions map to external callers and there are no registered users. For example, the extension 1001 would map to an external number. In that case, does it make sense to use the Mod voicemail or should I build a voicemail solution using dialplan/commands ? I need a voicemail solution to email voice messages. Thanks in advance for your input. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/bca3fdeb/attachment.html From mike at jerris.com Mon Nov 9 19:07:37 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Nov 2009 22:07:37 -0500 Subject: [Freeswitch-users] Mod Voicemail, Is it just for registered users? In-Reply-To: <5070fcbd0911091854t3d359deam6170d27fa426d9ab@mail.gmail.com> References: <5070fcbd0911091854t3d359deam6170d27fa426d9ab@mail.gmail.com> Message-ID: <06537308-F9FE-4A16-8F0F-E1491AA48C40@jerris.com> registration has nothing at all to do with mod_voicemail. It should work fine. On Nov 9, 2009, at 9:54 PM, Shameem Shiek wrote: > Dear Freeswitch users, > > I am building an app where the extensions map to external callers > and there are no registered users. For example, the extension 1001 > would map to an external number. In that case, does it make sense to > use the Mod voicemail or should I build a voicemail solution using > dialplan/commands ? I need a voicemail solution to email voice > messages. > From ujjval at simplesignal.com Mon Nov 9 21:08:56 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Mon, 9 Nov 2009 21:08:56 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> , <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175C650F1D@EXMBXCLUS01.citservers.local> OK, I may have solved this mystery, if I use application=answer and answer the call before the IVR which then flows into the Conference app, DTMF works from the AT&T phone.. So, if you face issues with Conferencing/IVR, answer the call before you invoke those apps... Problem I have now is that a Polycom old phone like 501s are not doing DTMF 2833 to the Freeswitch server...has anyone seen this..Call is not going thru PSTN...its IP to IP Polycom 501 through our SBC to the Freeswitch Server. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, November 02, 2009 9:08 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator you know I have heard this before... It seems to ONLY be AT&T /b On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote: > Yes, I think I did. However here is what furthur testing revelas. If > I dial in from AT&T cell phone, I do not see any DTMF using Don's > IVR.xml.conf to call my conf app. But when I dial the same number > using a Verizon Cell, it works. > > When I dial a number that is provisioned to call the Conf App > directly from the public.xml dialplan...it works even with the same > AT&T cell phone... > > Strange behaviour _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From paul.thirumalai at gmail.com Mon Nov 9 21:37:41 2009 From: paul.thirumalai at gmail.com (Paul Thirumalai) Date: Mon, 9 Nov 2009 21:37:41 -0800 Subject: [Freeswitch-users] Configuring freeswitch with voicepulse Message-ID: <900c9adf0911092137vf45ec94ie7473d2c08e5ae12@mail.gmail.com> Hello All I am trying to configure freeswitch so that it sends outgoing calls to the PSTN through voicepulse My configuration is as follows. I created a file $PREFIX/conf/sip_profiles/external/voicepulse.xml I also have a dial plan defined as follows When I dial an external number using extension 1000 I get the following message on the CLI ] freeswitch at ubuntu> 2009-11-10 00:35:44.365614 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1000 at 74.207.249.79[e4301180-cdba-11de-a864-8927fe94a9f0] 2009-11-10 00:35:44.366623 [INFO] mod_dialplan_xml.c:315 Processing Paul->5555555555 in context default 2009-11-10 00:35:44.368645 [NOTICE] switch_channel.c:602 New Channel sofia/external/5555555555 [e43092f4-cdba-11de-a864-8927fe94a9f0] 2009-11-10 00:35:47.59221 [NOTICE] sofia_glue.c:2698 Pre-Answer sofia/external/5555555555! 2009-11-10 00:35:47.59221 [INFO] switch_ivr_originate.c:2017 Sending early media 2009-11-10 00:35:47.60524 [INFO] mod_sofia.c:1506 Ring SDP: v=0 o=FreeSWITCH 1257800805 1257800806 IN IP4 74.207.249.79 s=FreeSWITCH c=IN IP4 74.207.249.79 t=0 0 m=audio 30542 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-11-10 00:35:47.60524 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1000 at 74.207.249.79! 2009-11-10 00:35:51.449542 [NOTICE] sofia.c:3849 Hangup sofia/external/5555555555 [CS_EXCHANGE_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2009-11-10 00:35:51.452539 [NOTICE] switch_ivr_bridge.c:419 Hangup sofia/internal/1000 at 74.207.249.79 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/internal/1000 at 74.207.249.79) Ended 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1000 at 74.207.249.79 [CS_DESTROY] 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/external/5555555555) Ended 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/5555555555 [CS_DESTROY] I am really new to VOIP and having a hard time with this. I am really not sure how to proceed. Any help would be really appreciated. Thanks Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/b37a55a1/attachment.html From jason at jasonjgw.net Mon Nov 9 22:06:46 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 10 Nov 2009 17:06:46 +1100 Subject: [Freeswitch-users] Configuring freeswitch with voicepulse In-Reply-To: <900c9adf0911092137vf45ec94ie7473d2c08e5ae12@mail.gmail.com> References: <900c9adf0911092137vf45ec94ie7473d2c08e5ae12@mail.gmail.com> Message-ID: <20091110060646.GA24954@jdc.jasonjgw.net> Paul Thirumalai wrote: > I am really new to VOIP and having a hard time with this. I am really not > sure how to proceed. Any help would be really appreciated. First, turn on debug logging (in fs_cli, it's /log debug) to obtain more information. The proxy variables in your configuration could be complicating the situation unnecessarily - try removing them and specifying only the realm. I don't think you want two proxy variables here. If you're just new to FreeSWITCH, leave the debug logging level on and read the logs in /opt/freeswitch/log/freeswitch.log to track down problems. From codecomplete at free.fr Tue Nov 10 01:45:01 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 10 Nov 2009 01:45:01 -0800 (PST) Subject: [Freeswitch-users] Displaying caller ID on LED? Message-ID: <26280730.post@talk.nabble.com> Hello I was wondering if someone had succesfully configured FS to display caller ID on a LED like this? http://usb.brando.com/prod_detail.php?prod_id=00575 That would be a nice alternative to displaying CID information on the user's PC screen when users need to see who's calling where they're not in front of their computer (doctors, auto mechanics, etc.) Thank you. -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280730.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Nov 10 02:06:58 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 10 Nov 2009 02:06:58 -0800 (PST) Subject: [Freeswitch-users] Displaying caller ID on LED? In-Reply-To: <26280730.post@talk.nabble.com> References: <26280730.post@talk.nabble.com> Message-ID: <26280912.post@talk.nabble.com> ... or alternatively, on one of those USB digital picture frames? www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B000087BHC -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From lakindia89 at gmail.com Tue Nov 10 02:51:14 2009 From: lakindia89 at gmail.com (lakshmanan) Date: Tue, 10 Nov 2009 02:51:14 -0800 (PST) Subject: [Freeswitch-users] Flushing the Event buffer in Perl Event Socket In-Reply-To: <191c3a030910300650w7b80568eu4c41c805b9372acc@mail.gmail.com> References: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com> <191c3a030910300650w7b80568eu4c41c805b9372acc@mail.gmail.com> Message-ID: <26281493.post@talk.nabble.com> Hi anthony, I was in a need of flushing the events buffer without reading it.I've done the following ESL(Async) program to flush the events. First I register for events. I answered the call and playback some message. Now the events would have been queued. I, then send "noevents". After sending that, I again register for events, and when I receive the events, I've not got the old events. I got only new events. But I don't know whether it is exactly a way to flush the events or not. I just need your suggestions or your thoughts on this. Here is the script: use lib "/usr/local/freeswitch/scripts/esl"; require ESL; use IO::Socket::INET; use Data::Dumper; my $ip = "192.168..0.0"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; my $con; for(;;) { my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { close($new_sock); next; } my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); print "Host name is $host\n"; $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s to %s\n", $uuid, $info->getHeader("caller-caller-id-number"), $info->getHeader("caller-destination-number"); $con->filter("Unique-Id", $uuid); $con->events("plain", "all"); $con->execute("answer"); $con->setEventLock("true"); $con->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); $con->send("noevents"); sleep(5); $con->events("plain", "all"); while(my $e = $con->recvEvent()) { print $e->serialize(); } } Anthony Minessale-2 wrote: > > read them in a timed loop of some small number of MS until you get a > timeout > meaning you have flushed them all. > > > On Fri, Oct 30, 2009 at 1:57 AM, velusamy velu > wrote: > >> Dear All, >> I receiving the events in while loop by using recvEventTimed method >> in ESL.pm. I have to flush that Event buffer after some particular time. >> How >> can I do it? >> >> Thanks, >> Velusamy >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Flushing-the-Event-buffer-in-Perl-Event-Socket-tp26125824p26281493.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From info at daccii.it Tue Nov 10 05:00:23 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Tue, 10 Nov 2009 14:00:23 +0100 Subject: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it Message-ID: <4AF963E7.2080104@daccii.it> Hi, yesterday i started to fix pronuce in mod_say_it for numbers, dates and times. I needed to add some sound files because these was necessary for a correct italian pronunce. I've patched these three functions: - play_group - it_say_time - it_say_general_count I've diff it against revision 15396 (i've updated freeswitch tree yesterday morning) Can you take a look to the patch? # Modification to play_group function In italian we pronunce 123 as "cento venti tre" and not "uno cento venti tre" so, if a is 1 just doesn't play the digit # Modification to it_say_time Our long date format is something like WDAY_NAME, WDAY_NUMBER MONTH_NAME YEAR so i converted the date pronunce to this. I've dropped am/pm logic, because we have 24h standard, and minutes related logic because, we don't have it. # Modification to it_say_general_count I rewrote number to string conversion to make it more readable (using just two math operations, a module and a division) and to drop the 999 milions limit (1*)(however more code should be changed to fully drop this limit). Changes are mainly related to millions and thousands pronunce: in italian, if you need to say 1 milion you doesn't say "uno milione" but "un milione" but to say 3 millions you say "tre milioni", while for thousand you doesn't pronunce "un" at all. 1*: i've noticed a little bug in xx_say_money in mod_say_xx ... it get up to 12 digits but in xx_say_general_count manage up to 9 digits, so the first three digits wouldn't get never pronunced Thank you -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: mod_say_it.c.fix-pronunce.patch Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/e68393dc/attachment.pl -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/e68393dc/attachment.vcf From brian at freeswitch.org Tue Nov 10 06:10:16 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 08:10:16 -0600 Subject: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it In-Reply-To: <4AF963E7.2080104@daccii.it> References: <4AF963E7.2080104@daccii.it> Message-ID: <5BB9DB8A-8309-463C-B06F-BDA2A2EDF97E@freeswitch.org> Can you post your patch to jira.freeswitch.org please. /b On Nov 10, 2009, at 7:00 AM, Albano Daniele Salvatore - Lavoro wrote: > Hi, > > yesterday i started to fix pronuce in mod_say_it for numbers, dates > and times. I needed to add some sound files because these was > necessary for a correct italian pronunce. > > I've patched these three functions: > - play_group > - it_say_time > - it_say_general_count > > I've diff it against revision 15396 (i've updated freeswitch tree > yesterday morning) > > Can you take a look to the patch? > > > > # Modification to play_group function > > In italian we pronunce 123 as "cento venti tre" and not "uno cento > venti tre" so, if a is 1 just doesn't play the digit > > > > # Modification to it_say_time > > Our long date format is something like > > WDAY_NAME, WDAY_NUMBER MONTH_NAME YEAR > > so i converted the date pronunce to this. > > I've dropped am/pm logic, because we have 24h standard, and minutes > related logic because, we don't have it. > > > > # Modification to it_say_general_count > > I rewrote number to string conversion to make it more readable > (using just two math operations, a module and a division) and to > drop the 999 milions limit (1*)(however more code should be changed > to fully drop this limit). > > Changes are mainly related to millions and thousands pronunce: in > italian, if you need to say 1 milion you doesn't say "uno milione" > but "un milione" but to say 3 millions you say "tre milioni", while > for thousand you doesn't pronunce "un" at all. > > > 1*: i've noticed a little bug in xx_say_money in mod_say_xx ... it > get up to 12 digits but in xx_say_general_count manage up to 9 > digits, so the first three digits wouldn't get never pronunced > > Thank you > > Index: src/mod/say/mod_say_it/mod_say_it.c > =================================================================== > --- src/mod/say/mod_say_it/mod_say_it.c (revisione 15396) > +++ src/mod/say/mod_say_it/mod_say_it.c (copia locale) > @@ -95,7 +95,9 @@ > { > > if (a) { > - say_file("digits/%d.wav", a); > + if (a != 1) { > + say_file("digits/%d.wav", a); > + } > say_file("digits/hundred.wav"); > } > > @@ -170,7 +172,7 @@ > char *tosay, switch_say_type_t type, switch_say_method_t > method, switch_input_args_t *args) > { > int in; > - int x = 0; > + int places_count = 0; > int places[9] = { 0 }; > char sbuf[13] = ""; > switch_status_t status; > @@ -179,26 +181,64 @@ > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Parse > Error!\n"); > return SWITCH_STATUS_GENERR; > } > - > + > + // Get in > in = atoi(tosay); > + > + // Check if number too big > + if (in > 999999999) { > + // Fail > + return SWITCH_STATUS_FALSE; > + } > > + // Check if number isin't zero > if (in != 0) { > - for (x = 8; x >= 0; x--) { > - int num = (int) pow(10, x); > - if ((places[(uint32_t) x] = in / num)) { > - in -= places[(uint32_t) x] * num; > - } > - } > - > + > + // Init x to 0 > + places_count = 0; > + > + // Loop until in is greater than zero > + do { > + // Get last digit > + places[places_count] = in % 10; > + > + // Drop last digit > + in = in / 10; > + } > + while(in > 0 && ++places_count > 0 /** fake check to put in > while */); > + > switch (method) { > case SSM_COUNTED: > case SSM_PRONOUNCED: > - if ((status = play_group(SSM_PRONOUNCED, places[8], places[7], > places[6], "digits/million.wav", session, args)) != > SWITCH_STATUS_SUCCESS) { > - return status; > - } > - if ((status = play_group(SSM_PRONOUNCED, places[5], places[4], > places[3], "digits/thousand.wav", session, args)) != > SWITCH_STATUS_SUCCESS) { > - return status; > - } > + > + // Check for milions > + if (places_count > 5) { > + // Check if the millions digit is one (digit 6 = 1, > digit 7 and 8 = 0) > + if (places[6] == 1 && places[7] == 0 && places[8] > == 0) { > + say_file("digits/un.wav"); > + say_file("digits/million.wav"); > + } else { > + // Play millions group (digits/million.wav > should be digits/millions.wav) > + if ((status = play_group(SSM_PRONOUNCED, places > [8], places[7], places[6], "digits/million.wav", session, args)) != > SWITCH_STATUS_SUCCESS) { > + return status; > + } > + } > + > + } > + > + // Check for thousands > + if (places_count > 2) { > + if (places[3] == 1 && places[4] == 0 && places[5] > == 0) { > + say_file("digits/thousand.wav"); > + } else { > + // Play thousand group > + if ((status = play_group(SSM_PRONOUNCED, places > [5], places[4], places[3], "digits/thousands.wav", session, args)) ! > = SWITCH_STATUS_SUCCESS) { > + return status; > + } > + } > + } > + > + // Play last group > if ((status = play_group(method, places[2], places[1], places[0], > NULL, session, args)) != SWITCH_STATUS_SUCCESS) { > return status; > } > @@ -370,36 +410,19 @@ > > if (say_date) { > say_file("time/day-%d.wav", tm.tm_wday); > + say_num(tm.tm_mday, SSM_PRONOUNCED); > say_file("time/mon-%d.wav", tm.tm_mon); > - say_num(tm.tm_mday, SSM_COUNTED); > say_num(tm.tm_year + 1900, SSM_PRONOUNCED); > } > > if (say_time) { > - int32_t hour = tm.tm_hour, pm = 0; > + say_file("time/hours.wav"); > + say_num(tm.tm_hour, SSM_PRONOUNCED); > > - if (hour > 12) { > - hour -= 12; > - pm = 1; > - } else if (hour == 12) { > - pm = 1; > - } else if (hour == 0) { > - hour = 12; > - pm = 0; > - } > - > - say_num(hour, SSM_PRONOUNCED); > - > - if (tm.tm_min > 9) { > + if (tm.tm_min) { > + say_file("time/and.wav"); > say_num(tm.tm_min, SSM_PRONOUNCED); > - } else if (tm.tm_min) { > - say_file("time/oh.wav"); > - say_num(tm.tm_min, SSM_PRONOUNCED); > - } else { > - say_file("time/oclock.wav"); > } > - > - say_file("time/%s.wav", pm ? "p-m" : "a-m"); > } > > return SWITCH_STATUS_SUCCESS; > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Tue Nov 10 06:11:11 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 08:11:11 -0600 Subject: [Freeswitch-users] Flushing the Event buffer in Perl Event Socket In-Reply-To: <26281493.post@talk.nabble.com> References: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com> <191c3a030910300650w7b80568eu4c41c805b9372acc@mail.gmail.com> <26281493.post@talk.nabble.com> Message-ID: <2CD889D1-934A-47CE-A938-1EB9D4325DD2@freeswitch.org> $| = 1; I think that is what you're lookin for. /b On Nov 10, 2009, at 4:51 AM, lakshmanan wrote: > I was in a need of flushing the events buffer without reading > it.I've done > the following ESL(Async) program to flush the events. From codecomplete at free.fr Tue Nov 10 06:13:17 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 10 Nov 2009 06:13:17 -0800 (PST) Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> Message-ID: <26284109.post@talk.nabble.com> Are non-English sound files available in the SVN version of the code? I just tried installing the French sound files, but got an error: Unknown target cd-sounds-fr-install Unknown target cd-moh-fr-install make[1]: *** [cd-sounds-fr-install] Error 1 make: *** [cd-sounds-fr-install] Error 2 make[1]: *** [cd-moh-fr-install] Error 1 make: *** [cd-moh-fr-install] Error 2 [1]+ Exit 2 make cd-sounds-fr-install -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26284109.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From info at daccii.it Tue Nov 10 06:38:33 2009 From: info at daccii.it (Daniele Salvatore Albano) Date: Tue, 10 Nov 2009 15:38:33 +0100 Subject: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it In-Reply-To: <5BB9DB8A-8309-463C-B06F-BDA2A2EDF97E@freeswitch.org> References: <4AF963E7.2080104@daccii.it> <5BB9DB8A-8309-463C-B06F-BDA2A2EDF97E@freeswitch.org> Message-ID: <4AF97AE9.8090103@daccii.it> Hi, patch posted to http://jira.freeswitch.org/browse/MODAPP-362 Best Regards, Daniele Brian West ha scritto: > Can you post your patch to jira.freeswitch.org please. > > /b -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 307 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/938fd236/attachment.vcf From rupa at rupa.com Tue Nov 10 06:56:23 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 10 Nov 2009 06:56:23 -0800 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> Message-ID: 2009/11/9 Jo?o Mesquita : > Beat me with a dead cat all you want but I rather the snom m3 than the > Siemens A580IP.... Siemens has very low volume which makes its call quality > suck despite of being ergonomic and all... Did you flip hte option in the base station that tells it to make the audio louder? > That gigaset application sucks and the base station is slow as hell... Maybe > I have a bad unit? I didn't play with any of the gigaset specific stuff, I've disabled any screen savers. Maybe if I had the more "fancy" handsets the apps would be more useful, but when using the base handset they are not. Oh, and the weather is in C rather than F -- good for the rest of the world but not for us in the US. The base station is very slow with firefox but when I use chrome isn't so bad. Dunno if it was a combination of extensions or what. Oh, and it looks like a new firmware came out for the Siemens today. Wonder what it fixes (and breaks). Hmm.. wonder where I can find a list of whats new. > The snom m3 has its downsides, but all and all, I am happy with the phone if > you consider its price tag here in South America where a Polycom can easily > cost over 200USD the cheapest unit. > > Regards, > > JM > > On Mon, Nov 9, 2009 at 9:01 PM, Anthony Minessale > wrote: >> >> asstra has one issue where if you look at them wrong they start telling >> the server that the media ip is 0.0.0.0 which we have never identified but >> they indeed seem to work better than snom m3 >> >> >> On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale >> wrote: >>> >>> >The Snom M3 is one of the ones that I was looking at - I would be >>> > interested in the "Pro's & Cons" ? >>> >>> Worst POS I have ever used, from a sound quality to ergonomics pov, tech >>> support was as bad... >>> >>> I have Aastra 480i CT's which work well. >>> >>> jlc >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From brian at freeswitch.org Tue Nov 10 07:03:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 09:03:58 -0600 Subject: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it In-Reply-To: <4AF97AE9.8090103@daccii.it> References: <4AF963E7.2080104@daccii.it> <5BB9DB8A-8309-463C-B06F-BDA2A2EDF97E@freeswitch.org> <4AF97AE9.8090103@daccii.it> Message-ID: Patch applied. // comments aren't allowed in .c files in our tree we try hard to weed them out... anyway its committed now. And thanks for your contribution. /b On Nov 10, 2009, at 8:38 AM, Daniele Salvatore Albano wrote: > Hi, > > patch posted to http://jira.freeswitch.org/browse/MODAPP-362 > > > Best Regards, > Daniele > > Brian West ha scritto: >> Can you post your patch to jira.freeswitch.org please. >> >> /b > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From piotr_zurek at biprotech.com Tue Nov 10 07:01:17 2009 From: piotr_zurek at biprotech.com (=?UTF-8?B?UGlvdHIgxbt1cmVr?=) Date: Tue, 10 Nov 2009 16:01:17 +0100 Subject: [Freeswitch-users] How to pick up someone's phone remotely. Message-ID: <4AF9803D.9050806@biprotech.com> Hello. Thank You developers for Freeswitch. I have installed it lately and it's working quite nicely, but I have one problem: I need to mimic behavior of my current analogue PBX installation using Freeswitch. This is the scenario: In the office with a few desks (extensions 1000-1010) and only one person behind one of desks (whatever extension - in example 1000). 1. There's incoming call on _one_ of extensions 1001-1010 2. The person on extension 1000 wants to answer this call on his phone so dials #37 and this call is redirected to his phone. That's how it works on my office on analogue PBX system. Anyone can answer a call from any other phone as long as it hasn't been answered already. I tried to use the intercept action (with global example in default config) but it's not what I need because it intercepts the call even if it's already answered. I need to intercept all but only unanswered calls. I tried to use Redirect but it does not work on other's extensions call's (or does it?). Please help. Peter ?urek -------------- next part -------------- A non-text attachment was scrubbed... Name: piotr_zurek.vcf Type: text/x-vcard Size: 414 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/76bc8e42/attachment.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3678 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/76bc8e42/attachment.bin From rupa at rupa.com Tue Nov 10 07:09:18 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 10 Nov 2009 07:09:18 -0800 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> Message-ID: On Tue, Nov 10, 2009 at 6:56 AM, Rupa Schomaker wrote: > Oh, and it looks like a new firmware came out for the Siemens today. > Wonder what it fixes (and breaks). ?Hmm.. wonder where I can find a > list of whats new. Well, it seems to totally break g722. I haven't had a chance to narrow it down further, but beware if g722 is important don't update... -- -Rupa From brian at freeswitch.org Tue Nov 10 07:16:08 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 09:16:08 -0600 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <4AF9803D.9050806@biprotech.com> References: <4AF9803D.9050806@biprotech.com> Message-ID: <7B15EB87-5EDD-4234-8512-B1536B25DBEA@freeswitch.org> Please see the global-intercept example in the default config. /b On Nov 10, 2009, at 9:01 AM, Piotr ?urek wrote: > Hello. > > Thank You developers for Freeswitch. > I have installed it lately and it's working quite nicely, but I have > one problem: > > I need to mimic behavior of my current analogue PBX installation > using Freeswitch. > > This is the scenario: > In the office with a few desks (extensions 1000-1010) and only one > person behind one of desks (whatever extension - in example 1000). > 1. There's incoming call on _one_ of extensions 1001-1010 > 2. The person on extension 1000 wants to answer this call on his > phone so dials #37 and this call is redirected to his phone. > > That's how it works on my office on analogue PBX system. Anyone can > answer a call from any other phone as long as it hasn't been > answered already. > > I tried to use the intercept action (with global example in default > config) but it's not what I need because it intercepts the call even > if it's already answered. I need to intercept all but only > unanswered calls. I tried to use Redirect but it does not work on > other's extensions call's (or does it?). > > Please help. > Peter ?urek > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From markmorreny at gmail.com Tue Nov 10 07:29:37 2009 From: markmorreny at gmail.com (mark morreny) Date: Tue, 10 Nov 2009 23:29:37 +0800 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20091109192904.GI9418@hijacked.us> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> <20091109192904.GI9418@hijacked.us> Message-ID: <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> Hi Thanks for the tips. May I ask how to split the file from hadoop to the shell? Is it like copying the file to certain dir? I can't find any mod_shell_stream related info from the wiki. Does anyone know how to use it? thx, mark On Tue, Nov 10, 2009 at 3:29 AM, Andrew Thompson wrote: > On Mon, Nov 09, 2009 at 08:59:54PM +0800, mark morreny wrote: > > Hi, > > > > Does anyone know how to playback based on files from hadoop storage. > > > > There is a libhdcp, and java api. Is there anyway to put together a > sample > > middle piece to move files from hadoop to freeswitch using memory space, > so > > there is no disk I/O? > > > > Any feedback or suggestion will be greatly appreciated. > > > > mod_shell_stream might work, if you can just spit out the raw audio to > the shell. Or write another stream module that works with libhdcp. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/3290a455/attachment.html From andrew at hijacked.us Tue Nov 10 07:56:33 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 10 Nov 2009 10:56:33 -0500 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> <20091109192904.GI9418@hijacked.us> <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> Message-ID: <20091110155633.GA194@hijacked.us> On Tue, Nov 10, 2009 at 11:29:37PM +0800, mark morreny wrote: > Hi > > Thanks for the tips. May I ask how to split the file from hadoop to the > shell? Is it like copying the file to certain dir? > > I can't find any mod_shell_stream related info from the wiki. Does anyone > know how to use it? > mod_shell_stream is undocumented, but from reading the code I gather it works like this: Module calls fork() and in the child process it runs an arbitrary shell command (specified in its config file?). The parent process then reads raw audio data from the child process and uses it as an audio source. So basicially you could write the shell command in anything, so long as it outputs raw audio to FS. Or maybe I read the code wrong when I skimmed over it. If you do get it working, please contribute some documentation to the wiki. Andrew From oseslija at gmail.com Tue Nov 10 08:06:22 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Tue, 10 Nov 2009 17:06:22 +0100 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <4AF9803D.9050806@biprotech.com> References: <4AF9803D.9050806@biprotech.com> Message-ID: <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> Add the following: . after in local extensions default example, or change it globally previously than this extension. You can join us on IRC if you can any more questions (sekil). Regards, Ognjen On Tue, Nov 10, 2009 at 4:01 PM, Piotr ?urek wrote: > Hello. > > Thank You developers for Freeswitch. > I have installed it lately and it's working quite nicely, but I have one > problem: > > I need to mimic behavior of my current analogue PBX installation using > Freeswitch. > > This is the scenario: > In the office with a few desks (extensions 1000-1010) and only one person > behind one of desks (whatever extension - in example 1000). > 1. There's incoming call on _one_ of extensions 1001-1010 > 2. The person on extension 1000 wants to answer this call on his phone so > dials #37 and this call is redirected to his phone. > > That's how it works on my office on analogue PBX system. Anyone can answer > a call from any other phone as long as it hasn't been answered already. > > I tried to use the intercept action (with global example in default config) > but it's not what I need because it intercepts the call even if it's already > answered. I need to intercept all but only unanswered calls. I tried to use > Redirect but it does not work on other's extensions call's (or does it?). > > Please help. > Peter ?urek > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/28693819/attachment.html From stevendt at primrosebank.net Tue Nov 10 08:24:02 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 10 Nov 2009 16:24:02 -0000 Subject: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) Message-ID: <9E5323D6B69B489384D2E89358CC5EC5@bp1.ad.bp.com> I'm close to getting my SPA3102 working - he says hopefully .. . . . Making and receiving calls seems to be OK, but the SAP3102 doesn't seem to want to let go of the phone line once it's got it. Example I can receive a call, nobody answers and it goes to voicemail - working so far. FreeSwitch processes the normal VoiceMail system playing the prompts and recording the call. At the end, it says "Goodbye" and hear a "click" (from the remote end) and I see the console message that 2009-11-10 15:52:52.625000 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/1000 at 192.168.1.181 [CS_EXECUTE] [NORMAL_CLEARING] 2009-11-10 15:52:52.625000 [NOTICE] switch_core_session.c:1086 Session 362 (sofia/internal/1000 at 192.168.1.181) Ended 2009-11-10 15:52:52.625000 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1000 at 192.168.1.181 [CS_DESTROY] The above messages would suggest to me that FreeSwitch is doing its stuff right, but I have posted a dump in the pastebin just in case. The SPA3102 does not want to relinquish the line until the remote caller hangs up. Has anyone had similar problems with the SPA3102 or has any ideas where I can look to get to the bottom of the problem. (I have just upgraded the SPA3102 to the latest 5.1.0 firmware) regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/bd825583/attachment-0001.html From jpitcher at nuvio.com Tue Nov 10 08:03:00 2009 From: jpitcher at nuvio.com (Jonathan Pitcher) Date: Tue, 10 Nov 2009 08:03:00 -0800 Subject: [Freeswitch-users] Dialplans and XML_CURL Message-ID: Good morning everyone. I have a question regarding using MOD XML_CURL and returning a dial plan. I have my system setup to respond with the following dialplan.
My question is this. Can extension one, use extension two and three without XML_CURL making another dialplan request? Thanks, Jonathan Pitcher -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/08c48564/attachment.html From mrene_lists at avgs.ca Tue Nov 10 08:55:58 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 10 Nov 2009 08:55:58 -0800 Subject: [Freeswitch-users] Dialplans and XML_CURL In-Reply-To: References: Message-ID: <7319C8C2-08D6-4E8F-AFAE-F9318700F6BD@avgs.ca> You'll get a single xml curl request, unless you use the transfer application, which will trigger another one. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 10-Nov-09, at 8:03 AM, Jonathan Pitcher wrote: > Good morning everyone. I have a question regarding using MOD > XML_CURL and returning a dial plan. > > I have my system setup to respond with the following dialplan. > > > > >
> > > > > > > > > > > > > > > > > > > > > >
> > > My question is this. Can extension one, use extension two and three > without XML_CURL making another dialplan request? > > Thanks, > > Jonathan Pitcher > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/f9f2e4c0/attachment.html From msc at freeswitch.org Tue Nov 10 09:32:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Nov 2009 09:32:38 -0800 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <26284109.post@talk.nabble.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> <26284109.post@talk.nabble.com> Message-ID: <87f2f3b90911100932i19c7c971y5fae90f6bb9f4dc0@mail.gmail.com> I believe that French and Spanish sounds are in the works by the community. The only other sounds I'm aware of are the Russian ones. -MC On Tue, Nov 10, 2009 at 6:13 AM, Fred-145 wrote: > > Are non-English sound files available in the SVN version of the code? > > I just tried installing the French sound files, but got an error: > > Unknown target cd-sounds-fr-install > Unknown target cd-moh-fr-install > make[1]: *** [cd-sounds-fr-install] Error 1 > make: *** [cd-sounds-fr-install] Error 2 > make[1]: *** [cd-moh-fr-install] Error 1 > make: *** [cd-moh-fr-install] Error 2 > [1]+ Exit 2 make cd-sounds-fr-install > -- > View this message in context: > http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26284109.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/370738d2/attachment.html From msc at freeswitch.org Tue Nov 10 09:44:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Nov 2009 09:44:17 -0800 Subject: [Freeswitch-users] Dialplans and XML_CURL In-Reply-To: <7319C8C2-08D6-4E8F-AFAE-F9318700F6BD@avgs.ca> References: <7319C8C2-08D6-4E8F-AFAE-F9318700F6BD@avgs.ca> Message-ID: <87f2f3b90911100944i55a75b5u44f46bd3f906a4b6@mail.gmail.com> On Tue, Nov 10, 2009 at 8:55 AM, Mathieu Rene wrote: > You'll get a single xml curl request, unless you use the transfer > application, which will trigger another one. > Just curious: what about execute_extension? Does that cause a new XML CURL request also? I didn't see anything on the wiki about that. I'll update the wiki mod_xml_curl page accordingly. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/5e8f7438/attachment.html From msc at freeswitch.org Tue Nov 10 09:53:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Nov 2009 09:53:11 -0800 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20091110155633.GA194@hijacked.us> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> <20091109192904.GI9418@hijacked.us> <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> <20091110155633.GA194@hijacked.us> Message-ID: <87f2f3b90911100953n78ae7a54kd253840217188827@mail.gmail.com> On Tue, Nov 10, 2009 at 7:56 AM, Andrew Thompson wrote: > On Tue, Nov 10, 2009 at 11:29:37PM +0800, mark morreny wrote: > > Hi > > > > Thanks for the tips. May I ask how to split the file from hadoop to the > > shell? Is it like copying the file to certain dir? > > > > I can't find any mod_shell_stream related info from the wiki. Does > anyone > > know how to use it? > > > > mod_shell_stream is undocumented, but from reading the code I gather it > works like this: > > Module calls fork() and in the child process it runs an arbitrary shell > command (specified in its config file?). The parent process then reads > raw audio data from the child process and uses it as an audio source. > > So basicially you could write the shell command in anything, so long as > it outputs raw audio to FS. > > Or maybe I read the code wrong when I skimmed over it. If you do get it > working, please contribute some documentation to the wiki. > > Andrew > > Andrew, Thanks for poking around in there. I made a stub for this mod on the wiki: http://wiki.freeswitch.org/wiki/Mod_shell_stream If anyone is familiar with it and could throw an example up there that would be much appreciated. Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/e29fbe92/attachment.html From info at daccii.it Tue Nov 10 10:05:00 2009 From: info at daccii.it (Daniele Salvatore Albano) Date: Tue, 10 Nov 2009 19:05:00 +0100 Subject: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it In-Reply-To: References: <4AF963E7.2080104@daccii.it> <5BB9DB8A-8309-463C-B06F-BDA2A2EDF97E@freeswitch.org> <4AF97AE9.8090103@daccii.it> Message-ID: <4AF9AB4C.6030507@daccii.it> Hi, thank you and for your work! Where i can find coding style rules? Best Regards, Daniele Brian West ha scritto: > Patch applied. // comments aren't allowed in .c files in our tree we > try hard to weed them out... anyway its committed now. And thanks for > your contribution. > > /b -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 307 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/5cdceb49/attachment.vcf From jmesquita at freeswitch.org Tue Nov 10 11:04:16 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 10 Nov 2009 17:04:16 -0200 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> Message-ID: Rupa, I tried flipping, yes, but no, it works bad... Thanks for the heads up on G722! Regards, JM On Tue, Nov 10, 2009 at 1:09 PM, Rupa Schomaker wrote: > On Tue, Nov 10, 2009 at 6:56 AM, Rupa Schomaker wrote: > > Oh, and it looks like a new firmware came out for the Siemens today. > > Wonder what it fixes (and breaks). Hmm.. wonder where I can find a > > list of whats new. > > Well, it seems to totally break g722. I haven't had a chance to > narrow it down further, but beware if g722 is important don't > update... > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/f41665d9/attachment-0001.html From oseslija at gmail.com Tue Nov 10 11:56:38 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Tue, 10 Nov 2009 20:56:38 +0100 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: <1257798401.10738.18.camel@sodium> References: <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> <1257798401.10738.18.camel@sodium> Message-ID: <4468a6770911101156q3083747bl77976f98e930c045@mail.gmail.com> Hey Hadley, jump up on irc sometimes. Regards, Ognjen On Mon, Nov 9, 2009 at 9:26 PM, Hadley Rich wrote: > On Mon, 2009-11-09 at 14:05 -0600, Brian West wrote: > > Get an ATA with a Dect handset it works much better... the Snom M3 and > > the Aastra are one in the same and they both do not live up to the > > quality or usability requirements. > > That said, they are better than what else is around. > > I'd call them average. Nothing to write home about but you don't need to > run away from them. > > hads > > -- > http://nicegear.co.nz > New Zealand's Open Source Hardware Supplier > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/6d1dfbea/attachment.html From sergey.kobzar at mail.ru Tue Nov 10 13:27:20 2009 From: sergey.kobzar at mail.ru (Sergey Kobzar) Date: Tue, 10 Nov 2009 23:27:20 +0200 Subject: [Freeswitch-users] SIP trunk without authentication Message-ID: <1352396721.20091110232720@mail.ru> Hello. I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP. I've searched through FS documentation and didn't find clear answer. Could you help me or maybe give a link to a doc which can help? Thanks. -- Sergey From mrene_lists at avgs.ca Tue Nov 10 13:43:04 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 10 Nov 2009 13:43:04 -0800 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: <1352396721.20091110232720@mail.ru> References: <1352396721.20091110232720@mail.ru> Message-ID: As easy as: in your dialplan. If you want to make a gateway out of it, you can enter whatever you want in username and password since they won't be used. (SIP works using challenge authentication which means the remote UA has to send you a packet requesting the credentials). Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote: > Hello. > > I'm FS newbie and want connect it to SIP provider which does not > require authentication - it make authentication using my IP. > > I've searched through FS documentation and didn't find clear answer. > > Could you help me or maybe give a link to a doc which can help? > > Thanks. > > > -- > Sergey > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nandy1925 at gmail.com Tue Nov 10 13:48:15 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 11 Nov 2009 05:48:15 +0800 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> Message-ID: <7d0bfd8c0911101348n5d7dfd20p224d972d68a1299d@mail.gmail.com> just change the dialplan/default.xml as mentioned by brian but i think you can't use # as the first key 'cuz it normally used as a Send key. you may change # to * (star key). On Wed, Nov 11, 2009 at 12:06 AM, Ognjen Seslija wrote: > Add the following: > > . > > after > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > > in local extensions default example, or change it globally previously than > this extension. You can join us on IRC if you can any more questions > (sekil). > > Regards, > Ognjen > > > > On Tue, Nov 10, 2009 at 4:01 PM, Piotr ?urek wrote: > >> Hello. >> >> Thank You developers for Freeswitch. >> I have installed it lately and it's working quite nicely, but I have one >> problem: >> >> I need to mimic behavior of my current analogue PBX installation using >> Freeswitch. >> >> This is the scenario: >> In the office with a few desks (extensions 1000-1010) and only one person >> behind one of desks (whatever extension - in example 1000). >> 1. There's incoming call on _one_ of extensions 1001-1010 >> 2. The person on extension 1000 wants to answer this call on his phone so >> dials #37 and this call is redirected to his phone. >> >> That's how it works on my office on analogue PBX system. Anyone can answer >> a call from any other phone as long as it hasn't been answered already. >> >> I tried to use the intercept action (with global example in default >> config) but it's not what I need because it intercepts the call even if it's >> already answered. I need to intercept all but only unanswered calls. I tried >> to use Redirect but it does not work on other's extensions call's (or does >> it?). >> >> Please help. >> Peter ?urek >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/45b24bf1/attachment.html From brian at freeswitch.org Tue Nov 10 14:23:43 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 16:23:43 -0600 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <7d0bfd8c0911101348n5d7dfd20p224d972d68a1299d@mail.gmail.com> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> <7d0bfd8c0911101348n5d7dfd20p224d972d68a1299d@mail.gmail.com> Message-ID: <030D9DFF-7AFE-4942-8BEF-B374F8600396@freeswitch.org> That depends on the phone... some let you do it.. some don't... WELCOME TO VOIP!!! /b On Nov 10, 2009, at 3:48 PM, Nandy Dagondon wrote: > just change the dialplan/default.xml as mentioned by brian but i > think you can't use # as the first key 'cuz it normally used as a > Send key. you may change # to * (star key). From ujjval at simplesignal.com Tue Nov 10 14:39:33 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Tue, 10 Nov 2009 14:39:33 -0800 Subject: [Freeswitch-users] Simple Conference Setup issue Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175CF5087E@EXMBXCLUS01.citservers.local> I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch and when I have 2 people dial in, looks like the Music on Hold never stops. Here is what my public.xml looks like: Help appreciated -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/cf1e9e40/attachment.html From brian at freeswitch.org Tue Nov 10 14:49:48 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 16:49:48 -0600 Subject: [Freeswitch-users] Simple Conference Setup issue In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175CF5087E@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175CF5087E@EXMBXCLUS01.citservers.local> Message-ID: <0B5A83F9-9754-44ED-A5C7-447B7F050255@freeswitch.org> What does your config look like? /b On Nov 10, 2009, at 4:39 PM, Ujjval Karihaloo wrote: > I am trying to call into a DID that is pointed to a Conf Bridge on > Freeswitch and when I have 2 people dial in, looks like the Music on > Hold never stops. > > Here is what my public.xml looks like: > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/bc2edeef/attachment-0001.html From ujjval at simplesignal.com Tue Nov 10 15:09:30 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Tue, 10 Nov 2009 15:09:30 -0800 Subject: [Freeswitch-users] Simple Conference Setup issue In-Reply-To: <0B5A83F9-9754-44ED-A5C7-447B7F050255@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175CF5087E@EXMBXCLUS01.citservers.local> <0B5A83F9-9754-44ED-A5C7-447B7F050255@freeswitch.org> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175CF508A9@EXMBXCLUS01.citservers.local> My mistake , it picked the default profile and was waiting for moderator in the conference.cof.xml file that is provided with the install. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, November 10, 2009 3:50 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Simple Conference Setup issue What does your config look like? /b On Nov 10, 2009, at 4:39 PM, Ujjval Karihaloo wrote: I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch and when I have 2 people dial in, looks like the Music on Hold never stops. Here is what my public.xml looks like: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/30b34810/attachment.html From malay.thakershi at continuityhealth.com Tue Nov 10 15:25:24 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Tue, 10 Nov 2009 17:25:24 -0600 Subject: [Freeswitch-users] Help with dynamic IVR Message-ID: <006101ca625d$14257480$3c705d80$@thakershi@continuityhealth.com> Hello. I am very new to FreeSwitch, Telephony and IVR. My goal is to prepare a student assessment IVR system as a college project. But this IVR is going to be dynamic. So for each student assessment may be different (number of questions, possible responses, flow of prompts, etc). Is it possible to achieve something like this with FreeSwitch? Most IVR we see are static (like a bank IVR system that flows always in same way). That is why I am confused. Please share your views. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/56a56a9c/attachment.html From lei.tlfly at gmail.com Tue Nov 10 17:55:15 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 11 Nov 2009 09:55:15 +0800 Subject: [Freeswitch-users] Help with dynamic IVR In-Reply-To: <-5075485054665589342@unknownmsgid> References: <-5075485054665589342@unknownmsgid> Message-ID: <50c41b4e0911101755j6e99a539pc0053befa9086ff9@mail.gmail.com> As I opinion, it's not necessary write ivr script for each student. A "static" ivr script load question and response dynamic is what you need. 2009/11/11 Malay Thakershi > Hello. I am very new to FreeSwitch, Telephony and IVR. > > > > My goal is to prepare a student assessment IVR system as a college project. > But this IVR is going to be dynamic. So for each student assessment may be > different (number of questions, possible responses, flow of prompts, etc). > Is it possible to achieve something like this with FreeSwitch? Most IVR we > see are static (like a bank IVR system that flows always in same way). That > is why I am confused. Please share your views. > > > > Malay Thakershi > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/19742746/attachment.html From markmorreny at gmail.com Tue Nov 10 19:02:10 2009 From: markmorreny at gmail.com (mark morreny) Date: Wed, 11 Nov 2009 11:02:10 +0800 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20091110155633.GA194@hijacked.us> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> <20091109192904.GI9418@hijacked.us> <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> <20091110155633.GA194@hijacked.us> Message-ID: <20ad6b920911101902w52165681tb47fe8f3aa2ae76e@mail.gmail.com> Hi Sorry to ask again. I know the command to copy file from hadoop file system to somewhere else. But how do I make a shell command to output raw audio? What command is it like? Is it like play()? I am confused. Thx, mark On Tue, Nov 10, 2009 at 11:56 PM, Andrew Thompson wrote: > On Tue, Nov 10, 2009 at 11:29:37PM +0800, mark morreny wrote: > > Hi > > > > Thanks for the tips. May I ask how to split the file from hadoop to the > > shell? Is it like copying the file to certain dir? > > > > I can't find any mod_shell_stream related info from the wiki. Does > anyone > > know how to use it? > > > > mod_shell_stream is undocumented, but from reading the code I gather it > works like this: > > Module calls fork() and in the child process it runs an arbitrary shell > command (specified in its config file?). The parent process then reads > raw audio data from the child process and uses it as an audio source. > > So basicially you could write the shell command in anything, so long as > it outputs raw audio to FS. > > Or maybe I read the code wrong when I skimmed over it. If you do get it > working, please contribute some documentation to the wiki. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/2f4d6193/attachment.html From quentusrex at gmail.com Tue Nov 10 19:22:34 2009 From: quentusrex at gmail.com (William King) Date: Tue, 10 Nov 2009 19:22:34 -0800 Subject: [Freeswitch-users] Help with dynamic IVR In-Reply-To: <50c41b4e0911101755j6e99a539pc0053befa9086ff9@mail.gmail.com> References: <-5075485054665589342@unknownmsgid> <50c41b4e0911101755j6e99a539pc0053befa9086ff9@mail.gmail.com> Message-ID: <4AFA2DFA.3070205@gmail.com> What you will probably want, if you are looking to go 'thicker' with this would be one of the IVR scripting languages and a database connection. For instance lua, and the database connection(either mysql or postgresql or sqlite). ' From there you have users, questions, and answers mapped in the database. Feel free to e-mail me about this off list for more assistance. -William King Lei Tang wrote: > As I opinion, it's not necessary write ivr script for each student. A > "static" ivr script load question and response dynamic is what you need. > > 2009/11/11 Malay Thakershi > > > Hello. I am very new to FreeSwitch, Telephony and IVR. > > > > My goal is to prepare a student assessment IVR system as a college > project. But this IVR is going to be dynamic. So for each student > assessment may be different (number of questions, possible > responses, flow of prompts, etc). Is it possible to achieve > something like this with FreeSwitch? Most IVR we see are static > (like a bank IVR system that flows always in same way). That is > why I am confused. Please share your views. > > > > Malay Thakershi > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Lei.Tang > lei.tlfly at gmail.com > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mitch.capper at gmail.com Tue Nov 10 20:01:44 2009 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 10 Nov 2009 23:01:44 -0500 Subject: [Freeswitch-users] Displaying caller ID on LED? In-Reply-To: <26280912.post@talk.nabble.com> References: <26280730.post@talk.nabble.com> <26280912.post@talk.nabble.com> Message-ID: I did something like this recently. From the dial plan it is easy to execute an external application on an incoming call with the caller's info. At that point if you can just push it down to the LCD panel all the better, but if your FS server is remote, and has no direct access to the client to render the caller ID, you will have to setup a fake push to get instant responses. You can do this through apache, or a simple tcp server but the idea being the client connects up to the server, and the server blocks until an incoming call comes in, it then responds to the client, and you have the caller id fairly instantly showing up. You could also use the event socket, heck even maybe use the event socket remotely if you wanted to, and then avoid some of the server side complexity too. ~Mitch On Tue, Nov 10, 2009 at 5:06 AM, Fred-145 wrote: > > ... or alternatively, on one of those USB digital picture frames? > > www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B000087BHC > -- > View this message in context: > http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/91f8f996/attachment.html From lakindia89 at gmail.com Tue Nov 10 20:10:52 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Wed, 11 Nov 2009 09:40:52 +0530 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events In-Reply-To: <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> References: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> Message-ID: <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> Here is the required detail. http://pastebin.freeswitch.org/11049 On Mon, Nov 9, 2009 at 10:04 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > 1) install gdb > 2) run support_d/fscore_db in the tree from the working directory of the > core. > 3) if you are not on svn trunk, "make current" and start over. > > > On Mon, Nov 9, 2009 at 5:53 AM, lakshmanan ganapathy > wrote: > >> Dear all, >> I did the below code, to callback a function when CHANNEL_EXECUTE_COMPLETE >> event comes. >> I executed the script for the 1st time and I got nothing. >> When I executed the script for the 2nd time, it ended with Sedmentation >> fault with core dumped. >> >> I was unable to attach the core dump file with this mail. >> Please specify how to send files to freeswitch user mailing list if need >> be. >> >> The freeswitch log is here: >> http://pastebin.freeswitch.org/11038 >> >> #!/usr/bin/perl >> use strict; >> use Data::Dumper; >> our $session; >> $session->answer(); >> my $events=new freeswitch::EventConsumer("CHANNEL_EXECUTE_COMPLETE"); >> $events->pop(1); >> $events->swig_e_callback_set("playvoice"); >> sub playvoice() >> { >> freeswitch::consoleLog("INFO","Call back function called\n"); >> } >> return 1; >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/8c7938e0/attachment.html From jmesquita at freeswitch.org Tue Nov 10 20:12:03 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 11 Nov 2009 01:12:03 -0300 Subject: [Freeswitch-users] Displaying caller ID on LED? In-Reply-To: References: <26280730.post@talk.nabble.com> <26280912.post@talk.nabble.com> Message-ID: If you donate one to the FsGui project, I can make it happen for you. Contact me off list if you are interested. Regards, JM On Wed, Nov 11, 2009 at 1:01 AM, Mitch Capper wrote: > I did something like this recently. From the dial plan it is easy to > execute an external application on an incoming call with the caller's info. > At that point if you can just push it down to the LCD panel all the better, > but if your FS server is remote, and has no direct access to the client to > render the caller ID, you will have to setup a fake push to get instant > responses. You can do this through apache, or a simple tcp server but the > idea being the client connects up to the server, and the server blocks until > an incoming call comes in, it then responds to the client, and you have the > caller id fairly instantly showing up. You could also use the event > socket, heck even maybe use the event socket remotely if you wanted to, and > then avoid some of the server side complexity too. > > ~Mitch > > > On Tue, Nov 10, 2009 at 5:06 AM, Fred-145 wrote: > >> >> ... or alternatively, on one of those USB digital picture frames? >> >> www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B000087BHC >> -- >> View this message in context: >> http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/6c164975/attachment.html From brian at freeswitch.org Tue Nov 10 20:39:01 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 22:39:01 -0600 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events In-Reply-To: <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> References: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> Message-ID: <2E2AE3DF-1EA7-45AA-9C64-359C5D7585B7@freeswitch.org> You need to install the debug packages so you the symbols because that backtrace is useless. /b On Nov 10, 2009, at 10:10 PM, lakshmanan ganapathy wrote: > Here is the required detail. > > http://pastebin.freeswitch.org/11049 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/b2b63900/attachment.html From mrene_lists at avgs.ca Tue Nov 10 21:18:40 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 10 Nov 2009 21:18:40 -0800 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events In-Reply-To: <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> References: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> Message-ID: It doesn't look like its trying to look for symbols inside freeswitch gdb /path/to/freeswitch/here /path/to/core/here bt thread apply all bt Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 10-Nov-09, at 8:10 PM, lakshmanan ganapathy wrote: > Here is the required detail. > > http://pastebin.freeswitch.org/11049 > > On Mon, Nov 9, 2009 at 10:04 PM, Anthony Minessale > wrote: > 1) install gdb > 2) run support_d/fscore_db in the tree from the working directory of > the core. > 3) if you are not on svn trunk, "make current" and start over. > > > On Mon, Nov 9, 2009 at 5:53 AM, lakshmanan ganapathy > wrote: > Dear all, > I did the below code, to callback a function when > CHANNEL_EXECUTE_COMPLETE event comes. > I executed the script for the 1st time and I got nothing. > When I executed the script for the 2nd time, it ended with > Sedmentation fault with core dumped. > > I was unable to attach the core dump file with this mail. > Please specify how to send files to freeswitch user mailing list if > need be. > > The freeswitch log is here: > http://pastebin.freeswitch.org/11038 > > #!/usr/bin/perl > use strict; > use Data::Dumper; > our $session; > $session->answer(); > my $events=new freeswitch::EventConsumer("CHANNEL_EXECUTE_COMPLETE"); > $events->pop(1); > $events->swig_e_callback_set("playvoice"); > sub playvoice() > { > freeswitch::consoleLog("INFO","Call back function called\n"); > } > return 1; > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/2309380e/attachment-0001.html From lakindia89 at gmail.com Tue Nov 10 21:22:49 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Wed, 11 Nov 2009 10:52:49 +0530 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events In-Reply-To: <2E2AE3DF-1EA7-45AA-9C64-359C5D7585B7@freeswitch.org> References: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> <2E2AE3DF-1EA7-45AA-9C64-359C5D7585B7@freeswitch.org> Message-ID: <7d79b3930911102122jf3f44t7bc035960980678d@mail.gmail.com> What is meant by debug packages. Kindly specify where it is available. On Wed, Nov 11, 2009 at 10:09 AM, Brian West wrote: > You need to install the debug packages so you the symbols because that > backtrace is useless. > /b > > On Nov 10, 2009, at 10:10 PM, lakshmanan ganapathy wrote: > > Here is the required detail. > > http://pastebin.freeswitch.org/11049 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/5f8269d5/attachment.html From lakindia89 at gmail.com Tue Nov 10 21:26:34 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Wed, 11 Nov 2009 10:56:34 +0530 Subject: [Freeswitch-users] Flushing the Event buffer in Perl Event Socket In-Reply-To: <2CD889D1-934A-47CE-A938-1EB9D4325DD2@freeswitch.org> References: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com> <191c3a030910300650w7b80568eu4c41c805b9372acc@mail.gmail.com> <26281493.post@talk.nabble.com> <2CD889D1-934A-47CE-A938-1EB9D4325DD2@freeswitch.org> Message-ID: <7d79b3930911102126u63ba7e8rce1f15ab6371ca2c@mail.gmail.com> That doesn't seems to work for me. Here is my need. I'm using Async in the Event socket outbound. I'll register for "events plain all" I'll answer the call. I'll playback a message. I'll sleep for 5 seconds. After that, I'll receive the events. I don't need the events that are for answer and playback. That action is completed and don't want to receive events for those application. I set $|=1 in my ESL script. But it doesn't seems to solve the above issue. Any help!!!!pls!!! On Tue, Nov 10, 2009 at 7:41 PM, Brian West wrote: > $| = 1; > > I think that is what you're lookin for. > > /b > > On Nov 10, 2009, at 4:51 AM, lakshmanan wrote: > > > I was in a need of flushing the events buffer without reading > > it.I've done > > the following ESL(Async) program to flush the events. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/ef17b7db/attachment.html From andrew at hijacked.us Tue Nov 10 21:38:17 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Wed, 11 Nov 2009 00:38:17 -0500 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20ad6b920911101902w52165681tb47fe8f3aa2ae76e@mail.gmail.com> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> <20091109192904.GI9418@hijacked.us> <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> <20091110155633.GA194@hijacked.us> <20ad6b920911101902w52165681tb47fe8f3aa2ae76e@mail.gmail.com> Message-ID: <20091111053816.GA19599@hijacked.us> On Wed, Nov 11, 2009 at 11:02:10AM +0800, mark morreny wrote: > Hi > > Sorry to ask again. > > I know the command to copy file from hadoop file system to somewhere else. > But how do I make a shell command to output raw audio? > What command is it like? Is it like play()? I am confused. > I was very nice and wrote up some documentation (and 2 examples) on the wiki page at http://wiki.freeswitch.org/wiki/Mod_shell_stream Now you know everything I know about using this module (which is a very cool module, by the way - thanks Tony). Andrew From lei.tlfly at gmail.com Wed Nov 11 05:43:58 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 11 Nov 2009 21:43:58 +0800 Subject: [Freeswitch-users] How to test FS rtp packet lost rate? Message-ID: <50c41b4e0911110543m7b5431ecu173d8386073fdb32@mail.gmail.com> Hi all, I'm testing a FS server using sipp, I found that sipp only show the retrans of sip packet, Does someone known is there a tool to test FS rtp packet lost rate in high concurrent call env? -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/9441f535/attachment.html From dome at tel.co.th Wed Nov 11 07:49:09 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 11 Nov 2009 22:49:09 +0700 Subject: [Freeswitch-users] How to test mod_distributor ? Message-ID: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> I found mod_distributor in SVN. I want to know how does it work ? BG Dome C. From kristian.kielhofner at gmail.com Wed Nov 11 08:49:33 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 11:49:33 -0500 Subject: [Freeswitch-users] How to test FS rtp packet lost rate? In-Reply-To: <50c41b4e0911110543m7b5431ecu173d8386073fdb32@mail.gmail.com> References: <50c41b4e0911110543m7b5431ecu173d8386073fdb32@mail.gmail.com> Message-ID: <2d9149cd0911110849u74c2d8d1jb90de8c20cacde9a@mail.gmail.com> The simplest way I know of is to bring up another call from a local phone and listen to the audio. At the same time run tcpdump/etc with a strict filter to capture the rtp to/from that phone. You can then run RTP stream analysis and the like in Wireshark to identify any lost packets. While this obviously won't identify any/all potential lost packets it will be a lot more practical than any of the alternatives: - Capturing all media streams for RTP analysis - Implementing RTCP to identify lost packets - Commercial hardware/software If FreeSWITCH, your machine, or your network are pushed to the max and falling apart you're most likely going to see audio problems on your single (captured) call. On Wed, Nov 11, 2009 at 8:43 AM, Lei Tang wrote: > Hi all, I'm testing a FS server using sipp, I found that sipp only show the > retrans of sip packet, Does someone known is there a tool to test FS rtp > packet lost rate in high concurrent call env? > > -- > Lei.Tang > lei.tlfly at gmail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From juanbackson at gmail.com Wed Nov 11 09:08:45 2009 From: juanbackson at gmail.com (Juan Backson) Date: Thu, 12 Nov 2009 01:08:45 +0800 Subject: [Freeswitch-users] how to rewrite freeswitch SDP Message-ID: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> Hi, I am using 1.0.4 version of freeswitch and I am doing proxy_media for all calls. Basically, I just proxy all media from one gateway to another with freeswitch serving as a middleman. In the outgoing invite, I found that the owner line ( o= ) in SDP is showing the originator's IP which I would like to avoid. Is there anyway to rewirte part of the SDP for the outgoing invite? thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/83d0e05a/attachment.html From kristian.kielhofner at gmail.com Wed Nov 11 09:15:16 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 12:15:16 -0500 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <030D9DFF-7AFE-4942-8BEF-B374F8600396@freeswitch.org> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> <7d0bfd8c0911101348n5d7dfd20p224d972d68a1299d@mail.gmail.com> <030D9DFF-7AFE-4942-8BEF-B374F8600396@freeswitch.org> Message-ID: <2d9149cd0911110915m9468422yc7da8d460e68335@mail.gmail.com> It's also configurable on some phones... As Brian said, welcome to VoIP! ;) On Tue, Nov 10, 2009 at 5:23 PM, Brian West wrote: > That depends on the phone... some let you do it.. some don't... > WELCOME TO VOIP!!! > > /b -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From kristian.kielhofner at gmail.com Wed Nov 11 09:23:07 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 12:23:07 -0500 Subject: [Freeswitch-users] how to rewrite freeswitch SDP In-Reply-To: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> References: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> Message-ID: <2d9149cd0911110923q2cf30d9ehddea6f9d1f96662a@mail.gmail.com> This might be a bit too obvious but unless you have a specific reason to use proxy_media (handling goofy codecs is a big one) you could just set proxy_media=false and FreeSWITH will proxy the media (effectively) and rewrite the entire SDP by default. On Wed, Nov 11, 2009 at 12:08 PM, Juan Backson wrote: > Hi, > > I am using 1.0.4 version of freeswitch and I am doing proxy_media for all > calls.? Basically, I just proxy all media from one gateway to another with > freeswitch serving as a middleman. > > In the outgoing invite, I found that the owner line ( o= ) in SDP is showing > the originator's IP which I would like to avoid. > > Is there anyway to rewirte part of the SDP for the outgoing invite? > > thanks, > jb > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Wed Nov 11 09:29:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Nov 2009 11:29:14 -0600 Subject: [Freeswitch-users] how to rewrite freeswitch SDP In-Reply-To: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> References: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> Message-ID: You use OpenSER /b On Nov 11, 2009, at 11:08 AM, Juan Backson wrote: > Hi, > > I am using 1.0.4 version of freeswitch and I am doing proxy_media > for all calls. Basically, I just proxy all media from one gateway > to another with freeswitch serving as a middleman. > > In the outgoing invite, I found that the owner line ( o= ) in SDP is > showing the originator's IP which I would like to avoid. > > Is there anyway to rewirte part of the SDP for the outgoing invite? > > thanks, > jb From msc at freeswitch.org Wed Nov 11 09:45:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Nov 2009 09:45:21 -0800 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> Message-ID: <87f2f3b90911110945p56104c19m5f7f06dc431a352d@mail.gmail.com> On Wed, Nov 11, 2009 at 7:49 AM, Dome Charoenyost wrote: > I found mod_distributor in SVN. I want to know how does it work ? > It's brand new - I haven't even seen it yet. I will start documenting it shortly. In the meantime if anyone else has started playing with it please let me know. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/85c7058c/attachment-0001.html From anthony.minessale at gmail.com Wed Nov 11 10:32:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Nov 2009 12:32:27 -0600 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> Message-ID: <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> see conf/autoload_configs/distributor.conf.xml in your dialplan you can use ${distributor(test)} which will cycle expanding to foo1 1/10 times and foo2 the other 9 so imagine if foo1 or foo2 were the names of gateways, or hosts of a remote box basic jist is to set total-weight to a number of arbitrary units then set several nodes with weight elements that add up to that number to break down how many times that node text should be returned out of the total. Remember to use the most simplified reduced value for your fractions to get the most variety. Setting total weight to 1000 and then 2 nodes with 100 and 900 would result in foo1 100 times in a row, then foo2 900 times in a row. On Wed, Nov 11, 2009 at 9:49 AM, Dome Charoenyost wrote: > I found mod_distributor in SVN. I want to know how does it work ? > > BG > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/a826214b/attachment.html From christian.loeschenkohl at xpirio.com Wed Nov 11 10:50:52 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Wed, 11 Nov 2009 19:50:52 +0100 Subject: [Freeswitch-users] how to rewrite freeswitch SDP In-Reply-To: <2d9149cd0911110923q2cf30d9ehddea6f9d1f96662a@mail.gmail.com> References: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> <2d9149cd0911110923q2cf30d9ehddea6f9d1f96662a@mail.gmail.com> Message-ID: <4AFB078C.9040008@xpirio.com> hi but this wouldn't work for larger volumens, g729 and t.38 or am i wrong on this? br On 2009-11-11 18:23, Kristian Kielhofner wrote: > This might be a bit too obvious but unless you have a specific reason > to use proxy_media (handling goofy codecs is a big one) you could just > set proxy_media=false and FreeSWITH will proxy the media (effectively) > and rewrite the entire SDP by default. > > On Wed, Nov 11, 2009 at 12:08 PM, Juan Backson wrote: >> Hi, >> >> I am using 1.0.4 version of freeswitch and I am doing proxy_media for all >> calls. Basically, I just proxy all media from one gateway to another with >> freeswitch serving as a middleman. >> >> In the outgoing invite, I found that the owner line ( o= ) in SDP is showing >> the originator's IP which I would like to avoid. >> >> Is there anyway to rewirte part of the SDP for the outgoing invite? >> >> thanks, >> jb >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From djbinter at yahoo.com Wed Nov 11 11:19:12 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 11 Nov 2009 11:19:12 -0800 (PST) Subject: [Freeswitch-users] mod_distributor for bridge Message-ID: <571698.13794.qm@web37508.mail.mud.yahoo.com> Anthony, Would this configuration work if we want to do load sharing 50/50: #distributor.conf.xml ---------------------------------------------------- #sip_profiles/external/carrier1.xml --------------------------------------------------- #dialplan/defalut/01_outbound_routes.xml Thank you, Dorn B. From anthony.minessale at gmail.com Wed Nov 11 11:37:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Nov 2009 13:37:31 -0600 Subject: [Freeswitch-users] mod_distributor for bridge In-Reply-To: <571698.13794.qm@web37508.mail.mud.yahoo.com> References: <571698.13794.qm@web37508.mail.mud.yahoo.com> Message-ID: <191c3a030911111137v232db254v15ffbee1b350c7ab@mail.gmail.com> you got 2 out of 3, the dialplan would look like this: On Wed, Nov 11, 2009 at 1:19 PM, DJB wrote: > Anthony, > > Would this configuration work if we want to do load sharing 50/50: > > > > #distributor.conf.xml > > > > > > > > > > > > ---------------------------------------------------- > #sip_profiles/external/carrier1.xml > > > > > > > > > > > > > > > > --------------------------------------------------- > #dialplan/defalut/01_outbound_routes.xml > > > > > data="${distributor(carrier1)}sofia/gateway/gateway1/$1"/> > > > > > data="${distributor(carrier1)}sofia/gateway/gateway2/$1"/> > > > > > > Thank you, > Dorn B. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/b7e8979d/attachment.html From kristian.kielhofner at gmail.com Wed Nov 11 11:56:05 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 14:56:05 -0500 Subject: [Freeswitch-users] how to rewrite freeswitch SDP In-Reply-To: References: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> Message-ID: <2d9149cd0911111156k3857d644jafd89d687f4dc1aa@mail.gmail.com> This is correct. The nathelper module and RTPProxy have an option to rewrite o= as well as c=. On Wed, Nov 11, 2009 at 12:29 PM, Brian West wrote: > You use OpenSER > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From djbinter at yahoo.com Wed Nov 11 11:57:29 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 11 Nov 2009 11:57:29 -0800 (PST) Subject: [Freeswitch-users] mod_distributor for bridge In-Reply-To: <191c3a030911111137v232db254v15ffbee1b350c7ab@mail.gmail.com> References: <571698.13794.qm@web37508.mail.mud.yahoo.com> <191c3a030911111137v232db254v15ffbee1b350c7ab@mail.gmail.com> Message-ID: <183302.38522.qm@web37501.mail.mud.yahoo.com> Thank you. I will test with the latest SVN; however, can you please advise how do I add it in modules.conf since I don't see the item in there so that I can rebuild the source. Regards, Dorn B. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 11:37:31 AM Subject: Re: [Freeswitch-users] mod_distributor for bridge you got 2 out of 3, the dialplan would look like this: On Wed, Nov 11, 2009 at 1:19 PM, DJB wrote: Anthony, > >>Would this configuration work if we want to do load sharing 50/50: > > > >>#distributor.conf.xml > > >> >> >> >> >> >> >> >> > >>---------------------------------------------------- >>#sip_profiles/external/carrier1.xml > >> >> >> >> >> >> >> >> >> >> >> >> >> > >>--------------------------------------------------- >>#dialplan/defalut/01_outbound_routes.xml > >> >> >> >> >> >> >> >> >> >> >> >> > > >>Thank you, >>Dorn B. > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/40e5b946/attachment-0001.html From kristian.kielhofner at gmail.com Wed Nov 11 11:58:12 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 14:58:12 -0500 Subject: [Freeswitch-users] how to rewrite freeswitch SDP In-Reply-To: <4AFB078C.9040008@xpirio.com> References: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> <2d9149cd0911110923q2cf30d9ehddea6f9d1f96662a@mail.gmail.com> <4AFB078C.9040008@xpirio.com> Message-ID: <2d9149cd0911111158g72a2107mf11803a880ba3c69@mail.gmail.com> Brian once told me (at ClueCon) that proxy_media isn't really that much lighter on the CPU. At least that's what I think he said. Anyone care to clarify/quantify? Anyways it would work for G729 (pass through is no problem) but not T.38 (it isn't a recognized codec at all). 2009/11/11 Christian L?schenkohl : > hi > > but this wouldn't work for larger volumens, g729 and t.38 > or am i wrong on this? > > br > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From msc at freeswitch.org Wed Nov 11 12:44:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Nov 2009 12:44:49 -0800 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> Message-ID: <87f2f3b90911111244r130945b7w7813c71fbbd12738@mail.gmail.com> Perfect. I'll have it documented by the end of the day. -MC On Wed, Nov 11, 2009 at 10:32 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > see conf/autoload_configs/distributor.conf.xml > > > > > > > > > > > > > > in your dialplan you can use > > ${distributor(test)} which will cycle expanding to foo1 1/10 times and foo2 > the other 9 > > so imagine if foo1 or foo2 were the names of gateways, or hosts of a remote > box > > basic jist is to set total-weight to a number of arbitrary units then set > several nodes with weight elements that add up to that number to break down > how many times that node text should be returned out of the total. > > Remember to use the most simplified reduced value for your fractions to get > the most variety. > > Setting total weight to 1000 and then 2 nodes with 100 and 900 would result > in foo1 100 times in a row, then foo2 900 times in a row. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/c4f3cea7/attachment.html From djbinter at yahoo.com Wed Nov 11 13:06:39 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 11 Nov 2009 13:06:39 -0800 (PST) Subject: [Freeswitch-users] mod_distributor for bridge In-Reply-To: <183302.38522.qm@web37501.mail.mud.yahoo.com> References: <571698.13794.qm@web37508.mail.mud.yahoo.com> <191c3a030911111137v232db254v15ffbee1b350c7ab@mail.gmail.com> <183302.38522.qm@web37501.mail.mud.yahoo.com> Message-ID: <878657.77487.qm@web37503.mail.mud.yahoo.com> Actually, I got it. I've added: applications/mod_distributor in the modules.conf I will start testing now. Thank you, Dorn B. ________________________________ From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 11:57:29 AM Subject: Re: [Freeswitch-users] mod_distributor for bridge Thank you. I will test with the latest SVN; however, can you please advise how do I add it in modules.conf since I don't see the item in there so that I can rebuild the source. Regards, Dorn B. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 11:37:31 AM Subject: Re: [Freeswitch-users] mod_distributor for bridge you got 2 out of 3, the dialplan would look like this: On Wed, Nov 11, 2009 at 1:19 PM, DJB wrote: Anthony, > >>Would this configuration work if we want to do load sharing 50/50: > > > >>#distributor.conf.xml > > >> >> >> >> >> >> >> >> > >>---------------------------------------------------- >>#sip_profiles/external/carrier1.xml > >> >> >> >> >> >> >> >> >> >> >> >> >> > >>--------------------------------------------------- >>#dialplan/defalut/01_outbound_routes.xml > >> >> >> >> >> >> >> >> >> >> >> >> > > >>Thank you, >>Dorn B. > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/82a39ba5/attachment.html From kristian.kielhofner at gmail.com Wed Nov 11 13:19:47 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 16:19:47 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted Message-ID: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> Full log and trace here: http://pastebin.freeswitch.org/11062 Pretty standard situation. User calls another user (same profile) and tries to place the call on hold (RFC 3264/sendonly). FS places call on hold and tries to start music but ends with: [local_stream://moh] already broadcasting...broadcast aborted ... and we don't get music. Any ideas why? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From djbinter at yahoo.com Wed Nov 11 13:35:28 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 11 Nov 2009 13:35:28 -0800 (PST) Subject: [Freeswitch-users] mod_distributor for bridge In-Reply-To: <878657.77487.qm@web37503.mail.mud.yahoo.com> References: <571698.13794.qm@web37508.mail.mud.yahoo.com> <191c3a030911111137v232db254v15ffbee1b350c7ab@mail.gmail.com> <183302.38522.qm@web37501.mail.mud.yahoo.com> <878657.77487.qm@web37503.mail.mud.yahoo.com> Message-ID: <164694.95836.qm@web37504.mail.mud.yahoo.com> Anthony, I did the test and the load sharing works great. However, I tried to test by failing the first gateway and the load sharing is working correctly, but is there a way that it would fail over and continue to the next one if any of the gateways failed within the list. I've tried with continue_on_fail=true, but it did not work. Thank you, Dorn B. ________________________________ From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 1:06:39 PM Subject: Re: [Freeswitch-users] mod_distributor for bridge Actually, I got it. I've added: applications/mod_distributor in the modules.conf I will start testing now. Thank you, Dorn B. ________________________________ From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 11:57:29 AM Subject: Re: [Freeswitch-users] mod_distributor for bridge Thank you. I will test with the latest SVN; however, can you please advise how do I add it in modules.conf since I don't see the item in there so that I can rebuild the source. Regards, Dorn B. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 11:37:31 AM Subject: Re: [Freeswitch-users] mod_distributor for bridge you got 2 out of 3, the dialplan would look like this: On Wed, Nov 11, 2009 at 1:19 PM, DJB wrote: Anthony, > >>Would this configuration work if we want to do load sharing 50/50: > > > >>#distributor.conf.xml > > >> >> >> >> >> >> >> >> > >>---------------------------------------------------- >>#sip_profiles/external/carrier1.xml > >> >> >> >> >> >> >> >> >> >> >> >> >> > >>--------------------------------------------------- >>#dialplan/defalut/01_outbound_routes.xml > >> >> >> >> >> >> >> >> >> >> >> >> > > >>Thank you, >>Dorn B. > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/45ddd594/attachment-0001.html From brian at freeswitch.org Wed Nov 11 13:52:27 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Nov 2009 15:52:27 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> Message-ID: I noticed you are sending the call to a socket... what did you do to the call in the socket? /b On Nov 11, 2009, at 3:19 PM, Kristian Kielhofner wrote: > Full log and trace here: > > http://pastebin.freeswitch.org/11062 > > Pretty standard situation. User calls another user (same profile) and > tries to place the call on hold (RFC 3264/sendonly). FS places call > on hold and tries to start music but ends with: > > [local_stream://moh] already broadcasting...broadcast aborted > > ... and we don't get music. Any ideas why? > > Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/577cc858/attachment.html From federico.omoto at gmail.com Wed Nov 11 13:58:28 2009 From: federico.omoto at gmail.com (Fede) Date: Wed, 11 Nov 2009 19:58:28 -0200 Subject: [Freeswitch-users] Unable to register UA Message-ID: <8b4221f20911111358u10502dcdva9382bed13cb81a6@mail.gmail.com> Hi! I'm trying to register a SIP UA to my FreeSWITCH server and for some reason I always get a "401 Unauthorized" response. I've tried with other UA (X-Lite and Ekiga) and they do work. The UA is: http://www.doddlephone.com My user configuration is: Can someone help me and tell me what I'm doing wrong? Here's the FreeSWITCH trace if it's useful: freeswitch at fc1160102.aspadmin.net> tport_wakeup_pri(0xae7056b0): events IN tport_recv_event(0xae7056b0) tport_recv_iovec(0xae7056b0) msg 0xae703f88 from (udp/216.75.60.102:5060) has 444 bytes, veclen = 1 recv 444 bytes from udp/[190.179.3.18]:4375 at 21:51:10.194004: ------------------------------------------------------------------------ REGISTER sip:216.75.60.102 SIP/2.0 From: sip:doddle at 216.75.60.102 ;tag=633f3915 To: sip:doddle at 216.75.60.102 Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 101 REGISTER Contact: Expires: 3600 Date: Wed, 11 Nov 2009 21:51:51 GMT Max-Forwards: 70 User-Agent: Doddle WebPhone Supported: replaces Via: SIP/2.0/UDP 192.168.0.1:4375;branch=z9hG4bK-0f9263f10caa;rport Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0xae7056b0): msg 0xae703f88 (444 bytes) from udp/ 190.179.3.18:5060/sip next=(nil) nta: received REGISTER sip:216.75.60.102 SIP/2.0 (CSeq 101) nta: Via check: received=190.179.3.18 nta: canonizing sip:216.75.60.102 with contact nta: REGISTER (101) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0xae7098a0, 0xae704890, 0xae718fc8) called soa_set_params(static::0xae719398, ...) called nua: nua_application_event: entering nua: nua_respond: entering nua(0xae718fc8): sent signal r_respond nua: nua_handle_destroy: entering nua(0xae718fc8): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering soa_set_params(static::0xae719398, ...) called tport_tsend(0xae7056b0) tpn = UDP/190.179.3.18:4375 tport_resolve addrinfo = 190.179.3.18:4375 tport_by_addrinfo(0xae7056b0): not found by name UDP/190.179.3.18:4375 tport_vsend returned 631 send 631 bytes to udp/[190.179.3.18]:4375 at 21:51:10.198733: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.1:4375 ;branch=z9hG4bK-0f9263f10caa;rport=4375;received=190.179.3.18 From: sip:doddle at 216.75.60.102 ;tag=633f3915 To: >;tag=U8QpcZyvrQ3Fg Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 101 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.5pre5-15326M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="216.75.60.102", nonce="132239a5-e37e-4698-af61-5df9b3b67de8", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ nta: sent 401 Unauthorized for REGISTER (101) nta: timer set to 32000 ms nta_leg_destroy((nil)) soa_destroy(static::0xae719398) called tport_wakeup_pri(0xae7056b0): events IN tport_recv_event(0xae7056b0) tport_recv_iovec(0xae7056b0) msg 0xae716af8 from (udp/216.75.60.102:5060) has 706 bytes, veclen = 1 recv 706 bytes from udp/[190.179.3.18]:4375 at 21:51:10.453646: ------------------------------------------------------------------------ REGISTER sip:216.75.60.102 SIP/2.0 From: sip:doddle at 216.75.60.102 ;tag=633f3915 To: sip:doddle at 216.75.60.102 Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 102 REGISTER Expires: 3600 Date: Wed, 11 Nov 2009 21:51:51 GMT Max-Forwards: 70 User-Agent: Doddle WebPhone Supported: replaces Authorization: Digest username="doddle", realm="216.75.60.102", nonce="132239a5-e37e-4698-af61-5df9b3b67de8", uri="sip:216.75.60.102", response="686ec180c04fc70be22aaca9eb21f5e9", algorithm=MD5, cnonce="155fb1307a89ffc0eaed1e0e94958e6e", qop=auth, nc=00000033 Via: SIP/2.0/UDP 192.168.0.1:4375;branch=z9hG4bK-36a1b6039a42;rport Contact: Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0xae7056b0): msg 0xae716af8 (706 bytes) from udp/ 190.179.3.18:5060/sip next=(nil) nta: received REGISTER sip:216.75.60.102 SIP/2.0 (CSeq 102) nta: Via check: received=190.179.3.18 nta: canonizing sip:216.75.60.102 with contact nta: REGISTER (102) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0xae7098a0, 0xae704890, 0xae71ae30) called soa_set_params(static::0xae71b340, ...) called nua: nua_application_event: entering nua: nua_respond: entering nua(0xae71ae30): sent signal r_respond nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0xae71ae30): sent signal r_destroy nua: nua_handle_magic: entering soa_set_params(static::0xae71b340, ...) called nua: nua_handle_destroy: entering tport_tsend(0xae7056b0) tpn = UDP/190.179.3.18:4375 tport_resolve addrinfo = 190.179.3.18:4375 tport_by_addrinfo(0xae7056b0): not found by name UDP/190.179.3.18:4375 tport_vsend returned 645 send 645 bytes to udp/[190.179.3.18]:4375 at 21:51:10.462570: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.1:4375 ;branch=z9hG4bK-36a1b6039a42;rport=4375;received=190.179.3.18 From: sip:doddle at 216.75.60.102 ;tag=633f3915 To: >;tag=vHHFetF0N0S2B Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 102 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.5pre5-15326M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="216.75.60.102", nonce="5a5ebe31-baf6-429c-a184-f835a22d7c24", stale="true", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ nta: sent 401 Unauthorized for REGISTER (102) nta_leg_destroy((nil)) soa_destroy(static::0xae71b340) called Thank you in advacen, Federico Omoto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/c670d981/attachment.html From kristian.kielhofner at gmail.com Wed Nov 11 14:20:40 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 17:20:40 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> Message-ID: <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> >From the trace: # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr.c:540 sofia/pjsip/nobody at 192.168.4.192 Command Execute set(sip_h_X-voalte-call-id=9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec) # EXECUTE sofia/pjsip/nobody at 192.168.4.192 set(sip_h_X-voalte-call-id=9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec) # 2009-11-11 11:23:58.909804 [DEBUG] mod_dptools.c:766 sofia/pjsip/nobody at 192.168.4.192 SET [sip_h_X-voalte-call-id]=[9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr.c:540 sofia/pjsip/nobody at 192.168.4.192 Command Execute ring_ready() # EXECUTE sofia/pjsip/nobody at 192.168.4.192 ring_ready() # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr.c:540 sofia/pjsip/nobody at 192.168.4.192 Command Execute bridge({originate_timeout=30,bypass_media=false,origination_caller_id_number=1001,origination_caller_id_name=Danielle Reed}sofia/voalte/huttoj at 192.168.4.180) # EXECUTE sofia/pjsip/nobody at 192.168.4.192 bridge({originate_timeout=30,bypass_media=false,origination_caller_id_number=1001,origination_caller_id_name=Danielle Reed}sofia/voalte/huttoj at 192.168.4.180) # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 0 = [originate_timeout=30] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 1 = [bypass_media=false] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 2 = [origination_caller_id_number=1001] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 3 = [origination_caller_id_name=Danielle Reed] # 2009-11-11 11:23:58.909804 [NOTICE] switch_channel.c:613 New Channel sofia/pjsip/huttoj at 192.168.4.180 [66e0041f-feff-49c1-baf8-0e5aa1ae99fa] 1) Set a SIP header (this is being removed, actually). 2) Indicate ring_ready (it's a long story). 3) Execute bridge to call the other device. On Wed, Nov 11, 2009 at 4:52 PM, Brian West wrote: > I noticed you are sending the call to a socket... what did you do to the > call in the socket? > /b -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From Prometheus001 at gmx.net Wed Nov 11 14:27:09 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 11 Nov 2009 23:27:09 +0100 Subject: [Freeswitch-users] att_xfer and Loopback Message-ID: <4AFB3A3D.1050602@gmx.net> Hello, I have some problems with attended transfer and loopback Scenario how id work - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped How it work until here: - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, B and C are connected (A still MOH) The dial string for C is dynamic and dependent on certain parameters, therefore C must be called via Loopback in our scenario. Here are the configs: In dialplan for calling B: Dialplan for executing the att_xfer: So this is pretty standard, except the loopback. SVN is 15322. Anybody has a solution for this? Best regards Peter From msc at freeswitch.org Wed Nov 11 14:31:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Nov 2009 14:31:57 -0800 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <87f2f3b90911111244r130945b7w7813c71fbbd12738@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> <87f2f3b90911111244r130945b7w7813c71fbbd12738@mail.gmail.com> Message-ID: <87f2f3b90911111431y7b68856cwfe8a97ec0754f86d@mail.gmail.com> FYI, I added some docs here: http://wiki.freeswitch.org/wiki/Mod_distributor Please feel free to add to it if you are doing anything interesting or creative that hasn't been covered. -MC On Wed, Nov 11, 2009 at 12:44 PM, Michael Collins wrote: > Perfect. I'll have it documented by the end of the day. > -MC > > > On Wed, Nov 11, 2009 at 10:32 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> see conf/autoload_configs/distributor.conf.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> in your dialplan you can use >> >> ${distributor(test)} which will cycle expanding to foo1 1/10 times and >> foo2 the other 9 >> >> so imagine if foo1 or foo2 were the names of gateways, or hosts of a >> remote box >> >> basic jist is to set total-weight to a number of arbitrary units then set >> several nodes with weight elements that add up to that number to break down >> how many times that node text should be returned out of the total. >> >> Remember to use the most simplified reduced value for your fractions to >> get the most variety. >> >> Setting total weight to 1000 and then 2 nodes with 100 and 900 would >> result in foo1 100 times in a row, then foo2 900 times in a row. >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/86b3835a/attachment-0001.html From kristian.kielhofner at gmail.com Wed Nov 11 14:33:15 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 17:33:15 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> Message-ID: <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64. On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner wrote: > From the trace: > ..snip.. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From sergey.kobzar at mail.ru Wed Nov 11 14:33:52 2009 From: sergey.kobzar at mail.ru (Sergey Kobzar) Date: Thu, 12 Nov 2009 00:33:52 +0200 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: References: <1352396721.20091110232720@mail.ru> Message-ID: <633434680.20091112003352@mail.ru> Mathieu, thanks for the help. I got external oubound calls working. The things are simpler then I expected. This is my configuration: I still have 2 questions: 1. Users must type '9' at the beginning, which means this is external call and it must go out through VoIP provider. My config: ... But I see that 9 still exists. 2. Ideally each internal number must have external one. In other words ${outbound_caller_id_number} must be mapped to int. number. Where can I do this? P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use FS default configuration for testing. Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote: > As easy as: > > in your dialplan. If you want to make a gateway out of it, you can > enter whatever you want in username and password since they won't be > used. (SIP works using challenge authentication which means the remote > UA has to send you a packet requesting the credentials). > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote: >> Hello. >> >> I'm FS newbie and want connect it to SIP provider which does not >> require authentication - it make authentication using my IP. >> >> I've searched through FS documentation and didn't find clear answer. >> >> Could you help me or maybe give a link to a doc which can help? >> >> Thanks. >> >> >> -- >> Sergey >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sergey From mrene_lists at avgs.ca Wed Nov 11 14:41:22 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 11 Nov 2009 14:41:22 -0800 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: <633434680.20091112003352@mail.ru> References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> Message-ID: $1 gives you the content of the first regex capture group, so the first ( ) group. ^9(\d{7,})$ would put it in $1 Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Nov-09, at 2:33 PM, Sergey Kobzar wrote: > Mathieu, thanks for the help. I got external oubound calls working. > The things are simpler then I expected. > > This is my configuration: > > > > > > > > > > > I still have 2 questions: > > 1. Users must type '9' at the beginning, which means this is external > call and it must go out through VoIP provider. My config: > > ... > > > But I see that 9 still exists. > > > 2. Ideally each internal number must have external one. In other words > ${outbound_caller_id_number} must be mapped to int. number. Where > can I do this? > > > P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use FS > default configuration for testing. > > > > Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote: > >> As easy as: >> > >> in your dialplan. If you want to make a gateway out of it, you can >> enter whatever you want in username and password since they won't be >> used. (SIP works using challenge authentication which means the >> remote >> UA has to send you a packet requesting the credentials). > >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca > > > > >> On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote: > >>> Hello. >>> >>> I'm FS newbie and want connect it to SIP provider which does not >>> require authentication - it make authentication using my IP. >>> >>> I've searched through FS documentation and didn't find clear answer. >>> >>> Could you help me or maybe give a link to a doc which can help? >>> >>> Thanks. >>> >>> >>> -- >>> Sergey >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Sergey > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Russell.Mosemann at cune.org Wed Nov 11 14:43:20 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Wed, 11 Nov 2009 22:43:20 -0000 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: <633434680.20091112003352@mail.ru> Message-ID: <20091111224320.1A44337F22E@mail.cune.org> > > ... > But I see that 9 still exists. Put the parentheses around the portion you want to capture. http://wiki.freeswitch.org/wiki/Regular_Expression -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From sergey.kobzar at mail.ru Wed Nov 11 15:04:12 2009 From: sergey.kobzar at mail.ru (Sergey Kobzar) Date: Thu, 12 Nov 2009 01:04:12 +0200 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> Message-ID: <1952719808.20091112010412@mail.ru> Ah, right. I was inattentive :) What about my 2nd question? Each user must have unique outbound number which is mapped to his internal number. How can I set ${outbound_caller_id_number} depending on calling internal number? Thursday, November 12, 2009, 12:41:22 AM, Mathieu wrote: > $1 gives you the content of the first regex capture group, so the > first ( ) group. > ^9(\d{7,})$ would put it in $1 > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > On 11-Nov-09, at 2:33 PM, Sergey Kobzar wrote: >> Mathieu, thanks for the help. I got external oubound calls working. >> The things are simpler then I expected. >> >> This is my configuration: >> >> >> >> >> >> >> >> >> >> >> I still have 2 questions: >> >> 1. Users must type '9' at the beginning, which means this is external >> call and it must go out through VoIP provider. My config: >> >> ... >> >> >> But I see that 9 still exists. >> >> >> 2. Ideally each internal number must have external one. In other words >> ${outbound_caller_id_number} must be mapped to int. number. Where >> can I do this? >> >> >> P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use FS >> default configuration for testing. >> >> >> >> Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote: >> >>> As easy as: >>> >> >>> in your dialplan. If you want to make a gateway out of it, you can >>> enter whatever you want in username and password since they won't be >>> used. (SIP works using challenge authentication which means the >>> remote >>> UA has to send you a packet requesting the credentials). >> >>> Mathieu Rene >>> Avant-Garde Solutions Inc >>> Office: + 1 (514) 664-1044 x100 >>> Cell: +1 (514) 664-1044 x200 >>> mrene at avgs.ca >> >> >> >> >>> On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote: >> >>>> Hello. >>>> >>>> I'm FS newbie and want connect it to SIP provider which does not >>>> require authentication - it make authentication using my IP. >>>> >>>> I've searched through FS documentation and didn't find clear answer. >>>> >>>> Could you help me or maybe give a link to a doc which can help? >>>> >>>> Thanks. >>>> >>>> >>>> -- >>>> Sergey >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >> >> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> >> -- >> Sergey >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sergey From mrene_lists at avgs.ca Wed Nov 11 15:09:30 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 11 Nov 2009 15:09:30 -0800 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: <1952719808.20091112010412@mail.ru> References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> Message-ID: Set it in the user directory entry. All variables all loaded whenever the user is authenticated (before the call hits the dialplan) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Nov-09, at 3:04 PM, Sergey Kobzar wrote: > Ah, right. I was inattentive :) > > What about my 2nd question? Each user must have unique outbound number > which is mapped to his internal number. > > How can I set ${outbound_caller_id_number} depending on calling > internal number? > > > > Thursday, November 12, 2009, 12:41:22 AM, Mathieu wrote: > >> $1 gives you the content of the first regex capture group, so the >> first ( ) group. > >> ^9(\d{7,})$ would put it in $1 > >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca > > > > >> On 11-Nov-09, at 2:33 PM, Sergey Kobzar wrote: > >>> Mathieu, thanks for the help. I got external oubound calls working. >>> The things are simpler then I expected. >>> >>> This is my configuration: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I still have 2 questions: >>> >>> 1. Users must type '9' at the beginning, which means this is >>> external >>> call and it must go out through VoIP provider. My config: >>> >>> ... >>> >>> >>> But I see that 9 still exists. >>> >>> >>> 2. Ideally each internal number must have external one. In other >>> words >>> ${outbound_caller_id_number} must be mapped to int. number. Where >>> can I do this? >>> >>> >>> P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use >>> FS >>> default configuration for testing. >>> >>> >>> >>> Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote: >>> >>>> As easy as: >>>> >>> >>>> in your dialplan. If you want to make a gateway out of it, you can >>>> enter whatever you want in username and password since they won't >>>> be >>>> used. (SIP works using challenge authentication which means the >>>> remote >>>> UA has to send you a packet requesting the credentials). >>> >>>> Mathieu Rene >>>> Avant-Garde Solutions Inc >>>> Office: + 1 (514) 664-1044 x100 >>>> Cell: +1 (514) 664-1044 x200 >>>> mrene at avgs.ca >>> >>> >>> >>> >>>> On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote: >>> >>>>> Hello. >>>>> >>>>> I'm FS newbie and want connect it to SIP provider which does not >>>>> require authentication - it make authentication using my IP. >>>>> >>>>> I've searched through FS documentation and didn't find clear >>>>> answer. >>>>> >>>>> Could you help me or maybe give a link to a doc which can help? >>>>> >>>>> Thanks. >>>>> >>>>> >>>>> -- >>>>> Sergey >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>> >>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Sergey >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Sergey > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris at cloudtel.com Wed Nov 11 17:16:50 2009 From: chris at cloudtel.com (Chris Burns) Date: Wed, 11 Nov 2009 20:16:50 -0500 Subject: [Freeswitch-users] Unable to register UA In-Reply-To: <8b4221f20911111358u10502dcdva9382bed13cb81a6@mail.gmail.com> References: <8b4221f20911111358u10502dcdva9382bed13cb81a6@mail.gmail.com> Message-ID: <200911112016.50824.chris@cloudtel.com> Your SIP UA needs to take the info in the 401 and use it to digest authenticate. If you trace a SIP UA that supports authentication you will see that they also get the 401/407 and only then are able to authenticate. This is just a fact of how digest auth works in SIP ... see section 22.4 The Digest Authentication Scheme: http://www.ietf.org/rfc/rfc3261.txt On November 11, 2009 04:58:28 pm Fede wrote: > Hi! > > I'm trying to register a SIP UA to my FreeSWITCH server and for some reason > I always get a "401 Unauthorized" response. I've tried with other UA > (X-Lite and Ekiga) and they do work. The UA is: http://www.doddlephone.com > > My user configuration is: > > > > > > > > > > > > > > > > > > Can someone help me and tell me what I'm doing wrong? > > Here's the FreeSWITCH trace if it's useful: > > freeswitch at fc1160102.aspadmin.net> tport_wakeup_pri(0xae7056b0): events IN > tport_recv_event(0xae7056b0) > tport_recv_iovec(0xae7056b0) msg 0xae703f88 from (udp/216.75.60.102:5060) > has 444 bytes, veclen = 1 > recv 444 bytes from udp/[190.179.3.18]:4375 at 21:51:10.194004: > ------------------------------------------------------------------------ > REGISTER sip:216.75.60.102 SIP/2.0 > From: sip:doddle at 216.75.60.102 ;tag=633f3915 > To: sip:doddle at 216.75.60.102 > Call-Id: 186e708700bcf9a944855105fc3dce0e > Cseq: 101 REGISTER > Contact: > Expires: 3600 > Date: Wed, 11 Nov 2009 21:51:51 GMT > Max-Forwards: 70 > User-Agent: Doddle WebPhone > Supported: replaces > Via: SIP/2.0/UDP 192.168.0.1:4375;branch=z9hG4bK-0f9263f10caa;rport > Content-Length: 0 > > ------------------------------------------------------------------------ > tport_deliver(0xae7056b0): msg 0xae703f88 (444 bytes) from udp/ > 190.179.3.18:5060/sip next=(nil) > nta: received REGISTER sip:216.75.60.102 SIP/2.0 (CSeq 101) > nta: Via check: received=190.179.3.18 > nta: canonizing sip:216.75.60.102 with contact > nta: REGISTER (101) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > soa_clone(static::0xae7098a0, 0xae704890, 0xae718fc8) called > soa_set_params(static::0xae719398, ...) called > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0xae718fc8): sent signal r_respond > nua: nua_handle_destroy: entering > nua(0xae718fc8): sent signal r_destroy > nua: nua_stack_set_params: entering > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > soa_set_params(static::0xae719398, ...) called > tport_tsend(0xae7056b0) tpn = UDP/190.179.3.18:4375 > tport_resolve addrinfo = 190.179.3.18:4375 > tport_by_addrinfo(0xae7056b0): not found by name UDP/190.179.3.18:4375 > tport_vsend returned 631 > send 631 bytes to udp/[190.179.3.18]:4375 at 21:51:10.198733: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.0.1:4375 > ;branch=z9hG4bK-0f9263f10caa;rport=4375;received=190.179.3.18 > From: sip:doddle at 216.75.60.102 ;tag=633f3915 > To: > > >;tag=U8QpcZyvrQ3Fg > > Call-Id: 186e708700bcf9a944855105fc3dce0e > Cseq: 101 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.0.5pre5-15326M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="216.75.60.102", > nonce="132239a5-e37e-4698-af61-5df9b3b67de8", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > nta: sent 401 Unauthorized for REGISTER (101) > nta: timer set to 32000 ms > nta_leg_destroy((nil)) > soa_destroy(static::0xae719398) called > tport_wakeup_pri(0xae7056b0): events IN > tport_recv_event(0xae7056b0) > tport_recv_iovec(0xae7056b0) msg 0xae716af8 from (udp/216.75.60.102:5060) > has 706 bytes, veclen = 1 > recv 706 bytes from udp/[190.179.3.18]:4375 at 21:51:10.453646: > ------------------------------------------------------------------------ > REGISTER sip:216.75.60.102 SIP/2.0 > From: sip:doddle at 216.75.60.102 ;tag=633f3915 > To: sip:doddle at 216.75.60.102 > Call-Id: 186e708700bcf9a944855105fc3dce0e > Cseq: 102 REGISTER > Expires: 3600 > Date: Wed, 11 Nov 2009 21:51:51 GMT > Max-Forwards: 70 > User-Agent: Doddle WebPhone > Supported: replaces > Authorization: Digest username="doddle", realm="216.75.60.102", > nonce="132239a5-e37e-4698-af61-5df9b3b67de8", uri="sip:216.75.60.102", > response="686ec180c04fc70be22aaca9eb21f5e9", algorithm=MD5, > cnonce="155fb1307a89ffc0eaed1e0e94958e6e", qop=auth, nc=00000033 > Via: SIP/2.0/UDP 192.168.0.1:4375;branch=z9hG4bK-36a1b6039a42;rport > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > tport_deliver(0xae7056b0): msg 0xae716af8 (706 bytes) from udp/ > 190.179.3.18:5060/sip next=(nil) > nta: received REGISTER sip:216.75.60.102 SIP/2.0 (CSeq 102) > nta: Via check: received=190.179.3.18 > nta: canonizing sip:216.75.60.102 with contact > nta: REGISTER (102) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > soa_clone(static::0xae7098a0, 0xae704890, 0xae71ae30) called > soa_set_params(static::0xae71b340, ...) called > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0xae71ae30): sent signal r_respond > nua: nua_handle_destroy: entering > nua: nua_stack_set_params: entering > nua(0xae71ae30): sent signal r_destroy > nua: nua_handle_magic: entering > soa_set_params(static::0xae71b340, ...) called > nua: nua_handle_destroy: entering > tport_tsend(0xae7056b0) tpn = UDP/190.179.3.18:4375 > tport_resolve addrinfo = 190.179.3.18:4375 > tport_by_addrinfo(0xae7056b0): not found by name UDP/190.179.3.18:4375 > tport_vsend returned 645 > send 645 bytes to udp/[190.179.3.18]:4375 at 21:51:10.462570: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.0.1:4375 > ;branch=z9hG4bK-36a1b6039a42;rport=4375;received=190.179.3.18 > From: sip:doddle at 216.75.60.102 ;tag=633f3915 > To: > > >;tag=vHHFetF0N0S2B > > Call-Id: 186e708700bcf9a944855105fc3dce0e > Cseq: 102 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.0.5pre5-15326M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="216.75.60.102", > nonce="5a5ebe31-baf6-429c-a184-f835a22d7c24", stale="true", algorithm=MD5, > qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > nta: sent 401 Unauthorized for REGISTER (102) > nta_leg_destroy((nil)) > soa_destroy(static::0xae71b340) called > > > > Thank you in advacen, > > Federico Omoto From lists at redbonez.net Wed Nov 11 16:20:37 2009 From: lists at redbonez.net (Adam Ford) Date: Wed, 11 Nov 2009 17:20:37 -0700 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> Message-ID: <00eb01ca632d$f719ecf0$e54dc6d0$@net> Hi everybody, I have setup a FreeSWITCH IP-PBX for my office using a T1 and Redfone foneBridge2, which uses Openzap, for my connection to the PSTN. I am trying to figure out if it is possible to forward a call that comes in through the T1/Openzap, back out to a PSTN number. An example would be, I am going to be out of the office and need to forward my office line to my cell phone. I have read what I could find in the wiki about redirect and deflect, but that appears to only deal with SIP providers/gateways. Can I use these applications with Openzap? And if so, what is the syntax for doing so? I have also noted that I can simply bridge the call out another line on the T1 through Openzap. However, that seems to tie up 2 lines just to forward a call. This is not a desirable solution. Thanks to anyone who can help. -Adam From Russell.Mosemann at cune.org Wed Nov 11 17:36:21 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Wed, 11 Nov 2009 19:36:21 -0600 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: <00eb01ca632d$f719ecf0$e54dc6d0$@net> References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> <00eb01ca632d$f719ecf0$e54dc6d0$@net> Message-ID: Adam Ford wrote: > I have also noted that I can simply bridge the call out another line on > the T1 through Openzap. However, that seems to tie up 2 lines just to > forward a call. This is not a desirable solution. That's the way it has to work with any phone system, including your cell phone. If your cell phone provider received a call for you and you had forwarded cell phone calls to another number, your cell phone provider would have to route the incoming call out another line to the next destination. That takes two lines (or channels). That's what forwarding means. One incoming call bridged to one outgoing call. -- Russell Mosemann From lists at redbonez.net Wed Nov 11 17:43:22 2009 From: lists at redbonez.net (Adam Ford) Date: Wed, 11 Nov 2009 18:43:22 -0700 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> <00eb01ca632d$f719ecf0$e54dc6d0$@net> Message-ID: <010401ca6339$860f0f20$922d2d60$@net> Alright. Thank you for your answer. I just had hoped there might be something better that I didn't know about, after reading about deflect on the wiki. -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Russell Mosemann Sent: Wednesday, November 11, 2009 6:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP Adam Ford wrote: > I have also noted that I can simply bridge the call out another line on > the T1 through Openzap. However, that seems to tie up 2 lines just to > forward a call. This is not a desirable solution. That's the way it has to work with any phone system, including your cell phone. If your cell phone provider received a call for you and you had forwarded cell phone calls to another number, your cell phone provider would have to route the incoming call out another line to the next destination. That takes two lines (or channels). That's what forwarding means. One incoming call bridged to one outgoing call. -- Russell Mosemann _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mrene_lists at avgs.ca Wed Nov 11 17:45:49 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 11 Nov 2009 17:45:49 -0800 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: <010401ca6339$860f0f20$922d2d60$@net> References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> <00eb01ca632d$f719ecf0$e54dc6d0$@net> <010401ca6339$860f0f20$922d2d60$@net> Message-ID: <061C000F-5E5D-4C30-875D-860F7162F61D@avgs.ca> There is something called 2B Channel Transfer that can make 2 channels of the same span be released, but providers don't always implement it. Im not exactly sure what kind of Q931 message we need to send down the TDM circuit though. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Nov-09, at 5:43 PM, Adam Ford wrote: > Alright. Thank you for your answer. I just had hoped there might be > something better that I didn't know about, after reading about > deflect on > the wiki. > > -Adam > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Russell > Mosemann > Sent: Wednesday, November 11, 2009 6:36 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Forwarding calls to an outside > number - > OpenZAP > > Adam Ford wrote: > >> I have also noted that I can simply bridge the call out another >> line on >> the T1 through Openzap. However, that seems to tie up 2 lines just to >> forward a call. This is not a desirable solution. > > That's the way it has to work with any phone system, including your > cell > phone. If your cell phone provider received a call for you and you had > forwarded cell phone calls to another number, your cell phone > provider would > have to route the incoming call out another line to the next > destination. > That takes two lines (or channels). That's what forwarding means. One > incoming call bridged to one outgoing call. > > -- > Russell Mosemann > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Nov 11 17:49:48 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Nov 2009 19:49:48 -0600 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: <010401ca6339$860f0f20$922d2d60$@net> References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> <00eb01ca632d$f719ecf0$e54dc6d0$@net> <010401ca6339$860f0f20$922d2d60$@net> Message-ID: deflect would work if the stack and your provider supported TBCT /b On Nov 11, 2009, at 7:43 PM, Adam Ford wrote: > Alright. Thank you for your answer. I just had hoped there might be > something better that I didn't know about, after reading about > deflect on > the wiki. > > -Adam From peter at cindyandpeter.com Wed Nov 11 18:16:59 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Wed, 11 Nov 2009 21:16:59 -0500 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> <00eb01ca632d$f719ecf0$e54dc6d0$@net> <010401ca6339$860f0f20$922d2d60$@net> Message-ID: <029b01ca633e$373828f0$a5a87ad0$@com> FWIW: If you're getting your T1 from Qwest, they have an option called TnR (Transfer and Release). You play a DTMF sequence followed by the destination number. They take the call back and bridge it to the new number. Of course, you pay to have the feature enabled and you pay per use. It's only SLIGHTLY cheaper than using two lines. Peter -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, November 11, 2009 8:50 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP deflect would work if the stack and your provider supported TBCT /b On Nov 11, 2009, at 7:43 PM, Adam Ford wrote: > Alright. Thank you for your answer. I just had hoped there might be > something better that I didn't know about, after reading about > deflect on > the wiki. > > -Adam _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dome at tel.co.th Wed Nov 11 18:28:52 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 12 Nov 2009 09:28:52 +0700 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <87f2f3b90911111431y7b68856cwfe8a97ec0754f86d@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> <87f2f3b90911111244r130945b7w7813c71fbbd12738@mail.gmail.com> <87f2f3b90911111431y7b68856cwfe8a97ec0754f86d@mail.gmail.com> Message-ID: <8ccbff060911111828q1aa92ff2kf279235983f25275@mail.gmail.com> Wow. we can use FS for sip dispatcher :) How to forward call in FS ? i mean 302 redirect not bridge ? BG Dome C. 2009/11/12 Michael Collins : > FYI, I added some docs here: > http://wiki.freeswitch.org/wiki/Mod_distributor > > Please feel free to add to it if you are doing anything interesting or > creative that hasn't been covered. > -MC > > On Wed, Nov 11, 2009 at 12:44 PM, Michael Collins > wrote: >> >> Perfect. I'll have it documented by the end of the day. >> -MC >> >> On Wed, Nov 11, 2009 at 10:32 AM, Anthony Minessale >> wrote: >>> >>> see conf/autoload_configs/distributor.conf.xml >>> >>> >>> ? >>> ??? >>> ??? >>> ??? >>> ????? >>> ????? >>> ??? >>> ? >>> >>> >>> >>> in your dialplan you can use >>> >>> ${distributor(test)} which will cycle expanding to foo1 1/10 times and >>> foo2 the other 9 >>> >>> so imagine if foo1 or foo2 were the names of gateways, or hosts of a >>> remote box >>> >>> basic jist is to set total-weight to a number of arbitrary units then set >>> several nodes with weight elements that add up to that number to break down >>> how many times that node text should be returned out of the total. >>> >>> Remember to use the most simplified reduced value for your fractions to >>> get the most variety. >>> >>> Setting total weight to 1000 and then 2 nodes with 100 and 900 would >>> result in foo1 100 times in a row, then foo2 900 times in a row. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dome at tel.co.th Wed Nov 11 18:42:42 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 12 Nov 2009 09:42:42 +0700 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <8ccbff060911111828q1aa92ff2kf279235983f25275@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> <87f2f3b90911111244r130945b7w7813c71fbbd12738@mail.gmail.com> <87f2f3b90911111431y7b68856cwfe8a97ec0754f86d@mail.gmail.com> <8ccbff060911111828q1aa92ff2kf279235983f25275@mail.gmail.com> Message-ID: <8ccbff060911111842g538e1d2fxe6177e4f2d56c8aa@mail.gmail.com> Got it from wiki Dome C. 2009/11/12 Dome Charoenyost : > Wow. we can use FS for sip dispatcher :) > How to forward call in FS ? i mean 302 redirect not bridge ? > > > BG > > Dome C. > > > 2009/11/12 Michael Collins : >> FYI, I added some docs here: >> http://wiki.freeswitch.org/wiki/Mod_distributor >> >> Please feel free to add to it if you are doing anything interesting or >> creative that hasn't been covered. >> -MC >> >> On Wed, Nov 11, 2009 at 12:44 PM, Michael Collins >> wrote: >>> >>> Perfect. I'll have it documented by the end of the day. >>> -MC >>> >>> On Wed, Nov 11, 2009 at 10:32 AM, Anthony Minessale >>> wrote: >>>> >>>> see conf/autoload_configs/distributor.conf.xml >>>> >>>> >>>> ? >>>> ??? >>>> ??? >>>> ??? >>>> ????? >>>> ????? >>>> ??? >>>> ? >>>> >>>> >>>> >>>> in your dialplan you can use >>>> >>>> ${distributor(test)} which will cycle expanding to foo1 1/10 times and >>>> foo2 the other 9 >>>> >>>> so imagine if foo1 or foo2 were the names of gateways, or hosts of a >>>> remote box >>>> >>>> basic jist is to set total-weight to a number of arbitrary units then set >>>> several nodes with weight elements that add up to that number to break down >>>> how many times that node text should be returned out of the total. >>>> >>>> Remember to use the most simplified reduced value for your fractions to >>>> get the most variety. >>>> >>>> Setting total weight to 1000 and then 2 nodes with 100 and 900 would >>>> result in foo1 100 times in a row, then foo2 900 times in a row. >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From anthony.minessale at gmail.com Wed Nov 11 19:14:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Nov 2009 21:14:23 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> Message-ID: <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> dont execute bridge that way, your bridge itself is the other thing already broadcasting. api uuid_transfer bridge:sofia/myprofile/foo at bar.com inline if you want to do more after the bridge set the variable park_after_bridge=true to make it go back to idle On Wed, Nov 11, 2009 at 4:33 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64. > > On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner > wrote: > > From the trace: > > > ..snip.. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/20585d2c/attachment-0001.html From anthony.minessale at gmail.com Wed Nov 11 20:11:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Nov 2009 22:11:59 -0600 Subject: [Freeswitch-users] att_xfer and Loopback In-Reply-To: <4AFB3A3D.1050602@gmx.net> References: <4AFB3A3D.1050602@gmx.net> Message-ID: <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> set/export the channel variable loopback_bowout=true so it's on the loopback leg On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX wrote: > Hello, > > I have some problems with attended transfer and loopback > > Scenario how id work > - A calls B > - B enters *4 gets an announcement and enter digits for C (A get MOH) > - C is called > - As soon as C picks up the call, A and C are connected and B is dropped > > How it work until here: > - A calls B > - B enters *4 gets an announcement and enter digits for C (A get MOH) > - C is called > - As soon as C picks up the call, B and C are connected (A still MOH) > > The dial string for C is dynamic and dependent on certain parameters, > therefore C must be called via Loopback in our scenario. > > > Here are the configs: > In dialplan for calling B: > > > Dialplan for executing the att_xfer: > > > > > > > > > > So this is pretty standard, except the loopback. SVN is 15322. > > Anybody has a solution for this? > > > Best regards > Peter > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/c1ad4625/attachment.html From anthony.minessale at gmail.com Wed Nov 11 20:12:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Nov 2009 22:12:53 -0600 Subject: [Freeswitch-users] att_xfer and Loopback In-Reply-To: <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> References: <4AFB3A3D.1050602@gmx.net> <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> Message-ID: <191c3a030911112012i63000f3j9867308057c5f318@mail.gmail.com> hit send too soon you want to set loopback_bowout=false This keeps loopback from trying to destroy itself when it sees a chance to cut out of the call path. On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > > set/export the channel variable loopback_bowout=true so it's on the > loopback leg > > > > > On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX wrote: > >> Hello, >> >> I have some problems with attended transfer and loopback >> >> Scenario how id work >> - A calls B >> - B enters *4 gets an announcement and enter digits for C (A get MOH) >> - C is called >> - As soon as C picks up the call, A and C are connected and B is dropped >> >> How it work until here: >> - A calls B >> - B enters *4 gets an announcement and enter digits for C (A get MOH) >> - C is called >> - As soon as C picks up the call, B and C are connected (A still MOH) >> >> The dial string for C is dynamic and dependent on certain parameters, >> therefore C must be called via Loopback in our scenario. >> >> >> Here are the configs: >> In dialplan for calling B: >> >> >> Dialplan for executing the att_xfer: >> >> >> >> >> >> >> >> >> >> So this is pretty standard, except the loopback. SVN is 15322. >> >> Anybody has a solution for this? >> >> >> Best regards >> Peter >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/b913d75a/attachment.html From lei.tlfly at gmail.com Wed Nov 11 22:09:57 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Thu, 12 Nov 2009 14:09:57 +0800 Subject: [Freeswitch-users] How to test FS rtp packet lost rate? In-Reply-To: <2d9149cd0911110849u74c2d8d1jb90de8c20cacde9a@mail.gmail.com> References: <50c41b4e0911110543m7b5431ecu173d8386073fdb32@mail.gmail.com> <2d9149cd0911110849u74c2d8d1jb90de8c20cacde9a@mail.gmail.com> Message-ID: <50c41b4e0911112209n24011465v5fb7e8300462a58d@mail.gmail.com> Hi, thanks Kristian for your answer, it make sense 2009/11/12 Kristian Kielhofner > The simplest way I know of is to bring up another call from a local > phone and listen to the audio. At the same time run tcpdump/etc with > a strict filter to capture the rtp to/from that phone. You can then > run RTP stream analysis and the like in Wireshark to identify any lost > packets. While this obviously won't identify any/all potential lost > packets it will be a lot more practical than any of the alternatives: > > - Capturing all media streams for RTP analysis > - Implementing RTCP to identify lost packets > - Commercial hardware/software > > If FreeSWITCH, your machine, or your network are pushed to the max and > falling apart you're most likely going to see audio problems on your > single (captured) call. > > On Wed, Nov 11, 2009 at 8:43 AM, Lei Tang wrote: > > Hi all, I'm testing a FS server using sipp, I found that sipp only show > the > > retrans of sip packet, Does someone known is there a tool to test FS rtp > > packet lost rate in high concurrent call env? > > > > -- > > Lei.Tang > > lei.tlfly at gmail.com > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/6403a1e2/attachment.html From Prometheus001 at gmx.net Thu Nov 12 00:38:36 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 12 Nov 2009 09:38:36 +0100 Subject: [Freeswitch-users] att_xfer and Loopback In-Reply-To: <191c3a030911112012i63000f3j9867308057c5f318@mail.gmail.com> References: <4AFB3A3D.1050602@gmx.net> <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> <191c3a030911112012i63000f3j9867308057c5f318@mail.gmail.com> Message-ID: <4AFBC98C.4070602@gmx.net> Thanks Anthony, however this rather deteriorated the situation. Now it works the following - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C both have no voice (and B is dropped) - When A hangs up, C hangs up Before it did: - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped - When A hangs up, C hangs up Best regards Peter Anthony Minessale schrieb: > hit send too soon > you want to set loopback_bowout=false > > This keeps loopback from trying to destroy itself when it sees a > chance to cut out of the call path. > > > On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale > > wrote: > > > set/export the channel variable loopback_bowout=true so it's on > the loopback leg > > > > > On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX > > wrote: > > Hello, > > I have some problems with attended transfer and loopback > > Scenario how id work > - A calls B > - B enters *4 gets an announcement and enter digits for C (A > get MOH) > - C is called > - As soon as C picks up the call, A and C are connected and B > is dropped > > How it work until here: > - A calls B > - B enters *4 gets an announcement and enter digits for C (A > get MOH) > - C is called > - As soon as C picks up the call, B and C are connected (A > still MOH) > > The dial string for C is dynamic and dependent on certain > parameters, > therefore C must be called via Loopback in our scenario. > > > Here are the configs: > In dialplan for calling B: > > > Dialplan for executing the att_xfer: > > expression="^attended_xfer$"> > > > > data="loopback/${attxfer_callthis}"/> > > > > So this is pretty standard, except the loopback. SVN is 15322. > > Anybody has a solution for this? > > > Best regards > Peter > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dschwartz at xconnect.net Thu Nov 12 02:03:09 2009 From: dschwartz at xconnect.net (David Schwartz) Date: Thu, 12 Nov 2009 12:03:09 +0200 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? Message-ID: <6EA53FAD386F9D46B97D49BFE148D51406359AED@ISR-JLM-MAIL1.xconnect.co.il> All of the example I see allow me to call FROM gtalk. Help? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/89985963/attachment.html From mcampbellsmith at gmail.com Thu Nov 12 04:08:39 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 12 Nov 2009 23:08:39 +1100 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? In-Reply-To: <6EA53FAD386F9D46B97D49BFE148D51406359AED@ISR-JLM-MAIL1.xconnect.co.il> References: <6EA53FAD386F9D46B97D49BFE148D51406359AED@ISR-JLM-MAIL1.xconnect.co.il> Message-ID: <33c87fa30911120408v2d081e79ja50d2799a594ce91@mail.gmail.com> Check this page out... maybe the info should be put on the wiki... http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ On Thu, Nov 12, 2009 at 9:03 PM, David Schwartz wrote: > All of the example I see allow me to call FROM gtalk. > > > > Help? > > > > Thanks, > > > > David > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From piotr_zurek at biprotech.com Thu Nov 12 04:32:26 2009 From: piotr_zurek at biprotech.com (=?UTF-8?B?UGlvdHIgxbt1cmVr?=) Date: Thu, 12 Nov 2009 13:32:26 +0100 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> Message-ID: <4AFC005A.4090200@biprotech.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/3fd76688/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: piotr_zurek.vcf Type: text/x-vcard Size: 414 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/3fd76688/attachment.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3678 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/3fd76688/attachment.bin From codecomplete at free.fr Thu Nov 12 04:50:25 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 12 Nov 2009 04:50:25 -0800 (PST) Subject: [Freeswitch-users] Displaying caller ID on LED? In-Reply-To: References: <26280730.post@talk.nabble.com> <26280912.post@talk.nabble.com> Message-ID: <26318100.post@talk.nabble.com> Mitch Capper wrote: > I did something like this recently. Thanks for the feedback. I'll see how Linux can be made to send stuff to a USB display. -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26318100.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Nov 12 04:51:45 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 12 Nov 2009 04:51:45 -0800 (PST) Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <87f2f3b90911100932i19c7c971y5fae90f6bb9f4dc0@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> <26284109.post@talk.nabble.com> <87f2f3b90911100932i19c7c971y5fae90f6bb9f4dc0@mail.gmail.com> Message-ID: <26318115.post@talk.nabble.com> mercutioviz wrote: > I believe that French and Spanish sounds are in the works by the > community. > The only other sounds I'm aware of are the Russian ones. Thanks for the tip. -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26318115.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Nov 12 04:59:18 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 12 Nov 2009 04:59:18 -0800 (PST) Subject: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) In-Reply-To: <9E5323D6B69B489384D2E89358CC5EC5@bp1.ad.bp.com> References: <9E5323D6B69B489384D2E89358CC5EC5@bp1.ad.bp.com> Message-ID: <26318213.post@talk.nabble.com> Dave Stevenson-4 wrote: > Has anyone had similar problems with the SPA3102 or has any ideas where I > can look to get to the bottom of the problem. (I have just upgraded the > SPA3102 to the latest 5.1.0 firmware) Before investigating further, you might want to ask in those forums to check that it's not an 3102-related issue instead of Freeswitch: http://forum.voxilla.com/linksys-sipura-voip-support-forum/ http://forums.whirlpool.net.au/forum/107?&g=100 -- View this message in context: http://old.nabble.com/SPA3102-Won%27t-drop-the-PSTN-line-%28UK%29-tp26286696p26318213.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dschwartz at xconnect.net Thu Nov 12 05:02:33 2009 From: dschwartz at xconnect.net (David Schwartz) Date: Thu, 12 Nov 2009 15:02:33 +0200 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? In-Reply-To: <33c87fa30911120408v2d081e79ja50d2799a594ce91@mail.gmail.com> Message-ID: <6EA53FAD386F9D46B97D49BFE148D51406359B4D@ISR-JLM-MAIL1.xconnect.co.il> Thanks Mark I read this and didn't find a dialplan (do I need one?) to make calls into gtalk? I mean how would I even dial in? via URI (e.g. someone at gmail.com)? Wouldn't that just send the call to gmail? What I am looking for is hard coding a number (e.g. 1010) that would enable me to call it and have it convert 1010 to someone at gmail.com who is NOT logged into FS and have the call goto gtalk via some other user (e.g. me at gmail.com) who IS logged into FS. Do you think this is possible? Thanks, David -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Campbell-Smith Sent: Thursday, November 12, 2009 2:09 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? Check this page out... maybe the info should be put on the wiki... http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ On Thu, Nov 12, 2009 at 9:03 PM, David Schwartz wrote: > All of the example I see allow me to call FROM gtalk. > > > > Help? > > > > Thanks, > > > > David > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From stevendt at primrosebank.net Thu Nov 12 05:32:31 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 12 Nov 2009 13:32:31 -0000 Subject: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) References: <9E5323D6B69B489384D2E89358CC5EC5@bp1.ad.bp.com> <26318213.post@talk.nabble.com> Message-ID: <135A6D0AA4AC476A841AF47564917926@bp1.ad.bp.com> Thanks for the pointers - I'll head off there now...... regards Dave ----- Original Message ----- From: "Fred-145" To: Sent: Thursday, November 12, 2009 12:59 PM Subject: Re: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) > > > Dave Stevenson-4 wrote: >> Has anyone had similar problems with the SPA3102 or has any ideas where I >> can look to get to the bottom of the problem. (I have just upgraded the >> SPA3102 to the latest 5.1.0 firmware) > > Before investigating further, you might want to ask in those forums to > check > that it's not an 3102-related issue instead of Freeswitch: > > http://forum.voxilla.com/linksys-sipura-voip-support-forum/ > http://forums.whirlpool.net.au/forum/107?&g=100 > > > -- > View this message in context: > http://old.nabble.com/SPA3102-Won%27t-drop-the-PSTN-line-%28UK%29-tp26286696p26318213.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From testeador01 at gmail.com Thu Nov 12 05:35:52 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 12 Nov 2009 08:35:52 -0500 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? In-Reply-To: <6EA53FAD386F9D46B97D49BFE148D51406359B4D@ISR-JLM-MAIL1.xconnect.co.il> References: <33c87fa30911120408v2d081e79ja50d2799a594ce91@mail.gmail.com> <6EA53FAD386F9D46B97D49BFE148D51406359B4D@ISR-JLM-MAIL1.xconnect.co.il> Message-ID: Hello, Obviously it is possible, next time try to search better, the answer is on the same blog Mark pointed you too: http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-jingle-all-the-way/ Good luck 2009/11/12 David Schwartz > Thanks Mark > > I read this and didn't find a dialplan (do I need one?) to make calls into > gtalk? I mean how would I even dial in? via URI (e.g. someone at gmail.com)? > Wouldn't that just send the call to gmail? > > What I am looking for is hard coding a number (e.g. 1010) that would enable > me to call it and have it convert 1010 to someone at gmail.com who is NOT > logged into FS and have the call goto gtalk via some other user (e.g. > me at gmail.com) who IS logged into FS. > > Do you think this is possible? > > Thanks, > > David > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark > Campbell-Smith > Sent: Thursday, November 12, 2009 2:09 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? > > Check this page out... maybe the info should be put on the wiki... > > http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ > > > On Thu, Nov 12, 2009 at 9:03 PM, David Schwartz > wrote: > > All of the example I see allow me to call FROM gtalk. > > > > > > > > Help? > > > > > > > > Thanks, > > > > > > > > David > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/cbd3bd39/attachment.html From dschwartz at xconnect.net Thu Nov 12 05:53:24 2009 From: dschwartz at xconnect.net (David Schwartz) Date: Thu, 12 Nov 2009 15:53:24 +0200 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? In-Reply-To: Message-ID: <6EA53FAD386F9D46B97D49BFE148D51406359B65@ISR-JLM-MAIL1.xconnect.co.il> Thanks I overlooked that :) D. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena Sent: Thursday, November 12, 2009 3:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? Hello, Obviously it is possible, next time try to search better, the answer is on the same blog Mark pointed you too: http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-jingle-all-the-way/ Good luck 2009/11/12 David Schwartz > Thanks Mark I read this and didn't find a dialplan (do I need one?) to make calls into gtalk? I mean how would I even dial in? via URI (e.g. someone at gmail.com)? Wouldn't that just send the call to gmail? What I am looking for is hard coding a number (e.g. 1010) that would enable me to call it and have it convert 1010 to someone at gmail.com who is NOT logged into FS and have the call goto gtalk via some other user (e.g. me at gmail.com) who IS logged into FS. Do you think this is possible? Thanks, David -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Campbell-Smith Sent: Thursday, November 12, 2009 2:09 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? Check this page out... maybe the info should be put on the wiki... http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ On Thu, Nov 12, 2009 at 9:03 PM, David Schwartz > wrote: > All of the example I see allow me to call FROM gtalk. > > > > Help? > > > > Thanks, > > > > David > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/cc901004/attachment.html From brian at freeswitch.org Thu Nov 12 05:59:29 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Nov 2009 07:59:29 -0600 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? In-Reply-To: <6EA53FAD386F9D46B97D49BFE148D51406359B4D@ISR-JLM-MAIL1.xconnect.co.il> References: <6EA53FAD386F9D46B97D49BFE148D51406359B4D@ISR-JLM-MAIL1.xconnect.co.il> Message-ID: This is just basic freeswitch dialplan concepts. It has nothing to do specifically with gtalk. Seems like you need to step back and do some more reading on the dialplan. ;) /b On Nov 12, 2009, at 7:02 AM, David Schwartz wrote: > What I am looking for is hard coding a number (e.g. 1010) that would > enable me to call it and have it convert 1010 to someone at gmail.com > who is NOT logged into FS and have the call goto gtalk via some > other user (e.g. me at gmail.com) who IS logged into FS. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/b4fbe386/attachment.html From peder at networkoblivion.com Thu Nov 12 06:25:48 2009 From: peder at networkoblivion.com (Peder) Date: Thu, 12 Nov 2009 08:25:48 -0600 Subject: [Freeswitch-users] Cisco 79x1 & Presence Message-ID: <034a01ca63a4$07c13790$1743a6b0$@com> Has anybody every figured out how to get presence working on a Cisco 79x1 w/ FreeSWITCH? I spent quite a bit of time 6+ months ago on it and could never get it to work. Peder From lei.tlfly at gmail.com Thu Nov 12 07:01:02 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Thu, 12 Nov 2009 23:01:02 +0800 Subject: [Freeswitch-users] hangup incoming call by Reason: Q.850; cause=1; text="Unallocated (unassigned) number" Message-ID: <50c41b4e0911120701r737ce492j5bf5f5be2fd15550@mail.gmail.com> Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal sip endpoint of FS. I added two dialplan in public dialplan xml file. as flow: Every thing is ok when call to number 88888. but when I call the second number "*114", fs hangup after accept and answer the call, I captured the sip packets and found FS sent a bye packet after answer the call. the cause is "Reason: Q.850;cause=1;text="Unallocated (unassigned) number"". But as the fs console log show, the call is answered and the correct ivr script is runned. Why FS hangup the call? Does somebody have any idea about this problem? ============sip packets=================== ********invite msg from softswitch INVITE sip:*114 at 10.37.143.6:5060;user=phone SIP/2.0 Contact: Content-Type: application/sdp To: From: xxxxxxxxx;tag=949132463135364198E42500 P-Asserted-Identity: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE Supported: 100rel,timer,replaces,diversion Expires: 155 Session-Expires: 1800 Min-SE: 90 Call-ID: 01FD10D1BD81400000010690 at sip-3 Max-Forwards: 70 CSeq: 1 INVITE Timestamp: 58520 Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK5C0F524645A70C943998751419749696 Content-Length: 150 v=0 o=- 54000602557 1258015146 IN IP4 10.4.35.59 s=SDP Data c=IN IP4 10.4.35.59 t=0 0 m=audio 30000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 ******FS ack SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK5C0F524645A70C943998751419749696 From: xxxxxxxxx ;tag=949132463135364198E42500 To: Call-ID: 01FD10D1BD81400000010690 at sip-3 CSeq: 1 INVITE Timestamp: 58520 0.000000 User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Content-Length: 0 *****FS answer the call (in lua script, I called session:answer() ) SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK5C0F524645A70C943998751419749696 From: xxxxxxxxx ;tag=949132463135364198E42500 To: ;tag=UjZcZUKZXjHcQ Call-ID: 01FD10D1BD81400000010690 at sip-3 CSeq: 1 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, refer Session-Expires: 1800;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 245 v=0 o=FreeSWITCH 1257988835 1257988836 IN IP4 10.37.143.6 s=FreeSWITCH c=IN IP4 10.37.143.6 t=0 0 m=audio 24890 RTP/AVP 8 120 a=rtpmap:8 PCMA/8000 a=rtpmap:120 telephone-event/8000 a=fmtp:120 0-16 a=silenceSupp:off - - - - a=ptime:20 ACK sip:*114 at 10.37.143.6:5060;transport=udp SIP/2.0 CSeq: 1 ACK To: ;tag=UjZcZUKZXjHcQ From: xxxxxxxxx;tag=949132463135364198E42500 Call-ID: 01FD10D1BD81400000010690 at sip-3 Max-Forwards: 70 Timestamp: 58520 Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK0CC4AE6EE59CA15F69429CDB97848C21 Content-Length: 0 *******FS hangup the call BYE sip:*114 at 10.37.143.6:5060;transport=udp SIP/2.0 Reason: Q.850;cause=1;text="Unallocated (unassigned) number" To: ;tag=UjZcZUKZXjHcQ From: xxxxxxxxx;tag=949132463135364198E42500 Call-ID: 01FD10D1BD81400000010690 at sip-3 Max-Forwards: 70 CSeq: 2 BYE Timestamp: 58521 Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bKBE2D7D86B44CA171A5D374ECAA99A1DB Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/27853ad9/attachment.html From codecomplete at free.fr Thu Nov 12 07:38:30 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 12 Nov 2009 07:38:30 -0800 (PST) Subject: [Freeswitch-users] Does OpenZap support CTR21? In-Reply-To: <20091106200458.AACDC3E5BEF@mail.cune.org> References: <26217371.post@talk.nabble.com> <7FD19B47-C121-48CD-98C2-2830BFDF1068@jerris.com> <26230864.post@talk.nabble.com> <20091106200458.AACDC3E5BEF@mail.cune.org> Message-ID: <26320837.post@talk.nabble.com> Russell.Mosemann wrote: > Yes, it should just work. I'd recommend Dahdi (complete), because Zaptel > is not being developed anymore. Thanks for the links. Turns out this card seems incompatible with the motherboard I have, so I'll concentrate on the Linksys 3102 instead. -- View this message in context: http://old.nabble.com/Does-OpenZap-support-CTR21--tp26217371p26320837.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Thu Nov 12 07:50:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Nov 2009 09:50:49 -0600 Subject: [Freeswitch-users] att_xfer and Loopback In-Reply-To: <4AFBC98C.4070602@gmx.net> References: <4AFB3A3D.1050602@gmx.net> <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> <191c3a030911112012i63000f3j9867308057c5f318@mail.gmail.com> <4AFBC98C.4070602@gmx.net> Message-ID: <191c3a030911120750p34d27d44u2fec4015caf2f367@mail.gmail.com> if you provide a console trace of both situations with console loglevel debug and put them on pastebin i can tell you what's happening. On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX wrote: > Thanks Anthony, > > however this rather deteriorated the situation. > Now it works the following > - A calls B > - B enters *4 gets an announcement and enters digits for C (A get MOH) > - C is called > - As soon as C picks up the call, A and C both have no voice (and B is > dropped) > - When A hangs up, C hangs up > > Before it did: > - A calls B > - B enters *4 gets an announcement and enters digits for C (A get MOH) > - C is called > - As soon as C picks up the call, A and C are connected and B is dropped > - When A hangs up, C hangs up > > Best regards > Peter > > Anthony Minessale schrieb: > > hit send too soon > > you want to set loopback_bowout=false > > > > This keeps loopback from trying to destroy itself when it sees a > > chance to cut out of the call path. > > > > > > On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale > > > > wrote: > > > > > > set/export the channel variable loopback_bowout=true so it's on > > the loopback leg > > > > > > > > > > On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX > > > wrote: > > > > Hello, > > > > I have some problems with attended transfer and loopback > > > > Scenario how id work > > - A calls B > > - B enters *4 gets an announcement and enter digits for C (A > > get MOH) > > - C is called > > - As soon as C picks up the call, A and C are connected and B > > is dropped > > > > How it work until here: > > - A calls B > > - B enters *4 gets an announcement and enter digits for C (A > > get MOH) > > - C is called > > - As soon as C picks up the call, B and C are connected (A > > still MOH) > > > > The dial string for C is dynamic and dependent on certain > > parameters, > > therefore C must be called via Loopback in our scenario. > > > > > > Here are the configs: > > In dialplan for calling B: > > > > > > Dialplan for executing the att_xfer: > > > > > expression="^attended_xfer$"> > > > > > > > > > data="loopback/${attxfer_callthis}"/> > > > > > > > > So this is pretty standard, except the loopback. SVN is 15322. > > > > Anybody has a solution for this? > > > > > > Best regards > > Peter > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/edb5436c/attachment-0001.html From mike at jerris.com Thu Nov 12 08:09:36 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 12 Nov 2009 11:09:36 -0500 Subject: [Freeswitch-users] hangup incoming call by Reason: Q.850; cause=1; text="Unallocated (unassigned) number" In-Reply-To: <50c41b4e0911120701r737ce492j5bf5f5be2fd15550@mail.gmail.com> References: <50c41b4e0911120701r737ce492j5bf5f5be2fd15550@mail.gmail.com> Message-ID: Take a look at the freeswitch debug log, it should tell you exactly why it hung up. Mike On Nov 12, 2009, at 10:01 AM, Lei Tang wrote: > Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal sip endpoint of FS. > I added two dialplan in public dialplan xml file. as flow: > > > > > > > > > > > > > Every thing is ok when call to number 88888. but when I call the second number "*114", fs hangup after accept and answer the call, I captured the sip packets and found FS sent a bye packet after answer the call. the cause is "Reason: Q.850;cause=1;text="Unallocated (unassigned) number"". But as the fs console log show, the call is answered and the correct ivr script is runned. Why FS hangup the call? Does somebody have any idea about this problem? > > > ============sip packets=================== > ********invite msg from softswitch > INVITE sip:*114 at 10.37.143.6:5060;user=phone SIP/2.0 > Contact: > Content-Type: application/sdp > To: > From: xxxxxxxxx;tag=949132463135364198E42500 > P-Asserted-Identity: > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE > Supported: 100rel,timer,replaces,diversion > Expires: 155 > Session-Expires: 1800 > Min-SE: 90 > Call-ID: 01FD10D1BD81400000010690 at sip-3 > Max-Forwards: 70 > CSeq: 1 INVITE > Timestamp: 58520 > Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 > Content-Length: 150 > > v=0 > o=- 54000602557 1258015146 IN IP4 10.4.35.59 > s=SDP Data > c=IN IP4 10.4.35.59 > t=0 0 > m=audio 30000 RTP/AVP 8 > a=rtpmap:8 PCMA/8000 > a=ptime:20 > > > ******FS ack > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 > From: xxxxxxxxx ;tag=949132463135364198E42500 > To: > Call-ID: 01FD10D1BD81400000010690 at sip-3 > CSeq: 1 INVITE > Timestamp: 58520 0.000000 > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 > Content-Length: 0 > > *****FS answer the call (in lua script, I called session:answer() ) > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 > From: xxxxxxxxx ;tag=949132463135364198E42500 > To: ;tag=UjZcZUKZXjHcQ > Call-ID: 01FD10D1BD81400000010690 at sip-3 > CSeq: 1 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Require: timer > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Session-Expires: 1800;refresher=uac > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 245 > > v=0 > o=FreeSWITCH 1257988835 1257988836 IN IP4 10.37.143.6 > s=FreeSWITCH > c=IN IP4 10.37.143.6 > t=0 0 > m=audio 24890 RTP/AVP 8 120 > a=rtpmap:8 PCMA/8000 > a=rtpmap:120 telephone-event/8000 > a=fmtp:120 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:*114 at 10.37.143.6:5060;transport=udp SIP/2.0 > CSeq: 1 ACK > To: ;tag=UjZcZUKZXjHcQ > From: xxxxxxxxx;tag=949132463135364198E42500 > Call-ID: 01FD10D1BD81400000010690 at sip-3 > Max-Forwards: 70 > Timestamp: 58520 > Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK0CC4AE6EE59CA15F69429CDB97848C21 > Content-Length: 0 > > *******FS hangup the call > BYE sip:*114 at 10.37.143.6:5060;transport=udp SIP/2.0 > Reason: Q.850;cause=1;text="Unallocated (unassigned) number" > To: ;tag=UjZcZUKZXjHcQ > From: xxxxxxxxx;tag=949132463135364198E42500 > Call-ID: 01FD10D1BD81400000010690 at sip-3 > Max-Forwards: 70 > CSeq: 2 BYE > Timestamp: 58521 > Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bKBE2D7D86B44CA171A5D374ECAA99A1DB > Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/8029d908/attachment.html From lists at redbonez.net Thu Nov 12 09:41:37 2009 From: lists at redbonez.net (Adam Ford) Date: Thu, 12 Nov 2009 10:41:37 -0700 Subject: [Freeswitch-users] Polycom SoundPoint IP501 In-Reply-To: <034a01ca63a4$07c13790$1743a6b0$@com> References: <034a01ca63a4$07c13790$1743a6b0$@com> Message-ID: <012901ca63bf$64152090$2c3f61b0$@net> Has anyone used a Polycom SoundPoint IP501 or similar hard phone with FreeSWITCH? I configured one to register with my FreeSWITCH server using one of the default sip profiles to test and I get "[DEBUG] sofia_reg.c:1688 SIP username 1001 does not match auth username" in the log file and the phone doesn't register. I have confirmed that the auth username and the display name are both 1001. Is there some additional configuration on the FreeSWITCH side to get these phones to register? Thanks for any help you can offer, -Adam From brian at freeswitch.org Thu Nov 12 09:50:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Nov 2009 11:50:28 -0600 Subject: [Freeswitch-users] Polycom SoundPoint IP501 In-Reply-To: <012901ca63bf$64152090$2c3f61b0$@net> References: <034a01ca63a4$07c13790$1743a6b0$@com> <012901ca63bf$64152090$2c3f61b0$@net> Message-ID: <50AD10F6-C050-40F7-A235-93E40213273D@freeswitch.org> Not sure what do you have in your config file for the polycom exactly? btw you hijacked the Cisco Presence thread by clicking reply.. and changing the subject please don't do that in the future. Click new message and input the address for the list. Thanks, Brian On Nov 12, 2009, at 11:41 AM, Adam Ford wrote: > Has anyone used a Polycom SoundPoint IP501 or similar hard phone with > FreeSWITCH? I configured one to register with my FreeSWITCH server > using one > of the default sip profiles to test and I get "[DEBUG] sofia_reg.c: > 1688 SIP > username 1001 does not match auth username" in the log file and the > phone > doesn't register. I have confirmed that the auth username and the > display > name are both 1001. Is there some additional configuration on the > FreeSWITCH > side to get these phones to register? > > Thanks for any help you can offer, > -Adam > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Thu Nov 12 09:50:49 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Nov 2009 11:50:49 -0600 Subject: [Freeswitch-users] Cisco 79x1 & Presence In-Reply-To: <034a01ca63a4$07c13790$1743a6b0$@com> References: <034a01ca63a4$07c13790$1743a6b0$@com> Message-ID: <859F9AF7-01A9-4621-B902-393E0B93DCDC@freeswitch.org> They do it in their own weird way... if you wanna track it down I know their are examples of it out there. /b On Nov 12, 2009, at 8:25 AM, Peder wrote: > Has anybody every figured out how to get presence working on a Cisco > 79x1 w/ > FreeSWITCH? I spent quite a bit of time 6+ months ago on it and > could never > get it to work. > > Peder > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From tina at a2unlimited.com Thu Nov 12 09:57:34 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Thu, 12 Nov 2009 12:57:34 -0500 Subject: [Freeswitch-users] Calls per second on FreeSWITCH Message-ID: <67387ff53247696360986ff66f5dc894.squirrel@emailmg.ipower.com> I'm trying to increase the number of calls per second that I can originate from FreeSWITCH, but I cannot seem to get more than two-per-second. (I am trying to use FS to initiate thousands of calls quickly) switch.conf.xml I beefed up the max-sessions and sessions-per-second in the switch.conf.xml file, but that did not seem to make any difference. Any thoughts? - Tina From mattdfong at gmail.com Thu Nov 12 10:17:12 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 13 Nov 2009 01:17:12 +0700 Subject: [Freeswitch-users] Calls per second on FreeSWITCH In-Reply-To: <67387ff53247696360986ff66f5dc894.squirrel@emailmg.ipower.com> References: <67387ff53247696360986ff66f5dc894.squirrel@emailmg.ipower.com> Message-ID: <4256bf830911121017u7e755453o987cd55359b21928@mail.gmail.com> Tina, How are you originating the calls? from the console? Try bgapi originate... --matt Voice Broadcasting - http://www.hellohunter.com/voice_blast.php On Fri, Nov 13, 2009 at 12:57 AM, wrote: > I'm trying to increase the number of calls per second that I can originate > from FreeSWITCH, but I cannot seem to get more than two-per-second. > > (I am trying to use FS to initiate thousands of calls quickly) > > switch.conf.xml > I beefed up the max-sessions and sessions-per-second in the > switch.conf.xml file, but that did not seem to make any difference. > > Any thoughts? > > - Tina > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/5828eb8e/attachment.html From lfurrea at gmail.com Thu Nov 12 10:38:54 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Thu, 12 Nov 2009 12:38:54 -0600 Subject: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) In-Reply-To: <135A6D0AA4AC476A841AF47564917926@bp1.ad.bp.com> References: <9E5323D6B69B489384D2E89358CC5EC5@bp1.ad.bp.com> <26318213.post@talk.nabble.com> <135A6D0AA4AC476A841AF47564917926@bp1.ad.bp.com> Message-ID: Remember you have a plain old regular analog connection between the FXO port of the SPA and your "phone line". The FXO circuit is just an analog switch (open or closed) if no one answers on the IP side and the person on the PSTN side hangs up, then the FXO side should sense a change in the polarity of voltage, a lack of current for a certain period or at least detect a tone played by your Telco to be able to go on hook and wait for another call. Try to google "disconnect supervision issues" so that you can get a clearer explanation of what you are experiencing. On Thu, Nov 12, 2009 at 7:32 AM, Dave Stevenson wrote: > Thanks for the pointers - I'll head off there now...... > > regards > Dave > > > > ----- Original Message ----- > From: "Fred-145" > To: > Sent: Thursday, November 12, 2009 12:59 PM > Subject: Re: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) > > > > > > > > Dave Stevenson-4 wrote: > >> Has anyone had similar problems with the SPA3102 or has any ideas where > I > >> can look to get to the bottom of the problem. (I have just upgraded the > >> SPA3102 to the latest 5.1.0 firmware) > > > > Before investigating further, you might want to ask in those forums to > > check > > that it's not an 3102-related issue instead of Freeswitch: > > > > http://forum.voxilla.com/linksys-sipura-voip-support-forum/ > > http://forums.whirlpool.net.au/forum/107?&g=100 > > > > > > -- > > View this message in context: > > > http://old.nabble.com/SPA3102-Won%27t-drop-the-PSTN-line-%28UK%29-tp26286696p26318213.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- firma -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/27be958e/attachment-0001.html From tina at a2unlimited.com Thu Nov 12 12:08:03 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Thu, 12 Nov 2009 15:08:03 -0500 Subject: [Freeswitch-users] Calls per second on FreeSWITCH In-Reply-To: References: Message-ID: Matt, Thank you so much! bgapi did the trick. - Tina > Tina, > > How are you originating the calls? from the console? Try bgapi > originate... > > --matt > Voice Broadcasting - http://www.hellohunter.com/voice_blast.php > > On Fri, Nov 13, 2009 at 12:57 AM, wrote: > >> I'm trying to increase the number of calls per second that I can >> originate >> from FreeSWITCH, but I cannot seem to get more than two-per-second. >> >> (I am trying to use FS to initiate thousands of calls quickly) >> >> switch.conf.xml >> I beefed up the max-sessions and sessions-per-second in the >> switch.conf.xml file, but that did not seem to make any difference. >> >> Any thoughts? >> >> - Tina >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From viper at fx-services.com Thu Nov 12 11:14:09 2009 From: viper at fx-services.com (Robin Vleij) Date: Thu, 12 Nov 2009 20:14:09 +0100 Subject: [Freeswitch-users] Large number of destinations Message-ID: <4AFC5E81.9020104@fx-services.com> Hi all, I'm currently building a proof-of-concept box using Freeswitch. Coming from Asterisk/Kamalio/OpenSER it looks very cool so far, very complete. The plan is to make some sort of SIP router, some would call it an SBC I guess. There will be no PBX stuff, just gateways that talk to each other. PSTN Gateways or other operators or systems. If a system is locally connected (say a local voip platform or interconnected partner), traffic to those destinations should be routed directly to that system and not out to PSTN. I'm looking at a potentially large nr of destination nrs or ranges. Not all those destinations are in the local ENUM so I can't use that as a routing system. I'm thinking about mod_lcr, but it seems more suited for eh ... LCR routing, which is not what I want to do here. I just want to define which nrs or nr ranges are "directly" connected, so that when someone calls there from whatever way they come in (I'm running just one instance and thought about defining all gateways/systems as gateways in the SIP profile), they should end up there and not at PSTN. I think I have two ways of doing this: 1. Make a HUGE XML dialplan and use that to fall back to when internal ENUM lookup doesn't give a result back to where a nr is located 2. Use LCR and find out some kind of way to load all of these destinations into a LCR table and use it in the "wrong" way, ie no costs are involved, it should just be a way to know which nrs or ranges are to be sent to which gateway. Nr1 is probably best (anyone experience how many conditions one can have dialplan_xml?), but say that we would exchange traffic with an operator, it would really suck writing an XML dialplan with 5000 number ranges. :) Anyone experience with this or ideas how this can be solved? Since it's a proof-of-concept it's unclear how exactly those customers or systems are looking. They might be 6000 (un)ported individual nrs, or just a few large ranges. /Robin From yehavi.bourvine at gmail.com Thu Nov 12 12:52:40 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 12 Nov 2009 22:52:40 +0200 Subject: [Freeswitch-users] Polycom SoundPoint IP501 In-Reply-To: <50AD10F6-C050-40F7-A235-93E40213273D@freeswitch.org> References: <034a01ca63a4$07c13790$1743a6b0$@com> <012901ca63bf$64152090$2c3f61b0$@net> <50AD10F6-C050-40F7-A235-93E40213273D@freeswitch.org> Message-ID: I am using Polycoms (430 and 501) with FreeSwitch. How do you provision them? Via WEB or config files? If you use config files than I can send you some sample files. Regards, __Yehavi: On Nov 12, 2009, at 11:41 AM, Adam Ford wrote: > > > Has anyone used a Polycom SoundPoint IP501 or similar hard phone with > > FreeSWITCH? I configured one to register with my FreeSWITCH server > > using one > > of the default sip profiles to test and I get "[DEBUG] sofia_reg.c: > > 1688 SIP > > username 1001 does not match auth username" in the log file and the > > phone > > doesn't register. I have confirmed that the auth username and the > > display > > name are both 1001. Is there some additional configuration on the > > FreeSWITCH > > side to get these phones to register? > > > > Thanks for any help you can offer, > > -Adam > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/014e75de/attachment.html From rupa at rupa.com Thu Nov 12 12:59:38 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 12 Nov 2009 14:59:38 -0600 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: <4AFC5E81.9020104@fx-services.com> References: <4AFC5E81.9020104@fx-services.com> Message-ID: Take a look at mod_easyroute. On Thu, Nov 12, 2009 at 1:14 PM, Robin Vleij wrote: > Hi all, > > I'm currently building a proof-of-concept box using Freeswitch. Coming > from Asterisk/Kamalio/OpenSER it looks very cool so far, very complete. > > The plan is to make some sort of SIP router, some would call it an SBC I > guess. There will be no PBX stuff, just gateways that talk to each > other. PSTN Gateways or other operators or systems. > > If a system is locally connected (say a local voip platform or > interconnected partner), traffic to those destinations should be routed > directly to that system and not out to PSTN. I'm looking at a > potentially large nr of destination nrs or ranges. Not all those > destinations are in the local ENUM so I can't use that as a routing system. > > I'm thinking about mod_lcr, but it seems more suited for eh ... LCR > routing, which is not what I want to do here. I just want to define > which nrs or nr ranges are "directly" connected, so that when someone > calls there from whatever way they come in (I'm running just one > instance and thought about defining all gateways/systems as gateways in > the SIP profile), they should end up there and not at PSTN. > > I think I have two ways of doing this: > > 1. Make a HUGE XML dialplan and use that to fall back to when internal > ENUM lookup doesn't give a result back to where a nr is located > 2. Use LCR and find out some kind of way to load all of these > destinations into a LCR table and use it in the "wrong" way, ie no costs > are involved, it should just be a way to know which nrs or ranges are to > be sent to which gateway. > > Nr1 is probably best (anyone experience how many conditions one can have > dialplan_xml?), but say that we would exchange traffic with an operator, > it would really suck writing an XML dialplan with 5000 number ranges. :) > > Anyone experience with this or ideas how this can be solved? Since it's > a proof-of-concept it's unclear how exactly those customers or systems > are looking. They might be 6000 (un)ported individual nrs, or just a few > large ranges. > > /Robin > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From tina at a2unlimited.com Thu Nov 12 13:19:26 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Thu, 12 Nov 2009 16:19:26 -0500 Subject: [Freeswitch-users] CDR for Failed Calls Message-ID: <2d6ab34c4cc1359c15811ed655b8451a.squirrel@emailmg.ipower.com> I am using xml_cdr to generate CDR results from FreeSWITCH servers, and I've noticed that failed call attempts are not showing up in the results. Whereas the failed attempt is showing up in the Master.csv file. For example, I've initiated some outbound calls that show up in the Master.csv as "RECOVERY_ON_TIMER_EXPIRE", but there is not record from the xml_cdr process. Is there a parameter that can be adjusted in xml_cdr.conf.xml that enables the submission of CDR data for failed calls? ------------------ Here is my current xml_cdr.conf.xml configuration: ------------------ From siniypin at gmail.com Thu Nov 12 14:27:02 2009 From: siniypin at gmail.com (RobertT) Date: Fri, 13 Nov 2009 01:27:02 +0300 Subject: [Freeswitch-users] tcp call misses sip message Message-ID: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> Hello everyone! I'v got strange problem with incomplete call via tcp transport. When I perform bridged call from one ua (no matter what transport udp or tcp) through FS this call's leg b message sequence (over tcp) lacks finishing SIP message what in it's turn cause the call to be disconnected by the client by timeout. Everything works fine with local calls, so I guess the problem is somewhere between UA and FS. There is no NAT and calls via udp are being established correctly. The problem is with tcp and tls as well. This is the sender's ua SIP trace: TX 1049 bytes Request msg INVITE/cseq=11615 (tdta0486C000) to UDP : RX 348 bytes Response msg 100/INVITE/cseq=11615 (rdata0482806C) from UDP : RX 813 bytes Response msg 407/INVITE/cseq=11615 (rdata0482806C) from UDP : TX 346 bytes Request msg ACK/cseq=11615 (tdta0486EFD0) to UDP : TX 1324 bytes Request msg INVITE/cseq=11616 (tdta0486C000) to UDP : RX 348 bytes Response msg 100/INVITE/cseq=11616 (rdata0482806C) from UDP : RX 1083 bytes Response msg 200/INVITE/cseq=11616 (rdata0482806C) from UDP : TX 360 bytes Request msg ACK/cseq=11616 (tdta04874E38) to UDP : And this is the reciever's SIP trace: RX 1167 bytes Request msg INVITE/cseq=122911315 (rdata04864E10) from tcp : TX 298 bytes Response msg 100/INVITE/cseq=122911315 (tdta0486D010) to tcp : TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : ------ I guess this is where ACK is supposed to arrive Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=1 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=2 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=3 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : .... Sofia profile config: and super-smart dialplan FS 1.0.5pre5 is running on Windows Server 2007SP1 64bit.This issue first occured with 1.0.4 release. Best regards, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/4a957142/attachment-0001.html From srinivas.ksvreddy at gmail.com Thu Nov 12 14:32:00 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Fri, 13 Nov 2009 04:02:00 +0530 Subject: [Freeswitch-users] mod event socket In-Reply-To: References: Message-ID: HI all, i have connected Freeswtich(mod event socket) through telnet(tcp) 8021 port, when i am trying to connect freeswtich it it taking 20 seconds to get response from FS, can i able to reduce tcp response time? thanks Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/a4d868c3/attachment.html From viper at fx-services.com Thu Nov 12 14:32:33 2009 From: viper at fx-services.com (Robin Vleij) Date: Thu, 12 Nov 2009 23:32:33 +0100 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: References: <4AFC5E81.9020104@fx-services.com> Message-ID: <4AFC8D01.9060401@fx-services.com> On 11/12/09 9:59 PM, Rupa Schomaker wrote: Hi! > Take a look at mod_easyroute. Cool, I remember "quick-reading" about that module and thinking "nah, not needed". Then when the plan changed and I needed the large amount of routes it didn't struck me that easyroute is what I need for what I want to do. Perfect. If I read it right, this is suited for "complete" nrs. So would I have a system connected with lots of DIDs, I would put them in easyroute. Then for systems with lots of number ranges, I would use mod_lcr. My dialplan context where I would handle inbound from anywhere would look like: 1. ENUM lookup to see if it's a ported nr to any directly connected system 2. mod_lcr lookup to see if it's any large nr range directly connected 3. mod_easyroute to see if it's any individual nr directly connected (via some gateway) 4. Give up :) Guess I'll get working on stuff in this order then. Thanks for the tip! /Robin From lists at redbonez.net Thu Nov 12 14:36:15 2009 From: lists at redbonez.net (Adam Ford) Date: Thu, 12 Nov 2009 15:36:15 -0700 Subject: [Freeswitch-users] Polycom SoundPoint IP501 In-Reply-To: References: <034a01ca63a4$07c13790$1743a6b0$@com> <012901ca63bf$64152090$2c3f61b0$@net> <50AD10F6-C050-40F7-A235-93E40213273D@freeswitch.org> Message-ID: <015101ca63e8$8cb47680$a61d6380$@net> I was trying to configure it just on the phone itself, but apparently even though it says Auth. User on the phone setting, it doesn't actually set the auth username according to the web interface. After using the web interface to configure the phone it works now. Thank you for your responses. -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi Bourvine Sent: Thursday, November 12, 2009 1:53 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom SoundPoint IP501 I am using Polycoms (430 and 501) with FreeSwitch. How do you provision them? Via WEB or config files? If you use config files than I can send you some sample files. Regards, __Yehavi: On Nov 12, 2009, at 11:41 AM, Adam Ford wrote: > Has anyone used a Polycom SoundPoint IP501 or similar hard phone with > FreeSWITCH? I configured one to register with my FreeSWITCH server > using one > of the default sip profiles to test and I get "[DEBUG] sofia_reg.c: > 1688 SIP > username 1001 does not match auth username" in the log file and the > phone > doesn't register. I have confirmed that the auth username and the > display > name are both 1001. Is there some additional configuration on the > FreeSWITCH > side to get these phones to register? > > Thanks for any help you can offer, > -Adam > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/2090f293/attachment.html From brian at freeswitch.org Thu Nov 12 14:46:11 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Nov 2009 16:46:11 -0600 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> Message-ID: <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> tack on a ;transport=tcp /b On Nov 12, 2009, at 4:27 PM, RobertT wrote: > From rupa at rupa.com Thu Nov 12 14:53:13 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 12 Nov 2009 16:53:13 -0600 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: <4AFC8D01.9060401@fx-services.com> References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> Message-ID: On Thu, Nov 12, 2009 at 4:32 PM, Robin Vleij wrote: > On 11/12/09 9:59 PM, Rupa Schomaker wrote: > If I read it right, this is suited for "complete" nrs. So would I have a > system connected with lots of DIDs, I would put them in easyroute. Then > for systems with lots of number ranges, I would use mod_lcr. lcr is based on prefix, so the boundaries for which the range is assigned may not match a prefix. You may be better off either: 1) denormalize your ranges and just insert all distinct #s 2) Modify mod_easyroute to support ranges 3) talk to SWK (he is on irc here and there) about his (non free) fancier routing options -- -Rupa From anthony.minessale at gmail.com Thu Nov 12 14:57:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Nov 2009 16:57:35 -0600 Subject: [Freeswitch-users] CDR for Failed Calls In-Reply-To: <2d6ab34c4cc1359c15811ed655b8451a.squirrel@emailmg.ipower.com> References: <2d6ab34c4cc1359c15811ed655b8451a.squirrel@emailmg.ipower.com> Message-ID: <191c3a030911121457k77e83f0alff64bc503b2708fc@mail.gmail.com> enable the b leg logging On Thu, Nov 12, 2009 at 3:19 PM, wrote: > I am using xml_cdr to generate CDR results from FreeSWITCH servers, and > I've noticed that failed call attempts are not showing up in the results. > > Whereas the failed attempt is showing up in the Master.csv file. > For example, I've initiated some outbound calls that show up in the > Master.csv as "RECOVERY_ON_TIMER_EXPIRE", but there is not record from the > xml_cdr process. > > Is there a parameter that can be adjusted in xml_cdr.conf.xml that enables > the submission of CDR data for failed calls? > > ------------------ > Here is my current xml_cdr.conf.xml configuration: > > > > > > > > > > > > > > > ------------------ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/4484890d/attachment-0001.html From msc at freeswitch.org Thu Nov 12 15:29:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Nov 2009 15:29:03 -0800 Subject: [Freeswitch-users] mod event socket In-Reply-To: References: Message-ID: <87f2f3b90911121529ubd6b61dxd9fb27036bba89ca@mail.gmail.com> What exactly are you typing when you connect? Also, which version of FS? -MC On Thu, Nov 12, 2009 at 2:32 PM, srinivasula reddy < srinivas.ksvreddy at gmail.com> wrote: > > > > HI all, > > i have connected Freeswtich(mod event socket) through telnet(tcp) 8021 > port, when i am trying to connect freeswtich it it taking 20 seconds to get > response from FS, > can i able to reduce tcp response time? > > thanks > Srinivasula Reddy K > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/c5ae1fa4/attachment.html From egable+freeswitch at gmail.com Thu Nov 12 17:49:17 2009 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Thu, 12 Nov 2009 20:49:17 -0500 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> Message-ID: Or, of course, there is always mod_xml_curl. Basically, XML dialplan on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a web application server, web application server responds with XML routing response, FreeSWITCH routes the call. On Thu, Nov 12, 2009 at 5:53 PM, Rupa Schomaker wrote: > On Thu, Nov 12, 2009 at 4:32 PM, Robin Vleij wrote: >> On 11/12/09 9:59 PM, Rupa Schomaker wrote: >> If I read it right, this is suited for "complete" nrs. So would I have a >> system connected with lots of DIDs, I would put them in easyroute. Then >> for systems with lots of number ranges, I would use mod_lcr. > > lcr is based on prefix, so the boundaries for which the range is > assigned may not match a prefix. ?You may be better off either: > > > 1) denormalize your ranges and just insert all distinct #s > > 2) Modify mod_easyroute to support ranges > > 3) talk to SWK (he is on irc here and there) about his (non free) > fancier routing options > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From orien at tx.rr.com Thu Nov 12 18:44:57 2009 From: orien at tx.rr.com (Orien Love) Date: Thu, 12 Nov 2009 20:44:57 -0600 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: References: Message-ID: <4AFCC829.2070507@tx.rr.com> Thank you Dana and Michael for your replies, I am getting a spa3000 in the mail soon so I can try it out and see if it will work for my needs, I am going to implement a automatic attendant thanks to the information provided. Since I have not had any replies about the atom board I am guessing that nobody has used one, Could somebody tell me what is a good CPU speed / Memory / FSB be? I really do not have a large budget and cannot afford to buy something that will not work. Thanks again Orien From frank at carmickle.com Thu Nov 12 19:04:30 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 12 Nov 2009 22:04:30 -0500 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: <4AFCC829.2070507@tx.rr.com> References: <4AFCC829.2070507@tx.rr.com> Message-ID: <20091113030429.GS11697@base.carmickle.com> On Thu, Nov 12, Orien Love wrote: > Since I have not had any replies about the atom board I am guessing > that nobody has used one, Could somebody tell me what is a good CPU > speed / Memory / FSB be? > I really do not have a large budget and cannot afford to buy > something that will not work. I have not used an Atom board yet but a few are in the plans. If you do any of them the 330 is the only one to go with as of now. 64 bit and dual core in 8w is pretty nice but then again I don't have one to test with so I can't say for sure. --FC From paul.thirumalai at gmail.com Thu Nov 12 22:27:05 2009 From: paul.thirumalai at gmail.com (Paul Thirumalai) Date: Thu, 12 Nov 2009 22:27:05 -0800 Subject: [Freeswitch-users] Configuring freeswitch with voicepulse In-Reply-To: <900c9adf0911092137vf45ec94ie7473d2c08e5ae12@mail.gmail.com> References: <900c9adf0911092137vf45ec94ie7473d2c08e5ae12@mail.gmail.com> Message-ID: <900c9adf0911122227l74cc638eq5dfab22e6e21caf@mail.gmail.com> Hi Jason Thanks for your response, I setup the configuration with 2 proxies based on the example of the freeswitch wiki. I looked at freeswitch.log and found the following line. Dialplan: sofia/internal/1000 at 74.207.249.79 Action set(effective_caller_id_number=12223334444) Dialplan: sofia/internal/1000 at 74.207.249.79 Action bridge(sofia/gateway/voicepulse/5035440933) 2009-11-13 01:16:23.653519 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/1000 at 74.207.249.79) State Change CS_ROUTING -> CS_EXECUTE 2009-11-13 01:16:23.653519 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1000 at 74.207.249.79 [BREAK] 2009-11-13 01:16:23.653519 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/1000 at 74.207.249.79) State ROUTING going to sleep 2009-11-13 01:16:23.653519 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1000 at 74.207.249.79) Running State Change CS_EXECUTE 2009-11-13 01:16:23.653519 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1000 at 74.207.249.79) State EXECUTE 2009-11-13 01:16:23.653519 [DEBUG] mod_sofia.c:173 sofia/internal/ 1000 at 74.207.249.79 SOFIA EXECUTE 2009-11-13 01:16:23.653519 [DEBUG] switch_core_state_machine.c:151 sofia/internal/1000 at 74.207.249.79 Standard EXECUTE EXECUTE sofia/internal/1000 at 74.207.249.79hash(insert/74.207.249.79-spymap/1000/115be3f6-d01c-11de-8360-976b377ef920) EXECUTE sofia/internal/1000 at 74.207.249.79hash(insert/74.207.249.79-last_dial/1000/5035440933) If this makes sense to someone ,could you please gently guide me in the right direction. Thanks Paul On Mon, Nov 9, 2009 at 9:37 PM, Paul Thirumalai wrote: > Hello All > I am trying to configure freeswitch so that it sends outgoing calls to the > PSTN through voicepulse > My configuration is as follows. > I created a file $PREFIX/conf/sip_profiles/external/voicepulse.xml > > > > > > > > > > > > > > > > > > > > > > > I also have a dial plan defined as follows > > > > data="effective_caller_id_number=12223334444"/> > > > > > > > > When I dial an external number using extension 1000 I get the following > message on the CLI > > ] > freeswitch at ubuntu> 2009-11-10 00:35:44.365614 [NOTICE] > switch_channel.c:602 New Channel sofia/internal/1000 at 74.207.249.79[e4301180-cdba-11de-a864-8927fe94a9f0] > 2009-11-10 00:35:44.366623 [INFO] mod_dialplan_xml.c:315 Processing > Paul->5555555555 in context default > 2009-11-10 00:35:44.368645 [NOTICE] switch_channel.c:602 New Channel > sofia/external/5555555555 [e43092f4-cdba-11de-a864-8927fe94a9f0] > 2009-11-10 00:35:47.59221 [NOTICE] sofia_glue.c:2698 Pre-Answer > sofia/external/5555555555! > 2009-11-10 00:35:47.59221 [INFO] switch_ivr_originate.c:2017 Sending early > media > 2009-11-10 00:35:47.60524 [INFO] mod_sofia.c:1506 Ring SDP: > v=0 > o=FreeSWITCH 1257800805 1257800806 IN IP4 74.207.249.79 > s=FreeSWITCH > c=IN IP4 74.207.249.79 > t=0 0 > m=audio 30542 RTP/AVP 0 101 > a=rtpmap:0 pcmu/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2009-11-10 00:35:47.60524 [NOTICE] mod_sofia.c:1509 Pre-Answer > sofia/internal/1000 at 74.207.249.79! > 2009-11-10 00:35:51.449542 [NOTICE] sofia.c:3849 Hangup > sofia/external/5555555555 [CS_EXCHANGE_MEDIA] [NORMAL_TEMPORARY_FAILURE] > 2009-11-10 00:35:51.452539 [NOTICE] switch_ivr_bridge.c:419 Hangup > sofia/internal/1000 at 74.207.249.79 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 1 > (sofia/internal/1000 at 74.207.249.79) Ended > 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/1000 at 74.207.249.79 [CS_DESTROY] > 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 2 > (sofia/external/5555555555) Ended > 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/external/5555555555 [CS_DESTROY] > > > I am really new to VOIP and having a hard time with this. I am really not > sure how to proceed. Any help would be really appreciated. > > Thanks > Paul > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/97055a96/attachment.html From siniypin at gmail.com Thu Nov 12 23:30:05 2009 From: siniypin at gmail.com (RobertT) Date: Fri, 13 Nov 2009 10:30:05 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> Message-ID: <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> but FS does use tcp for that call leg -> RX 1167 bytes ... from *tcp* ...: And after all there can be other SIP transports combinations FS should interconnect... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/c2e9bdb3/attachment.html From viper at fx-services.com Fri Nov 13 02:23:05 2009 From: viper at fx-services.com (Robin Vleij) Date: Fri, 13 Nov 2009 11:23:05 +0100 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> Message-ID: <4AFD3389.6090409@fx-services.com> On 11/13/09 2:49 AM, Eliot Gable wrote: Hi Eliot, > Or, of course, there is always mod_xml_curl. Basically, XML dialplan > on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a > web application server, web application server responds with XML > routing response, FreeSWITCH routes the call. Yeah, been looking at that one, really cool idea. Then I could build my routing database in any way I want. I'm just worried about performance and the extra delay it'll introduce. But technically with my complex routing demands this would be the right solution, instead of a mix of modules (which probably brings the same extra load on the machine). I'll fiddle a bit. :) /Robin From viper at fx-services.com Fri Nov 13 02:25:09 2009 From: viper at fx-services.com (Robin Vleij) Date: Fri, 13 Nov 2009 11:25:09 +0100 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> Message-ID: <4AFD3405.3040902@fx-services.com> On 11/12/09 11:53 PM, Rupa Schomaker wrote: Hi, > lcr is based on prefix, so the boundaries for which the range is > assigned may not match a prefix. You may be better off either: OK, I think I can forget lcr, it's too far off what I want to do to make it work with some fixes. > 1) denormalize your ranges and just insert all distinct #s > 2) Modify mod_easyroute to support ranges Will look at this. When I come up with something interesting I'll put it on the wiki. In the meantime I've begun looking at xml_curl also, that might in the end really be the best one. I can then build a database+php that responds to stuff in whatever way we want. > 3) talk to SWK (he is on irc here and there) about his (non free) > fancier routing options If I can't get it like I want, I'll look for him. :) Thanks for all the pointers and help! /Robin From juanbackson at gmail.com Fri Nov 13 05:13:12 2009 From: juanbackson at gmail.com (Juan Backson) Date: Fri, 13 Nov 2009 21:13:12 +0800 Subject: [Freeswitch-users] need desperate help with zombie channels Message-ID: <27c25bc40911130513u405062c7kb775e14d04761fd4@mail.gmail.com> Hi, I am having difficulty trying to figure out why there are bunch of zombie channels in my system. It seems to me that these zombies come from apr_thread pool. Does anyone have any idea what may be the cause of these problems? freeswitch at internal> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure b789468a-4412-490b-bc66-32f149ba4d1d,outbound,2009-11-13 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 ,,,XML,default,,,,, 7e1ecaaa-b2d8-47a0-9982-25cd44186d4e,outbound,2009-11-13 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 ,,,XML,default,,,,, 01fa2ff6-f807-4ef0-b988-70a9fe8c4536,outbound,2009-11-13 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 ,CS_EXCHANGE_MEDIA,a88999001,a88999001,192.168.1.116, 999100 at 192.168.1.116:9342,incre_call_stat,125 165 182 235 13 3184093 0,XML,default,,,,, 0271541f-f0b5-482c-b05d-b196f85121be,inbound,2009-11-13 20:15:35,1258114535,sofia/external/88999001 at 192.168.1.116:7342 ,CS_EXECUTE,sipp,88999001,192.168.1.116,88999100,hangup,NORMAL_CLEARING,XML,default,,,,, 7e4ccfec-a4ad-4817-9a82-f1166b34576f,outbound,2009-11-13 20:15:35,1258114536,sofia/external/999100 at 192.168.1.116:9342 ,CS_CONSUME_MEDIA,a88999001,a88999001,192.168.1.116, 999100 at 192.168.1.116:9342,,,XML,default,,,,, 5 total. freeswitch at internal> uuid_kill b789468a-4412-490b-bc66-32f149ba4d1d -ERR No Such Channel! These channels actually do not exist in the system! Here is my gcore output with 5 zombies out of 100K test calls : Thread 21 (process 8946): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 128000} #2 0x00002b3cb3bfb8ca in switch_console_loop () at src/switch_console.c:819 arg = 1 thread = (switch_thread_t *) 0x2aaab00320d0 thd_attr = (switch_threadattr_t *) 0x2aaab0032070 pool = (switch_memory_pool_t *) 0x2aaab0031f88 __func__ = "switch_console_loop" __PRETTY_FUNCTION__ = "switch_console_loop" #3 0x0000000000402884 in main (argc=1, argv=) at src/switch.c:753 pid_path = "/usr/local/freeswitch/log/freeswitch.pid", '\0' pid_buffer = "8946", '\0' old_pid_buffer = '\0' pid_len = 4 old_pid_len = 4198811 err = 0x2b3cb3cec77d "Success" ---Type to continue, or q to quit--- nf = 0 runas_user = runas_group = nc = 0 pid = x = opts = opts_str = '\0' local_argv = {0x7ffff6f08c15 "./freeswitch", 0x0 } arg_argv = {0x0 } alt_dirs = 0 known_opt = high_prio = 0 flags = 65 ret = destroy_status = fd = (switch_file_t *) 0xb6293e0 pool = (switch_memory_pool_t *) 0xb629368 rlp = {rlim_cur = 245760, rlim_max = 245760} waste = 0 __PRETTY_FUNCTION__ = "main" Thread 20 (process 20699): ---Type to continue, or q to quit--- #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 0} #2 0x00002aaaab35e926 in read_packet (listener=0x2aaae7523d08, event=0x2aab3b5ab058, timeout=0) at mod_event_socket.c:1255 do_sleep = 1 '\001' mlen = 0 bytes = 0 mbuf = '\0' buf = '\0' len = 123 status = SWITCH_STATUS_BREAK count = start = 1258117263 pop = (void *) 0x2aaad12f6540 ptr = 0x2aab3b5a98a0 "" crcount = 0 '\0' channel = (switch_channel_t *) 0x0 clen = __func__ = "read_packet" __PRETTY_FUNCTION__ = "read_packet" ---Type to continue, or q to quit--- #3 0x00002aaaab36347a in listener_run (thread=, obj=0x2aaae7523d08) at mod_event_socket.c:2093 listener = (listener_t *) 0x0 buf = '\0' len = 1024 status = event = (switch_event_t *) 0x0 reply = "\000OK log level [7]", '\0' session = (switch_core_session_t *) 0x0 channel = revent = (switch_event_t *) 0x0 var = __PRETTY_FUNCTION__ = "listener_run" __func__ = "listener_run" #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 19 (process 14505): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. ---Type to continue, or q to quit--- #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaaf49798, data=0x7afe0080) at misc/apr_queue.c:276 rv = 0 #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( thread=, obj=) at src/switch_event.c:248 pop = (void *) 0x0 event = (switch_event_t *) 0x0 queue = (switch_queue_t *) 0x2aaaaaf49798 my_id = 1 __func__ = "switch_event_dispatch_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 18 (process 9334): #0 0x0000003054e0d2cb in read () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3cd50c8 in read_char (el=0x2aaab0028180, cp=0x4027002f "") at read.c:294 num_read = 1076297860 tried = 0 ---Type to continue, or q to quit--- #2 0x00002b3cb3cd4ceb in el_gets (el=0x2aaab0028180, nread=0x40270084) at read.c:241 cmdnum = 112 'p' num = -1321754256 ch = 0 '\0' #3 0x00002b3cb3bfc4bb in console_thread (thread=, obj=) at src/switch_console.c:464 arg = 1 count = 1 line = 0x2aaab0034e70 "\n" pool = (switch_memory_pool_t *) 0x2aaab0031f88 __func__ = "console_thread" __PRETTY_FUNCTION__ = "console_thread" #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 17 (process 9333): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 ---Type to continue, or q to quit--- tv = {tv_sec = 0, tv_usec = 0} #2 0x00002b3cb3c53895 in softtimer_runtime () at src/switch_time.c:464 current_ms = 692 x = 690 tick = 292 ts = last = 1258117283599783 fwd_errs = 0 rev_errs = 0 __func__ = "softtimer_runtime" #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x0, obj=0x0) at src/switch_loadable_module.c:94 status = ts = (switch_core_thread_session_t *) 0x0 module = (switch_loadable_module_t *) 0xb6c4e00 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 16 (process 9332): ---Type to continue, or q to quit--- #0 0x0000003054e0d4eb in accept () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c707a4 in apr_socket_accept (new=0x416b4020, sock=0xbcfde38, connection_context=0x2aaacda27718) at network_io/unix/sockets.c:187 No locals. #2 0x00002aaaab35f889 in mod_event_socket_runtime () at mod_event_socket.c:2324 pool = (switch_memory_pool_t *) 0xbcfdc88 listener_pool = (switch_memory_pool_t *) 0x2aaacda27718 rv = sa = (switch_sockaddr_t *) 0xbcfdd68 inbound_socket = (switch_socket_t *) 0x2aaacda277f8 listener = x = __func__ = "mod_event_socket_runtime" #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x14f, obj=0x2aaacda27948) at src/switch_loadable_module.c:94 status = ts = (switch_core_thread_session_t *) 0x2aaacda27948 module = (switch_loadable_module_t *) 0x2aaaac0058c0 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 ---Type to continue, or q to quit--- No symbol table info available. #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 15 (process 9330): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 55000} #2 0x00002aaab503cc4c in node_thread_run (thread=, obj=) at mod_fifo.c:580 val = (void *) 0x0 var = (const void *) 0x0 idle_consumers = hi = (switch_hash_index_t *) 0x0 ppl_waiting = 0 consumer_total = 1087699264 node = #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. ---Type to continue, or q to quit--- Thread 14 (process 9329): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 100} #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( thread=, obj=) at sofia.c:763 profile = (sofia_profile_t *) 0xbce2310 ireg_loops = 18 gateway_loops = 0 loops = 72 qsize = 4294966782 pop = (void *) 0x0 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 13 (process 9328): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 ---Type to continue, or q to quit--- No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 0} #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( thread=, obj=) at sofia.c:763 profile = (sofia_profile_t *) 0x2aaab000eb10 ireg_loops = 5 gateway_loops = 0 loops = 93 qsize = 4294966782 pop = (void *) 0x0 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 12 (process 9327): #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 No symbol table info available. #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce71c0, tout=1000) at su_epoll_port.c:495 ---Type to continue, or q to quit--- j = 198076976 n = 0 events = 0 index = 10922 version = 3 M = 4 ev = 0x41204ef0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce71c0) at su_base_port.c:349 tout = 1000 tout2 = 0 __PRETTY_FUNCTION__ = "su_base_port_run" #3 0x00002aaab45c6c51 in su_port_run (self=0xbce71c0) at su_port.h:326 base = (su_virtual_port_t *) 0xbce71c0 #4 0x00002aaab45c6c29 in su_root_run (self=0xbce72a0) at su_root.c:819 __PRETTY_FUNCTION__ = "su_root_run" #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404f7ac0) at su_pthread_port.c:324 arg = (struct clone_args *) 0x0 task = {{sut_port = 0xbce71c0, sut_root = 0xbce72a0}} zap = 1 #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 ---Type to continue, or q to quit--- No symbol table info available. #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 11 (process 9326): #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 No symbol table info available. #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce78b0, tout=1000) at su_epoll_port.c:495 j = -1342070512 n = 10922 events = 0 index = 10922 version = 3 M = 4 ev = 0x411c8ef0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce78b0) at su_base_port.c:349 tout = 1000 tout2 = 0 __PRETTY_FUNCTION__ = "su_base_port_run" #3 0x00002aaab45c6c51 in su_port_run (self=0xbce78b0) at su_port.h:326 ---Type to continue, or q to quit--- base = (su_virtual_port_t *) 0xbce78b0 #4 0x00002aaab45c6c29 in su_root_run (self=0x2aaab001a060) at su_root.c:819 __PRETTY_FUNCTION__ = "su_root_run" #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404bbac0) at su_pthread_port.c:324 arg = (struct clone_args *) 0x0 task = {{sut_port = 0xbce78b0, sut_root = 0x2aaab001a060}} zap = 1 #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 10 (process 9325): #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 No symbol table info available. #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce6c30, tout=1000) at su_epoll_port.c:495 j = -1268971119 n = 10922 events = 0 index = 0 version = 1 ---Type to continue, or q to quit--- M = 4 ev = 0x404f7c40 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbce6c30, tout=1000) at su_base_port.c:467 now = {tv_sec = 3467106082, tv_usec = 971475} __PRETTY_FUNCTION__ = "su_base_port_step" #3 0x00002aaab45c6d6a in su_port_step (self=0xbce6c30, tout=1000) at su_port.h:340 base = (su_virtual_port_t *) 0xbce6c30 #4 0x00002aaab45c6d32 in su_root_step (self=0xbce4650, tout=1000) at su_root.c:858 __PRETTY_FUNCTION__ = "su_root_step" #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( thread=, obj=) at sofia.c:973 profile = (sofia_profile_t *) 0xbce2310 pool = node = (sip_alias_node_t *) 0x0 s_event = (switch_event_t *) 0x0 sanity = worker_thread = (switch_thread_t *) 0xbce36a0 st = SWITCH_STATUS_SUCCESS __func__ = "sofia_profile_thread_run" ---Type to continue, or q to quit--- __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 9 (process 9324): #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 No symbol table info available. #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbcdffb0, tout=1000) at su_epoll_port.c:495 j = -1268971119 n = 10922 events = 0 index = 0 version = 1 M = 4 ev = 0x404bbc40 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbcdffb0, tout=1000) at su_base_port.c:467 now = {tv_sec = 3467106083, tv_usec = 525146} __PRETTY_FUNCTION__ = "su_base_port_step" ---Type to continue, or q to quit--- #3 0x00002aaab45c6d6a in su_port_step (self=0xbcdffb0, tout=1000) at su_port.h:340 base = (su_virtual_port_t *) 0xbcdffb0 #4 0x00002aaab45c6d32 in su_root_step (self=0xbcdfe00, tout=1000) at su_root.c:858 __PRETTY_FUNCTION__ = "su_root_step" #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( thread=, obj=) at sofia.c:973 profile = (sofia_profile_t *) 0x2aaab000eb10 pool = node = (sip_alias_node_t *) 0x0 s_event = (switch_event_t *) 0x0 sanity = worker_thread = (switch_thread_t *) 0x2aaab000fea0 st = SWITCH_STATUS_SUCCESS __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 8 (process 8999): ---Type to continue, or q to quit--- #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 444000} #2 0x00002b3cb3c14e2a in switch_scheduler_task_thread ( thread=, obj=) at src/switch_scheduler.c:171 __func__ = "switch_scheduler_task_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 7 (process 8998): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 100} #2 0x00002b3cb3c054f5 in switch_core_sql_thread ( thread=, obj=) at src/switch_core_sqldb.c:220 ---Type to continue, or q to quit--- pop = (void *) 0x2aaabf3d6220 itterations = 0 trans = 0 '\0' nothing_in_queue = 1 '\001' len = 100 sql_len = 4844546 sqlbuf = 0x2aab135c7010 "" sql = newlen = lc = 0 __PRETTY_FUNCTION__ = "switch_core_sql_thread" __func__ = "switch_core_sql_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 6 (process 8995): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb64c158, data=0x40893088) at misc/apr_queue.c:276 ---Type to continue, or q to quit--- rv = 0 #2 0x00002b3cb3c48ff1 in log_thread (t=, obj=) at src/switch_log.c:288 pop = (void *) 0x0 node = (switch_log_node_t *) 0x0 binding = (switch_log_binding_t *) 0x0 __func__ = "log_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 5 (process 8951): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaac355a8, data=0x40bec070) at misc/apr_queue.c:276 rv = 0 #2 0x00002b3cb3c1fb14 in switch_event_thread (thread=, obj=) at src/switch_event.c:291 pop = (void *) 0x0 event = ---Type to continue, or q to quit--- queue = (switch_queue_t *) 0x2aaaaac355a8 index = 0 my_id = 2 __func__ = "switch_event_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 4 (process 8950): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaab705a8, data=0x4060a070) at misc/apr_queue.c:276 rv = 0 #2 0x00002b3cb3c1fb14 in switch_event_thread (thread=, obj=) at src/switch_event.c:291 pop = (void *) 0x0 event = queue = (switch_queue_t *) 0x2aaaaab705a8 index = 0 my_id = 1 ---Type to continue, or q to quit--- __func__ = "switch_event_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 3 (process 8949): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb638fa8, data=0x405ce070) at misc/apr_queue.c:276 rv = 0 #2 0x00002b3cb3c1fb14 in switch_event_thread (thread=, obj=) at src/switch_event.c:291 pop = (void *) 0x0 event = queue = (switch_queue_t *) 0xb638fa8 index = 0 my_id = 0 __func__ = "switch_event_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. ---Type to continue, or q to quit--- #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 2 (process 8948): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaacfa5a8, data=0x40592080) at misc/apr_queue.c:276 rv = 0 #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( thread=, obj=) at src/switch_event.c:248 pop = (void *) 0x0 event = (switch_event_t *) 0x0 queue = (switch_queue_t *) 0x2aaaaacfa5a8 my_id = 0 __func__ = "switch_event_dispatch_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. ---Type to continue, or q to quit--- Thread 1 (process 8947): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 451000} #2 0x00002b3cb3c00c95 in pool_thread (thread=, obj=) at src/switch_core_memory.c:490 x = #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. (gdb) (gdb) (gdb) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/4426c281/attachment-0001.html From piotr_zurek at biprotech.com Fri Nov 13 05:59:22 2009 From: piotr_zurek at biprotech.com (=?UTF-8?B?UGlvdHIgxbt1cmVr?=) Date: Fri, 13 Nov 2009 14:59:22 +0100 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <4AFC005A.4090200@biprotech.com> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> <4AFC005A.4090200@biprotech.com> Message-ID: <4AFD663A.1030707@biprotech.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/514ab68c/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: piotr_zurek.vcf Type: text/x-vcard Size: 414 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/514ab68c/attachment.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3678 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/514ab68c/attachment.bin From tayeb.meftah at gmail.com Fri Nov 13 08:31:38 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Fri, 13 Nov 2009 16:31:38 +0000 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: <1352396721.20091110232720@mail.ru> References: <1352396721.20091110232720@mail.ru> Message-ID: <4AFD89EA.2000905@gmail.com> hi use sofia/internal/$1 at your provider ip where $1 is the number thanks Sergey Kobzar a ?crit : > Hello. > > I'm FS newbie and want connect it to SIP provider which does not > require authentication - it make authentication using my IP. > > I've searched through FS documentation and didn't find clear answer. > > Could you help me or maybe give a link to a doc which can help? > > Thanks. > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From anthony.minessale at gmail.com Fri Nov 13 07:40:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Nov 2009 09:40:52 -0600 Subject: [Freeswitch-users] need desperate help with zombie channels In-Reply-To: <27c25bc40911130513u405062c7kb775e14d04761fd4@mail.gmail.com> References: <27c25bc40911130513u405062c7kb775e14d04761fd4@mail.gmail.com> Message-ID: <191c3a030911130740n200b54d1o3785cc703d8d1438@mail.gmail.com> >From the looks of that you probably have an equal number of zombie event socket processes. We do not get involved in load testing. Consider consulting at freeswitch.orgfor professional help. On Fri, Nov 13, 2009 at 7:13 AM, Juan Backson wrote: > Hi, > > I am having difficulty trying to figure out why there are bunch of zombie > channels in my system. It seems to me that these zombies come from > apr_thread pool. > > Does anyone have any idea what may be the cause of these problems? > > > freeswitch at internal> show channels > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure > b789468a-4412-490b-bc66-32f149ba4d1d,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 > ,,,XML,default,,,,, > 7e1ecaaa-b2d8-47a0-9982-25cd44186d4e,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 > ,,,XML,default,,,,, > 01fa2ff6-f807-4ef0-b988-70a9fe8c4536,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_EXCHANGE_MEDIA,a88999001,a88999001,192.168.1.116, > 999100 at 192.168.1.116:9342,incre_call_stat,125 165 182 235 13 3184093 > 0,XML,default,,,,, > 0271541f-f0b5-482c-b05d-b196f85121be,inbound,2009-11-13 > 20:15:35,1258114535,sofia/external/88999001 at 192.168.1.116:7342 > ,CS_EXECUTE,sipp,88999001,192.168.1.116,88999100,hangup,NORMAL_CLEARING,XML,default,,,,, > 7e4ccfec-a4ad-4817-9a82-f1166b34576f,outbound,2009-11-13 > 20:15:35,1258114536,sofia/external/999100 at 192.168.1.116:9342 > ,CS_CONSUME_MEDIA,a88999001,a88999001,192.168.1.116, > 999100 at 192.168.1.116:9342,,,XML,default,,,,, > > 5 total. > > freeswitch at internal> uuid_kill b789468a-4412-490b-bc66-32f149ba4d1d > -ERR No Such Channel! > > These channels actually do not exist in the system! > > > Here is my gcore output with 5 zombies out of 100K test calls : > > > Thread 21 (process 8946): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 128000} > #2 0x00002b3cb3bfb8ca in switch_console_loop () at > src/switch_console.c:819 > arg = 1 > thread = (switch_thread_t *) 0x2aaab00320d0 > thd_attr = (switch_threadattr_t *) 0x2aaab0032070 > pool = (switch_memory_pool_t *) 0x2aaab0031f88 > __func__ = "switch_console_loop" > __PRETTY_FUNCTION__ = "switch_console_loop" > #3 0x0000000000402884 in main (argc=1, argv=) > at src/switch.c:753 > pid_path = "/usr/local/freeswitch/log/freeswitch.pid", '\0' > > pid_buffer = "8946", '\0' > old_pid_buffer = '\0' > pid_len = 4 > old_pid_len = 4198811 > err = 0x2b3cb3cec77d "Success" > ---Type to continue, or q to quit--- > nf = 0 > runas_user = > runas_group = > nc = 0 > pid = > x = > opts = > opts_str = '\0' > local_argv = {0x7ffff6f08c15 "./freeswitch", 0x0 times>} > arg_argv = {0x0 } > alt_dirs = 0 > known_opt = > high_prio = 0 > flags = 65 > ret = > destroy_status = > fd = (switch_file_t *) 0xb6293e0 > pool = (switch_memory_pool_t *) 0xb629368 > rlp = {rlim_cur = 245760, rlim_max = 245760} > waste = 0 > __PRETTY_FUNCTION__ = "main" > > Thread 20 (process 20699): > ---Type to continue, or q to quit--- > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002aaaab35e926 in read_packet (listener=0x2aaae7523d08, > event=0x2aab3b5ab058, timeout=0) at mod_event_socket.c:1255 > do_sleep = 1 '\001' > mlen = 0 > bytes = 0 > mbuf = '\0' > buf = '\0' > len = 123 > status = SWITCH_STATUS_BREAK > count = > start = 1258117263 > pop = (void *) 0x2aaad12f6540 > ptr = 0x2aab3b5a98a0 "" > crcount = 0 '\0' > channel = (switch_channel_t *) 0x0 > clen = > __func__ = "read_packet" > __PRETTY_FUNCTION__ = "read_packet" > ---Type to continue, or q to quit--- > #3 0x00002aaaab36347a in listener_run (thread=, > obj=0x2aaae7523d08) at mod_event_socket.c:2093 > listener = (listener_t *) 0x0 > buf = '\0' > len = 1024 > status = > event = (switch_event_t *) 0x0 > reply = "\000OK log level [7]", '\0' > session = (switch_core_session_t *) 0x0 > channel = > revent = (switch_event_t *) 0x0 > var = > __PRETTY_FUNCTION__ = "listener_run" > __func__ = "listener_run" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 19 (process 14505): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaaf49798, > data=0x7afe0080) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( > thread=, obj=) > at src/switch_event.c:248 > pop = (void *) 0x0 > event = (switch_event_t *) 0x0 > queue = (switch_queue_t *) 0x2aaaaaf49798 > my_id = 1 > __func__ = "switch_event_dispatch_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 18 (process 9334): > #0 0x0000003054e0d2cb in read () from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3cd50c8 in read_char (el=0x2aaab0028180, cp=0x4027002f "") > at read.c:294 > num_read = 1076297860 > tried = 0 > ---Type to continue, or q to quit--- > #2 0x00002b3cb3cd4ceb in el_gets (el=0x2aaab0028180, nread=0x40270084) > at read.c:241 > cmdnum = 112 'p' > num = -1321754256 > ch = 0 '\0' > #3 0x00002b3cb3bfc4bb in console_thread (thread=, > obj=) at src/switch_console.c:464 > arg = 1 > count = 1 > line = 0x2aaab0034e70 "\n" > pool = (switch_memory_pool_t *) 0x2aaab0031f88 > __func__ = "console_thread" > __PRETTY_FUNCTION__ = "console_thread" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 17 (process 9333): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > ---Type to continue, or q to quit--- > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002b3cb3c53895 in softtimer_runtime () at src/switch_time.c:464 > current_ms = 692 > x = 690 > tick = 292 > ts = > last = 1258117283599783 > fwd_errs = 0 > rev_errs = 0 > __func__ = "softtimer_runtime" > #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x0, obj=0x0) > at src/switch_loadable_module.c:94 > status = > ts = (switch_core_thread_session_t *) 0x0 > module = (switch_loadable_module_t *) 0xb6c4e00 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 16 (process 9332): > ---Type to continue, or q to quit--- > #0 0x0000003054e0d4eb in accept () from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c707a4 in apr_socket_accept (new=0x416b4020, > sock=0xbcfde38, > connection_context=0x2aaacda27718) at network_io/unix/sockets.c:187 > No locals. > #2 0x00002aaaab35f889 in mod_event_socket_runtime () > at mod_event_socket.c:2324 > pool = (switch_memory_pool_t *) 0xbcfdc88 > listener_pool = (switch_memory_pool_t *) 0x2aaacda27718 > rv = > sa = (switch_sockaddr_t *) 0xbcfdd68 > inbound_socket = (switch_socket_t *) 0x2aaacda277f8 > listener = > x = > __func__ = "mod_event_socket_runtime" > #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x14f, > obj=0x2aaacda27948) at src/switch_loadable_module.c:94 > status = > ts = (switch_core_thread_session_t *) 0x2aaacda27948 > module = (switch_loadable_module_t *) 0x2aaaac0058c0 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > ---Type to continue, or q to quit--- > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 15 (process 9330): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 55000} > #2 0x00002aaab503cc4c in node_thread_run (thread=, > obj=) at mod_fifo.c:580 > val = (void *) 0x0 > var = (const void *) 0x0 > idle_consumers = > hi = (switch_hash_index_t *) 0x0 > ppl_waiting = 0 > consumer_total = 1087699264 > node = > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > ---Type to continue, or q to quit--- > > Thread 14 (process 9329): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100} > #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( > thread=, obj=) at sofia.c:763 > profile = (sofia_profile_t *) 0xbce2310 > ireg_loops = 18 > gateway_loops = 0 > loops = 72 > qsize = 4294966782 > pop = (void *) 0x0 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 13 (process 9328): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > ---Type to continue, or q to quit--- > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( > thread=, obj=) at sofia.c:763 > profile = (sofia_profile_t *) 0x2aaab000eb10 > ireg_loops = 5 > gateway_loops = 0 > loops = 93 > qsize = 4294966782 > pop = (void *) 0x0 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 12 (process 9327): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce71c0, > tout=1000) > at su_epoll_port.c:495 > ---Type to continue, or q to quit--- > j = 198076976 > n = 0 > events = 0 > index = 10922 > version = 3 > M = 4 > ev = 0x41204ef0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce71c0) > at su_base_port.c:349 > tout = 1000 > tout2 = 0 > __PRETTY_FUNCTION__ = "su_base_port_run" > #3 0x00002aaab45c6c51 in su_port_run (self=0xbce71c0) at su_port.h:326 > base = (su_virtual_port_t *) 0xbce71c0 > #4 0x00002aaab45c6c29 in su_root_run (self=0xbce72a0) at su_root.c:819 > __PRETTY_FUNCTION__ = "su_root_run" > #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404f7ac0) > at su_pthread_port.c:324 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0xbce71c0, sut_root = 0xbce72a0}} > zap = 1 > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > ---Type to continue, or q to quit--- > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 11 (process 9326): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce78b0, > tout=1000) > at su_epoll_port.c:495 > j = -1342070512 > n = 10922 > events = 0 > index = 10922 > version = 3 > M = 4 > ev = 0x411c8ef0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce78b0) > at su_base_port.c:349 > tout = 1000 > tout2 = 0 > __PRETTY_FUNCTION__ = "su_base_port_run" > #3 0x00002aaab45c6c51 in su_port_run (self=0xbce78b0) at su_port.h:326 > ---Type to continue, or q to quit--- > base = (su_virtual_port_t *) 0xbce78b0 > #4 0x00002aaab45c6c29 in su_root_run (self=0x2aaab001a060) at > su_root.c:819 > __PRETTY_FUNCTION__ = "su_root_run" > #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404bbac0) > at su_pthread_port.c:324 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0xbce78b0, sut_root = 0x2aaab001a060}} > zap = 1 > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 10 (process 9325): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce6c30, > tout=1000) > at su_epoll_port.c:495 > j = -1268971119 > n = 10922 > events = 0 > index = 0 > version = 1 > ---Type to continue, or q to quit--- > M = 4 > ev = 0x404f7c40 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbce6c30, tout=1000) > at su_base_port.c:467 > now = {tv_sec = 3467106082, tv_usec = 971475} > __PRETTY_FUNCTION__ = "su_base_port_step" > #3 0x00002aaab45c6d6a in su_port_step (self=0xbce6c30, tout=1000) > at su_port.h:340 > base = (su_virtual_port_t *) 0xbce6c30 > #4 0x00002aaab45c6d32 in su_root_step (self=0xbce4650, tout=1000) > at su_root.c:858 > __PRETTY_FUNCTION__ = "su_root_step" > #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( > thread=, obj=) at sofia.c:973 > profile = (sofia_profile_t *) 0xbce2310 > pool = > node = (sip_alias_node_t *) 0x0 > s_event = (switch_event_t *) 0x0 > sanity = > worker_thread = (switch_thread_t *) 0xbce36a0 > st = SWITCH_STATUS_SUCCESS > __func__ = "sofia_profile_thread_run" > ---Type to continue, or q to quit--- > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 9 (process 9324): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbcdffb0, > tout=1000) > at su_epoll_port.c:495 > j = -1268971119 > n = 10922 > events = 0 > index = 0 > version = 1 > M = 4 > ev = 0x404bbc40 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbcdffb0, tout=1000) > at su_base_port.c:467 > now = {tv_sec = 3467106083, tv_usec = 525146} > __PRETTY_FUNCTION__ = "su_base_port_step" > ---Type to continue, or q to quit--- > #3 0x00002aaab45c6d6a in su_port_step (self=0xbcdffb0, tout=1000) > at su_port.h:340 > base = (su_virtual_port_t *) 0xbcdffb0 > #4 0x00002aaab45c6d32 in su_root_step (self=0xbcdfe00, tout=1000) > at su_root.c:858 > __PRETTY_FUNCTION__ = "su_root_step" > #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( > thread=, obj=) at sofia.c:973 > profile = (sofia_profile_t *) 0x2aaab000eb10 > pool = > node = (sip_alias_node_t *) 0x0 > s_event = (switch_event_t *) 0x0 > sanity = > worker_thread = (switch_thread_t *) 0x2aaab000fea0 > st = SWITCH_STATUS_SUCCESS > __func__ = "sofia_profile_thread_run" > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 8 (process 8999): > ---Type to continue, or q to quit--- > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 444000} > #2 0x00002b3cb3c14e2a in switch_scheduler_task_thread ( > thread=, obj=) > at src/switch_scheduler.c:171 > __func__ = "switch_scheduler_task_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 7 (process 8998): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100} > #2 0x00002b3cb3c054f5 in switch_core_sql_thread ( > thread=, obj=) > at src/switch_core_sqldb.c:220 > ---Type to continue, or q to quit--- > pop = (void *) 0x2aaabf3d6220 > itterations = 0 > trans = 0 '\0' > nothing_in_queue = 1 '\001' > len = 100 > sql_len = 4844546 > sqlbuf = 0x2aab135c7010 "" > sql = > newlen = > lc = 0 > __PRETTY_FUNCTION__ = "switch_core_sql_thread" > __func__ = "switch_core_sql_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 6 (process 8995): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb64c158, data=0x40893088) > at misc/apr_queue.c:276 > ---Type to continue, or q to quit--- > rv = 0 > #2 0x00002b3cb3c48ff1 in log_thread (t=, > obj=) at src/switch_log.c:288 > pop = (void *) 0x0 > node = (switch_log_node_t *) 0x0 > binding = (switch_log_binding_t *) 0x0 > __func__ = "log_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 5 (process 8951): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaac355a8, > data=0x40bec070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > ---Type to continue, or q to quit--- > queue = (switch_queue_t *) 0x2aaaaac355a8 > index = 0 > my_id = 2 > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 4 (process 8950): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaab705a8, > data=0x4060a070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > queue = (switch_queue_t *) 0x2aaaaab705a8 > index = 0 > my_id = 1 > ---Type to continue, or q to quit--- > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 3 (process 8949): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb638fa8, data=0x405ce070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > queue = (switch_queue_t *) 0xb638fa8 > index = 0 > my_id = 0 > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 2 (process 8948): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaacfa5a8, > data=0x40592080) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( > thread=, obj=) > at src/switch_event.c:248 > pop = (void *) 0x0 > event = (switch_event_t *) 0x0 > queue = (switch_queue_t *) 0x2aaaaacfa5a8 > my_id = 0 > __func__ = "switch_event_dispatch_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > ---Type to continue, or q to quit--- > Thread 1 (process 8947): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 451000} > #2 0x00002b3cb3c00c95 in pool_thread (thread=, > obj=) at src/switch_core_memory.c:490 > x = > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > (gdb) > (gdb) > (gdb) > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/c1957db0/attachment-0001.html From mgende at gendesign.com Fri Nov 13 08:03:32 2009 From: mgende at gendesign.com (Michael Gende) Date: Fri, 13 Nov 2009 10:03:32 -0600 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: <20091113030429.GS11697@base.carmickle.com> References: <4AFCC829.2070507@tx.rr.com> <20091113030429.GS11697@base.carmickle.com> Message-ID: Hey Orien, I've put FS on a couple of different commodity hardware platforms, from a 1U (dual CPU, dual core, Gig of memory) server to an old Dell PC (less than a gig of memory, single CPU, a few years old so its a dog) and found I had plenty of juice for a small office, say. On the FS website there is a suggested hardware list if I'm not mistaken. However, depending upon how hard you plan to hammer the system usage-wise, any above average PC platform would probably serve well, in my humble - if not entirely educated - opinion. Regards, Mike G. On Thu, Nov 12, 2009 at 9:04 PM, Frank Carmickle wrote: > On Thu, Nov 12, Orien Love wrote: > > Since I have not had any replies about the atom board I am guessing > > that nobody has used one, Could somebody tell me what is a good CPU > > speed / Memory / FSB be? > > I really do not have a large budget and cannot afford to buy > > something that will not work. > > I have not used an Atom board yet but a few are in the plans. If you do > any of them the 330 is the only one to go with as of now. 64 bit and dual > core in 8w is pretty nice but then again I don't have one to test with so I > can't say for sure. > > --FC > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/37700ebf/attachment.html From jerry.richards at teotech.com Fri Nov 13 09:18:55 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 13 Nov 2009 09:18:55 -0800 Subject: [Freeswitch-users] How To Disable MD5 Authentication? Message-ID: <094BEE1BBB684AD692DD928A6E6E4EAD@greyhawk.tonecommander.com> How can I disable MD5 Authentication upon registration? Best Regards, Jerry From brian at freeswitch.org Fri Nov 13 09:26:12 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Nov 2009 11:26:12 -0600 Subject: [Freeswitch-users] How To Disable MD5 Authentication? In-Reply-To: <094BEE1BBB684AD692DD928A6E6E4EAD@greyhawk.tonecommander.com> References: <094BEE1BBB684AD692DD928A6E6E4EAD@greyhawk.tonecommander.com> Message-ID: <5185D4ED-FCAE-42B0-BE24-52BF35463DAA@freeswitch.org> Are you still wanting to authenticate users? auth-calls=false, blind- registration=true on the profile. /b On Nov 13, 2009, at 11:18 AM, Jerry Richards wrote: > > How can I disable MD5 Authentication upon registration? > > Best Regards, > Jerry > From egable+freeswitch at gmail.com Fri Nov 13 09:58:16 2009 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Fri, 13 Nov 2009 12:58:16 -0500 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: <4AFD3389.6090409@fx-services.com> References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> <4AFD3389.6090409@fx-services.com> Message-ID: Performance is not an issue. I clocked 300 calls per second on such a setup using a Dell R710 with two XEON X5570s and 32 GB RAM as the FreeSWITCH server and a Dell 2950 4-core system with 8 GB RAM as the app server. The app server was at 15% - 20% idle at that rate and the Dell R710 was 65% - 70% idle. The main bottleneck I ran into was using the limit application with ODBC. A mutex lock around the ODBC calls meant that I could only pull 160 calls per second, even though the app server was 55% - 60% idle at that rate, because the ODBC call took 1/160th of a second to complete and all the requests were serialized. In theory, you should get better performance using mod_xml_curl because FreeSWITCH will not have to parse a large XML dial plan. One of the drawbacks of the XML dial plan is that any time it tries to locate a route element, it must perform an XML linear search until it finds the correct child (as can be seen in the source code). Thus, searching the XML dialplan is O(n) operation while mod_xml_curl is typically constant time, or at worst, O(log n), depending on how you are storing / querying your data from your database system. Actually, I suppose you could just be a bad programmer and end up making it exponential, but I'm assuming you know how to write code and design your database in a way that avoids that. I have been considering writing a hash cache for the XML dialplan so that lookups can become constant time, but I have no idea when or if I will find the time to do that. :) On Fri, Nov 13, 2009 at 5:23 AM, Robin Vleij wrote: > On 11/13/09 2:49 AM, Eliot Gable wrote: > > Hi Eliot, > >> Or, of course, there is always mod_xml_curl. Basically, XML dialplan >> on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a >> web application server, web application server responds with XML >> routing response, FreeSWITCH routes the call. > > Yeah, been looking at that one, really cool idea. Then I could build my > routing database in any way I want. I'm just worried about performance > and the extra delay it'll introduce. But technically with my complex > routing demands this would be the right solution, instead of a mix of > modules (which probably brings the same extra load on the machine). > > I'll fiddle a bit. :) > > /Robin > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From egable+freeswitch at gmail.com Fri Nov 13 10:16:47 2009 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Fri, 13 Nov 2009 13:16:47 -0500 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: <4AF4AF73.8070804@tx.rr.com> References: <4AF4AF73.8070804@tx.rr.com> Message-ID: I have built some low-wattage systems before, and from what I have seen, you want to stay away from the low-wattage processors. On a price per performance per watt scale, the lowest wattage Core 2 Duo processors are the best bet. I have a logic-supply ITX system here at home that has a VIA processor in it and it is dirt slow. It cost about $350 - $400. It sucks about 5W max, according to my Watt meter but it takes forever to do anything. Compiling FreeSWITCH on it is an absolute nightmare (it takes hours). I have an alternate MicroATX system with a Core 2 Duo and it pulls about 15W all the time. If I compile FreeSWITCH on that system, it takes a few minutes. The Core 2 Duo system was bought about 6 months after the VIA system. The Core 2 Duo cost about $270 for everything, including 2 GB of RAM. The amount of money saved by running the VIA system at 5W is nowhere close to the inconvenience of waiting for it to do anything. Also, I had to go through about 5-6 different Linux distributions before I found one that would actually install on the VIA processor. I think Suse Linux was the one that finally worked on it. Tried CentOS, Slackware, Ubuntu, Debian, and a couple of others. Now, I have not tried the Atom processor, so it could be very different. However, I have read the Tom's Hardware review that also showed that the Core 2 Duo was several times better on the price per performance per watt scale than the Atom processor, and again, it was mainly because the Atom processor took so much longer to do anything. On Fri, Nov 6, 2009 at 6:21 PM, Orien Love wrote: > First of all, Thanks to the help I received on my pfSense installation, > especially to Michael. ?I have a basic test system up and running. I am > still waiting on some hardware but the base system is working!!!! > > I am looking on advice on how to set up a simple office PBX, 20 phones > and 4 outside lines.with 2 or 3 "operator" phones and the rest will be > extensions. > > Here is my plan, please let me know if it does not make sense, or if I > am going about it > > System Hardware > ?4 spa3000's to handle the outside lines. > ?2-3 polycom 601 phones with expansion modules (Operator phones) > ?18 polycom 330 or other phones for desks. > ?2-24 port cisco POE switches > ?1 pfSense server. > > System Design. > > ?Extension Numbers 2xx > ?Outside line access 1xxxxxxxxxx > ?groups 3xx > ?auto-attendent ??? > > here are my questions > ? ?#1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of > memory handle this proposed system? (Here is the MB I am thing of using > MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) > ? ?#2 how do I pool my spa 3000 FXO lines so that the outgoing calls > use the first available line? also how do insure that metro (non long > distance) calls go to a specific line if available? > > I have learned a lot on how to set up Polycom 601 phones, I am planning > on writing a how to document, is there any specific format? > > Thanks Orien > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From jerry.richards at teotech.com Fri Nov 13 13:59:12 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 13 Nov 2009 13:59:12 -0800 Subject: [Freeswitch-users] Accessing Config Info From Database Message-ID: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> Is there a way to access configuration information from a database (e.g. SQL) rather than from the XML files? Best Regards, Jerry From pjintheusa at gmail.com Fri Nov 13 14:26:59 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 13 Nov 2009 17:26:59 -0500 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> Message-ID: <367751820911131426j46bdf6f4t78b535ea989dfccb@mail.gmail.com> Take a look at http://wiki.freeswitch.org/wiki/Mod_xml_curl to get started. On Fri, Nov 13, 2009 at 4:59 PM, Jerry Richards wrote: > Is there a way to access configuration information from a database (e.g. > SQL) rather than from the XML files? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/1edf5d84/attachment.html From leon at scarlet-internet.nl Fri Nov 13 14:28:55 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Fri, 13 Nov 2009 23:28:55 +0100 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> Message-ID: <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> Hi, You can use mod_xml_curl (generate xml on a webserver): http://wiki.freeswitch.org/wiki/Mod_xml_curl or mod_xml_odbc (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Mod_xml_odbc or LUA together with luasql (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration regards, Leon On Fri, 2009-11-13 at 13:59 -0800, Jerry Richards wrote: > Is there a way to access configuration information from a database (e.g. > SQL) rather than from the XML files? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From robert.hadley at teotech.com Fri Nov 13 09:26:40 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Fri, 13 Nov 2009 09:26:40 -0800 Subject: [Freeswitch-users] Freeswitch configure error using --srcdir option Message-ID: <3894885261AF440BB1025F7488B92E97@greyhawk.tonecommander.com> Hello All On CentOS 5.3, I am trying to build Freeswitch in a different directory and use the -srcdir= option. One reason I want to do this to have Debug and Release build targets from the same source. It doesn't work, the configure errors when it gets to the first library subdirectory lib/srtp and tries to configure in there. The steps I am doing are: 1. Building as root 2. Unzip freeswitch-1.0.4-tar.gz in /opt 3. cd into /opt/freeswitch-1.0.4 4. mkdir Debug 5. cd Debug 6. ../configure -srcdir=".." CFLAGS="-g -ggdb -O2" 7. After several seconds of configuring I get: === configuring in libs/srtp (/opt/freeswitch-1.0.4/Debug/libs/srtp) configure: running /bin/sh ../../../libs/srtp/configure.gnu --disable-option-checking '--prefix=/usr/local/freeswitch' 'CFLAGS=-g -ggdb -O2' --cache-file=/dev/null --srcdir=../../../libs/srtp ../../../libs/srtp/configure.gnu: line 2: ./configure: No such file or directory configure: error: ../../../libs/srtp/configure.gnu failed for libs/srtp [root at roberth-c53 Debug]# The file that's executing is this: [root at roberth-c53 srtp]# cd libs/srtp; cat ../../../libs/srtp/configure.gnu #! /bin/sh ./configure "$@" --disable-shared --with-pic Please tell me if I understood the -srcdir option correctly and if there is a way to do build in a different directory. Thanks, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/354433a3/attachment-0001.html From lists at tigertech.com Fri Nov 13 16:23:19 2009 From: lists at tigertech.com (Robert L Mathews) Date: Fri, 13 Nov 2009 16:23:19 -0800 Subject: [Freeswitch-users] Using play_and_get_digits (or IVR) without a final delay Message-ID: <4AFDF877.5040007@tigertech.com> Hi, I'm a new FreeSWITCH convert from asterisk. It's great software; thanks for making it. I'm trying to play a sound file while listening for possible digits dialed (although in most cases callers will not be dialing anything). If callers do start dialing an extension while the sound file is playing, I want them to be able to dial it slowly without any problems. So the inter-digit timeout should be, say, 2 seconds. However, if people don't start dialing anything while the sound file is playing, I don't want any delay at the end of it. I've tried play_and_get_digits with a 2000 timeout -- but that causes a 2 second "dead air" pause at the end of the sound file if callers don't dial anything. I've also tried using a trivial IVR menu to simulate this, but it has the same problem. Interestingly, it doesn't *look* like IVRs should have the problem, because they allow both "inter-digit-timeout" and "timeout" to be specified separately -- but at least in FreeSWITCH 1.0.4, a short "timeout" value always interrupts the dialing of digits, even if "inter-digit-timeout" is much longer. Is there any way to play a sound file from the dial plan with a long inter-digit delay, but without any final delay if no digits are dialed? Thanks for your time! -- Robert L Mathews, Tiger Technologies From brian at freeswitch.org Fri Nov 13 16:56:38 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Nov 2009 18:56:38 -0600 Subject: [Freeswitch-users] Freeswitch configure error using --srcdir option In-Reply-To: <3894885261AF440BB1025F7488B92E97@greyhawk.tonecommander.com> References: <3894885261AF440BB1025F7488B92E97@greyhawk.tonecommander.com> Message-ID: Don't use --srcdir we don't fully support that and the howto guides do not mention it AT ALL. So doing things that are not in the howto aren't really tested nor supported. /b On Nov 13, 2009, at 11:26 AM, Robert Hadley wrote: > Hello All > > On CentOS 5.3, I am trying to build Freeswitch in a different > directory and use the ?srcdir= option. One reason I want to do this > to have Debug and Release build targets from the same source. > > It doesn?t work, the configure errors when it gets to the first > library subdirectory lib/srtp and tries to configure in there. > > The steps I am doing are: > Building as root > Unzip freeswitch-1.0.4-tar.gz in /opt > cd into /opt/freeswitch-1.0.4 > mkdir Debug > cd Debug > ../configure ?srcdir=?..? CFLAGS=?-g ?ggdb ?O2? > After several seconds of configuring I get: > === configuring in libs/srtp (/opt/freeswitch-1.0.4/Debug/libs/srtp) > configure: running /bin/sh ../../../libs/srtp/configure.gnu -- > disable-option-checking '--prefix=/usr/local/freeswitch' 'CFLAGS=-g > -ggdb -O2' --cache-file=/dev/null --srcdir=../../../libs/srtp > ../../../libs/srtp/configure.gnu: line 2: ./configure: No such file > or directory > configure: error: ../../../libs/srtp/configure.gnu failed for libs/ > srtp > [root at roberth-c53 Debug]# > > The file that?s executing is this: > [root at roberth-c53 srtp]# cd libs/srtp; cat ../../../libs/srtp/ > configure.gnu > #! /bin/sh > ./configure "$@" --disable-shared --with-pic > > > Please tell me if I understood the ?srcdir option correctly and if > there is a way to do build in a different directory. > > Thanks, > Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/dd2c9a64/attachment.html From mike at jerris.com Fri Nov 13 18:12:36 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 13 Nov 2009 21:12:36 -0500 Subject: [Freeswitch-users] Freeswitch configure error using --srcdir option In-Reply-To: References: <3894885261AF440BB1025F7488B92E97@greyhawk.tonecommander.com> Message-ID: <375DC77F-138C-49C7-8CD0-A8A05C485588@jerris.com> Patches to make this work would be gladly accepted. Mike On Nov 13, 2009, at 7:56 PM, Brian West wrote: > Don't use --srcdir we don't fully support that and the howto guides do not mention it AT ALL. So doing things that are not in the howto aren't really tested nor supported. > > /b > > On Nov 13, 2009, at 11:26 AM, Robert Hadley wrote: > >> Hello All >> >> On CentOS 5.3, I am trying to build Freeswitch in a different directory and use the ?srcdir= option. One reason I want to do this to have Debug and Release build targets from the same source. >> >> It doesn?t work, the configure errors when it gets to the first library subdirectory lib/srtp and tries to configure in there. >> >> The steps I am doing are: >> Building as root >> Unzip freeswitch-1.0.4-tar.gz in /opt >> cd into /opt/freeswitch-1.0.4 >> mkdir Debug >> cd Debug >> ../configure ?srcdir=?..? CFLAGS=?-g ?ggdb ?O2? >> After several seconds of configuring I get: >> === configuring in libs/srtp (/opt/freeswitch-1.0.4/Debug/libs/srtp) >> configure: running /bin/sh ../../../libs/srtp/configure.gnu --disable-option-checking '--prefix=/usr/local/freeswitch' 'CFLAGS=-g -ggdb -O2' --cache-file=/dev/null --srcdir=../../../libs/srtp >> ../../../libs/srtp/configure.gnu: line 2: ./configure: No such file or directory >> configure: error: ../../../libs/srtp/configure.gnu failed for libs/srtp >> [root at roberth-c53 Debug]# >> >> The file that?s executing is this: >> [root at roberth-c53 srtp]# cd libs/srtp; cat ../../../libs/srtp/configure.gnu >> #! /bin/sh >> ./configure "$@" --disable-shared --with-pic >> >> >> Please tell me if I understood the ?srcdir option correctly and if there is a way to do build in a different directory. >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/a3e11532/attachment-0001.html From andrew at hijacked.us Fri Nov 13 22:48:40 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Sat, 14 Nov 2009 01:48:40 -0500 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> Message-ID: <20091114064840.GC21765@hijacked.us> On Fri, Nov 13, 2009 at 11:28:55PM +0100, Leon de Rooij wrote: > Hi, > > You can use mod_xml_curl (generate xml on a webserver): > > http://wiki.freeswitch.org/wiki/Mod_xml_curl > > or mod_xml_odbc (generate xml in freeswitch): > > http://wiki.freeswitch.org/wiki/Mod_xml_odbc > > or LUA together with luasql (generate xml in freeswitch): > > http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration > Or, if you're really crazy, the erlang module can do it too (even dynamically): http://wiki.freeswitch.org/wiki/Mod_erlang_event#XML_search_bindings :P Andrew From woodydickson at gmail.com Sat Nov 14 00:31:12 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sat, 14 Nov 2009 16:31:12 +0800 Subject: [Freeswitch-users] problem with mod_xml_odbc Message-ID: Hi, I am having problem trying to use mod_xml_odbc using freeswitch-1.0.5pre. Here is the error I am getting: 2009-11-15 00:17:23.571293 [INFO] mod_xml_odbc.c:647 XML ODBC module loading... 2009-11-15 00:17:23.571354 [NOTICE] mod_xml_odbc.c:563 Binding XML Search Function [directory] 2009-11-15 00:17:23.572299 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-11-15 00:17:23.572361 [CRIT] mod_xml_odbc.c:617 Cannot Open ODBC Database! 2009-11-15 00:17:23.572397 [ERR] mod_xml_odbc.c:650 Unable to load xml_odbc config file 2009-11-15 00:17:23.572424 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_xml_odbc.so **Module load routine returned an error** In my config, I have: [root at localhost autoload_configs]# isql myodbc -v +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> How can I fix this problem? thanks, woody -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/c4533556/attachment.html From dome at tel.co.th Sat Nov 14 01:42:32 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 14 Nov 2009 16:42:32 +0700 Subject: [Freeswitch-users] FreeSWITCH Now Supports Broadvoice BV16, BV32 Voice Codecs Message-ID: <8ccbff060911140142j5fa12113l5a63084200719a0@mail.gmail.com> http://freeswitch.org/node/217 Very fast develop :) one reason why i love FS. another is good performance Dome C. From brian at freeswitch.org Sat Nov 14 03:14:29 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Nov 2009 05:14:29 -0600 Subject: [Freeswitch-users] FreeSWITCH Now Supports Broadvoice BV16, BV32 Voice Codecs In-Reply-To: <8ccbff060911140142j5fa12113l5a63084200719a0@mail.gmail.com> References: <8ccbff060911140142j5fa12113l5a63084200719a0@mail.gmail.com> Message-ID: I'm working out one final detail with Broadcom about on wire bitpacking so you might not be compatible with any device that uses these codecs just yet due to confusion in the RFC and the API. Seems the G.192 bitpacking might have been used on the wire for the devices adding un-needed overhead and causing it to not officially be compatible with the RFC. FreeSWITCH can talk to FreeSWITCH without a problem. /b On Nov 14, 2009, at 3:42 AM, Dome Charoenyost wrote: > http://freeswitch.org/node/217 > > Very fast develop :) > one reason why i love FS. another is good performance > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Sat Nov 14 03:15:50 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Nov 2009 05:15:50 -0600 Subject: [Freeswitch-users] problem with mod_xml_odbc In-Reply-To: References: Message-ID: <2CF76C78-1550-421E-96D2-9B3383E674E5@freeswitch.org> Try using isql to connect to the db... /b On Nov 14, 2009, at 2:31 AM, Woody Dickson wrote: > Hi, > > I am having problem trying to use mod_xml_odbc using > freeswitch-1.0.5pre. From viper at fx-services.com Sat Nov 14 05:31:20 2009 From: viper at fx-services.com (Robin Vleij) Date: Sat, 14 Nov 2009 14:31:20 +0100 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> <4AFD3389.6090409@fx-services.com> Message-ID: <4AFEB128.8090606@fx-services.com> On 11/13/09 6:58 PM, Eliot Gable wrote: Hi Eliot! > Performance is not an issue. I clocked 300 calls per second on such a > setup using a Dell R710 with two XEON X5570s and 32 GB RAM as the > FreeSWITCH server and a Dell 2950 4-core system with 8 GB RAM as the > app server. The app server was at 15% - 20% idle at that rate and the Haha, that should do the trick for me. Sounds realy good. I still prefer to use the internal ENUM system for all lookups, but if that's not possible then the mod_xml_curl method is the one for me I think. That takes all routing complexity away from the FS system. > meant that I could only pull 160 calls per second, even though the app > server was 55% - 60% idle at that rate, because the ODBC call took > 1/160th of a second to complete and all the requests were serialized. OK, that's a problem. On the other hand 160 cps is still very good. If I keep an eye on where I use the limit app, it'll be OK. > are storing / querying your data from your database system. Actually, > I suppose you could just be a bad programmer and end up making it > exponential, but I'm assuming you know how to write code and design > your database in a way that avoids that. There's people in my department that know a few things about that, so I can always let my design get approved by them. :) > I have been considering writing a hash cache for the XML dialplan so > that lookups can become constant time, but I have no idea when or if I > will find the time to do that. :) I know the feeling, working on three things at the same time. But it sounds really good all of this. Have to start thinking about the app server design, see how we can do that. It's always a balance, since in the beginning it will be a very small nr of routes. Having the xml_curl setup is a lot of overkill there. But I also know that when connected systems increases there's no time anymore to do it properly, so it's probably best to build the rocketship right from the start. :) /robin From paul.thirumalai at gmail.com Sat Nov 14 17:11:31 2009 From: paul.thirumalai at gmail.com (Paul Thirumalai) Date: Sat, 14 Nov 2009 17:11:31 -0800 Subject: [Freeswitch-users] Question about channel creation messages in logfile Message-ID: <900c9adf0911141711h11e0e251r748034dfa92959f9@mail.gmail.com> Hello All I am a freeswitch newbie. I have managed to get internal phones setup on different computers at home, using X-lite My issue is with making outbound calls. I'm trying to use voicepulse for that. My Freeswitch server IP is 11.111.123.23 My issue is as follows. I used extension 1000 (X-Lite softphone) to make an out going call my cell phone (555-123-1234) >From the freeswitch.log logfile I see that the channel for 1000 goes from CS_NEW-> CS_INIT->CS_ROUTING . At this point it starts going through the dial plans and finds the correct dialplan. According to the dialplan the outgoing call needs to go to sofia/gateway/voicepulse/5551231234 and I see the following lines in the logfile EXECUTE sofia/internal/1000 at 11.111.123.23hash(insert/11.111.123.23-last_dial/global/115be3f6-d01c-11de-8360-976b377ef920) EXECUTE sofia/internal/1000 at 11.111.123.23bridge(sofia/gateway/voicepulse/5551231234) * *This message makes sense to me and appears to be right. Freeswitch is trying to bridge a call from extension # 1000 to outbound number 5551231234 Now the very next line I see freeswitch attempting to create a channel to the outbound number. 2009-11-13 01:16:23.654519 [NOTICE] switch_channel.c:602 New Channel sofia/external/5551231234 [115c7c76-d01c-11de-8360-976b377ef920] Please correct me if I'm mistaken, but isnt freeswitch supposed to create a channel for sofia/gateway/voicepulse/5551231234 and not sjofia/external/5551231235 In any case the channel to sofia/external/5551231234 changes state from CS_NEW->CS_INIT->CS_ROUTING->CS_CONSUME_MEDIA . At this point the freeswitch gets an Remote SDP, I am assuming from voicepulse. In the end I see message which reads 2009-11-13 01:16:30.880417 [DEBUG] mod_sofia.c:306 sofia/external/5035440933 Overriding SIP cause 503 with 500 from the other leg Can someone please tell me what this error means. SIP error code 500 is what I get in X-Lite also. Also could someone please explain what Pre-Answer is. Thanks Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/3b403cfb/attachment.html From abeka at greatiam.com Sat Nov 14 17:11:52 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Sun, 15 Nov 2009 01:11:52 +0000 Subject: [Freeswitch-users] Registration Error 408 Message-ID: <4AFF5558.3080408@greatiam.com> Hello Please pardon me if the solution to this is somewhere already that I have been unable to locate. I have just got a straight out-of-the-box build of FS. According to the wiki, I should be able to test using user IDs 1001 and 1002. However, I am get the above error. If I, however, un-tick register with domain I do net get the error but does not communicate either. Is there a conf that I should have done ? Thanks in advance. Abeka From abeka at greatiam.com Sat Nov 14 17:18:57 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Sun, 15 Nov 2009 01:18:57 +0000 Subject: [Freeswitch-users] Registration Error - 408 timeout Message-ID: <4AFF5701.8010508@greatiam.com> Hello Please pardon me if the solution to this is somewhere already that I have been unable to locate. I have just got a straight out-of-the-box build of FS. According to the wiki, I should be able to test using user IDs 1001 and 1002. However, I am get the above error. If I, however, un-tick register with domain I do net get the error but does not communicate either. Is there a conf that I should have done ? I am using X-lite3 Thanks in advance. Abeka From brian at freeswitch.org Sat Nov 14 17:41:48 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Nov 2009 19:41:48 -0600 Subject: [Freeswitch-users] Question about channel creation messages in logfile In-Reply-To: <900c9adf0911141711h11e0e251r748034dfa92959f9@mail.gmail.com> References: <900c9adf0911141711h11e0e251r748034dfa92959f9@mail.gmail.com> Message-ID: <635CA8D5-7AB8-470E-AA9D-D98A21A049BB@freeswitch.org> Paul, The channel name is useless it means absolutely NOTHING in FreeSWITCH... what matters in FreeSWITCH is the uuid. You can even set the channel names to what ever you like with the include app in dptools. set_name,Name the channel,,mod_dptools So to answer your question. No it shouldn't have the gateway name in the channel name... As for the 500 error your provider responded 500 Internal Server Error. Pre-Answer is usually early media. /b On Nov 14, 2009, at 7:11 PM, Paul Thirumalai wrote: > Hello All > I am a freeswitch newbie. I have managed to get internal phones > setup on different computers at home, using X-lite > My issue is with making outbound calls. I'm trying to use voicepulse > for that. My Freeswitch server IP is 11.111.123.23 > > My issue is as follows. I used extension 1000 (X-Lite softphone) to > make an out going call my cell phone (555-123-1234) > From the freeswitch.log logfile I see that the channel for 1000 goes > from CS_NEW-> CS_INIT->CS_ROUTING . At this point it starts going > through the dial plans and finds the correct dialplan. According to > the dialplan the outgoing call needs to go to sofia/gateway/ > voicepulse/5551231234 and I see the following lines in the logfile > EXECUTE sofia/internal/1000 at 11.111.123.23 hash(insert/11.111.123.23- > last_dial/global/115be3f6-d01c-11de-8360-976b377ef920) > EXECUTE sofia/internal/1000 at 11.111.123.23 bridge(sofia/gateway/ > voicepulse/5551231234) > > > This message makes sense to me and appears to be right. Freeswitch > is trying to bridge a call from extension # 1000 to outbound number > 5551231234 > > Now the very next line I see freeswitch attempting to create a > channel to the outbound number. > > 2009-11-13 01:16:23.654519 [NOTICE] switch_channel.c:602 New Channel > sofia/external/5551231234 [115c7c76-d01c-11de-8360-976b377ef920] > > Please correct me if I'm mistaken, but isnt freeswitch supposed to > create a channel for sofia/gateway/voicepulse/5551231234 and not > sjofia/external/5551231235 > > > In any case the channel to sofia/external/5551231234 changes state > from CS_NEW->CS_INIT->CS_ROUTING->CS_CONSUME_MEDIA . At this point > the freeswitch gets an Remote SDP, I am assuming from voicepulse. > > In the end I see message which reads > 2009-11-13 01:16:30.880417 [DEBUG] mod_sofia.c:306 sofia/external/ > 5035440933 Overriding SIP cause 503 with 500 from the other leg > > Can someone please tell me what this error means. SIP error code 500 > is what I get in X-Lite also. > > Also could someone please explain what Pre-Answer is. > > Thanks > Paul > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/527b3e02/attachment-0001.html From brian at freeswitch.org Sat Nov 14 17:54:33 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Nov 2009 19:54:33 -0600 Subject: [Freeswitch-users] Registration Error 408 In-Reply-To: <4AFF5558.3080408@greatiam.com> References: <4AFF5558.3080408@greatiam.com> Message-ID: <34223AC5-699B-499B-A3B9-CED0F9CF1C59@freeswitch.org> I'm going to venture to guess you're doing this all on the same machine? /b On Nov 14, 2009, at 7:11 PM, Samuel Abekah-Mensah wrote: > Hello > > Please pardon me if the solution to this is somewhere already that I > have been unable to locate. I have just got a straight out-of-the-box > build of FS. According to the wiki, I should be able to test using > user > IDs 1001 and 1002. However, I am get the above error. If I, however, > un-tick register with domain I do net get the error but does not > communicate either. Is there a conf that I should have done ? > > Thanks in advance. > > Abeka > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From paul.thirumalai at gmail.com Sat Nov 14 18:38:06 2009 From: paul.thirumalai at gmail.com (Paul Thirumalai) Date: Sat, 14 Nov 2009 18:38:06 -0800 Subject: [Freeswitch-users] (no subject) In-Reply-To: <900c9adf0911141836r690909w84a0c281f9c0610c@mail.gmail.com> References: <900c9adf0911141836r690909w84a0c281f9c0610c@mail.gmail.com> Message-ID: <900c9adf0911141838o3472a8edv2da66c399c36a6e4@mail.gmail.com> Hi Brian, Thanks for your clarification. By uuid do you mean uuid of the channel? Can you tell me how I could determine the uuid of the channel Thanks Paul On Sat, Nov 14, 2009 at 6:36 PM, Paul Thirumalai wrote: > Paul, > The channel name is useless it means absolutely NOTHING in > FreeSWITCH... what matters in FreeSWITCH is the uuid. You can even > set the channel names to what ever you like with the include app in > > dptools. > > set_name,Name the channel,,mod_dptools > > So to answer your question. No it shouldn't have the gateway name in > the channel name... As for the 500 error your provider responded 500 > > Internal Server Error. > > Pre-Answer is usually early media. > > /b > > On Nov 14, 2009, at 7:11 PM, Paul Thirumalai wrote: > > >* Hello All > *>* I am a freeswitch newbie. I have managed to get internal phones > *>* setup on different computers at home, using X-lite > *>* My issue is with making outbound calls. I'm trying to use voicepulse > *>* for that. My Freeswitch server IP is 11.111.123.23 > *>* > *>* My issue is as follows. I used extension 1000 (X-Lite softphone) to > *>* make an out going call my cell phone (555-123-1234) > *>* From the freeswitch.log logfile I see that the channel for 1000 goes > *>* from CS_NEW-> CS_INIT->CS_ROUTING . At this point it starts going > *>* through the dial plans and finds the correct dialplan. According to > *>* the dialplan the outgoing call needs to go to sofia/gateway/ > *>* voicepulse/5551231234 and I see the following lines in the logfile > *>* EXECUTE sofia/internal/1000 at 11.111.123.23 hash(insert/11.111.123.23- > *>* last_dial/global/115be3f6-d01c-11de-8360-976b377ef920) > *>* EXECUTE sofia/internal/1000 at 11.111.123.23 bridge(sofia/gateway/ > *>* voicepulse/5551231234) > *>* > *>* > *>* This message makes sense to me and appears to be right. Freeswitch > *>* is trying to bridge a call from extension # 1000 to outbound number > *>* 5551231234 > *>* > *>* Now the very next line I see freeswitch attempting to create a > *>* channel to the outbound number. > *>* > *>* 2009-11-13 01:16:23.654519 [NOTICE] switch_channel.c:602 New Channel > *>* sofia/external/5551231234 [115c7c76-d01c-11de-8360-976b377ef920] > *>* > *>* Please correct me if I'm mistaken, but isnt freeswitch supposed to > *>* create a channel for sofia/gateway/voicepulse/5551231234 and not > *>* sjofia/external/5551231235 > *>* > *>* > *>* In any case the channel to sofia/external/5551231234 changes state > *>* from CS_NEW->CS_INIT->CS_ROUTING->CS_CONSUME_MEDIA . At this point > *>* the freeswitch gets an Remote SDP, I am assuming from voicepulse. > *>* > *>* In the end I see message which reads > *>* 2009-11-13 01:16:30.880417 [DEBUG] mod_sofia.c:306 sofia/external/ > *>* 5035440933 Overriding SIP cause 503 with 500 from the other leg > *>* > *>* Can someone please tell me what this error means. SIP error code 500 > *>* is what I get in X-Lite also. > *>* > *>* Also could someone please explain what Pre-Answer is. > *>* > *>* Thanks > *>* Paul > *>* > *>* _______________________________________________ > *>* FreeSWITCH-users mailing list > *>* FreeSWITCH-users at lists.freeswitch.org > *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > *>* users > *>* http://www.freeswitch.org > * > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/14b6f8c5/attachment.html From tina at a2unlimited.com Sat Nov 14 18:41:13 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Sat, 14 Nov 2009 21:41:13 -0500 Subject: [Freeswitch-users] fs_cli Error Message-ID: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> I'm trying to setup fs_cli on a server that is not running the FreeSWITCH server, and I keep getting the following error: "error while loading shared libraries: libedit.so.0: cannot open shared object file: No such file or directory" When I go to one of my FreeSWITCH servers, where fs_cli is working fine, I cannot find the existence of libedit.so.0 anywhere on the server, so I'm not sure what I'm missing... Any thoughts? - Tina From jason at jasonjgw.net Sat Nov 14 18:53:18 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 15 Nov 2009 13:53:18 +1100 Subject: [Freeswitch-users] fs_cli Error In-Reply-To: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> References: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> Message-ID: <20091115025318.GA2052@jdc.jasonjgw.net> tina at a2unlimited.com wrote: > I'm trying to setup fs_cli on a server that is not running the FreeSWITCH > server, and I keep getting the following error: > > "error while loading shared libraries: libedit.so.0: cannot open shared > object file: No such file or directory" For me, under Debian, it's in the libedit2 package. However, fs_cli isn't looking for it, at least not directly. From william.suffill at gmail.com Sat Nov 14 18:57:14 2009 From: william.suffill at gmail.com (William Suffill) Date: Sat, 14 Nov 2009 21:57:14 -0500 Subject: [Freeswitch-users] fs_cli Error In-Reply-To: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> References: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> Message-ID: <6b65470d0911141857h1ab0696fie35a32bf83e265a5@mail.gmail.com> Libedit shared library isn't on the box and fs_cli needs it. It's included as part of the FreeSWITCH build process so any boxes you have FreeSWITCH installed on would have it. You should be able to install libedit on the box you want to use fs_cli on to fix this. -- W From brian at freeswitch.org Sat Nov 14 18:58:16 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Nov 2009 20:58:16 -0600 Subject: [Freeswitch-users] (no subject) In-Reply-To: <900c9adf0911141838o3472a8edv2da66c399c36a6e4@mail.gmail.com> References: <900c9adf0911141836r690909w84a0c281f9c0610c@mail.gmail.com> <900c9adf0911141838o3472a8edv2da66c399c36a6e4@mail.gmail.com> Message-ID: <005E2F47-F1F8-4931-AB5D-1CBBC30590A3@freeswitch.org> show channels its the first item listed. /b On Nov 14, 2009, at 8:38 PM, Paul Thirumalai wrote: > Hi Brian, > Thanks for your clarification. By uuid do you mean uuid of the > channel? > > Can you tell me how I could determine the uuid of the channel > > Thanks > Paul > > On Sat, Nov 14, 2009 at 6:36 PM, Paul Thirumalai > wrote: > Paul, > The channel name is useless it means absolutely NOTHING in > FreeSWITCH... what matters in FreeSWITCH is the uuid. You can even > > set the channel names to what ever you like with the include app in > > dptools. > > set_name,Name the channel,,mod_dptools > > So to answer your question. No it shouldn't have the gateway name in > the channel name... As for the 500 error your provider responded 500 > > > Internal Server Error. > > Pre-Answer is usually early media. > > /b > > On Nov 14, 2009, at 7:11 PM, Paul Thirumalai wrote: > > > Hello All > > I am a freeswitch newbie. I have managed to get internal phones > > > > setup on different computers at home, using X-lite > > My issue is with making outbound calls. I'm trying to use voicepulse > > for that. My Freeswitch server IP is 11.111.123.23 > > > > > > My issue is as follows. I used extension 1000 (X-Lite softphone) to > > make an out going call my cell phone (555-123-1234) > > From the freeswitch.log logfile I see that the channel for 1000 goes > > > > from CS_NEW-> CS_INIT->CS_ROUTING . At this point it starts going > > through the dial plans and finds the correct dialplan. According to > > the dialplan the outgoing call needs to go to sofia/gateway/ > > > > voicepulse/5551231234 and I see the following lines in the logfile > > EXECUTE sofia/internal/1000 at 11.111.123.23 hash(insert/ > 11.111.123.23- > > > > last_dial/global/115be3f6-d01c-11de-8360-976b377ef920) > > EXECUTE sofia/internal/1000 at 11.111.123.23 bridge(sofia/gateway/ > > > > voicepulse/5551231234) > > > > > > This message makes sense to me and appears to be right. Freeswitch > > is trying to bridge a call from extension # 1000 to outbound number > > > > 5551231234 > > > > Now the very next line I see freeswitch attempting to create a > > channel to the outbound number. > > > > 2009-11-13 01:16:23.654519 [NOTICE] switch_channel.c:602 New Channel > > > > sofia/external/5551231234 [115c7c76-d01c-11de-8360-976b377ef920] > > > > Please correct me if I'm mistaken, but isnt freeswitch supposed to > > create a channel for sofia/gateway/voicepulse/5551231234 and not > > > > sjofia/external/5551231235 > > > > > > In any case the channel to sofia/external/5551231234 changes state > > from CS_NEW->CS_INIT->CS_ROUTING->CS_CONSUME_MEDIA . At this point > > > > the freeswitch gets an Remote SDP, I am assuming from voicepulse. > > > > In the end I see message which reads > > 2009-11-13 01:16:30.880417 [DEBUG] mod_sofia.c:306 sofia/external/ > > > > 5035440933 Overriding SIP cause 503 with 500 from the other leg > > > > Can someone please tell me what this error means. SIP error code 500 > > is what I get in X-Lite also. > > > > > > Also could someone please explain what Pre-Answer is. > > > > Thanks > > Paul > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/af361aae/attachment-0001.html From anthony.minessale at gmail.com Sat Nov 14 20:16:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Nov 2009 22:16:53 -0600 Subject: [Freeswitch-users] fs_cli Error In-Reply-To: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> References: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> Message-ID: <191c3a030911142016o6d2d01e9vcf0f3512a7c0b49@mail.gmail.com> from build root: cd libs/esl make the resulting fs_cli that is in that dir one you type make should be more portable than the one created by the top level make in FS On Sat, Nov 14, 2009 at 8:41 PM, wrote: > I'm trying to setup fs_cli on a server that is not running the FreeSWITCH > server, and I keep getting the following error: > > "error while loading shared libraries: libedit.so.0: cannot open shared > object file: No such file or directory" > > When I go to one of my FreeSWITCH servers, where fs_cli is working fine, I > cannot find the existence of libedit.so.0 anywhere on the server, so I'm > not sure what I'm missing... > > Any thoughts? > > - Tina > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/155749cc/attachment.html From samuelmukoti at gmail.com Sun Nov 15 08:39:18 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Sun, 15 Nov 2009 18:39:18 +0200 Subject: [Freeswitch-users] FS mod_SQL Message-ID: <2584B7AF-4F61-483A-86C6-A9A1961E8EA8@gmail.com> Hi, I'm a newbie to FS, and I wanted to implement a setup where I provision the sip endpoints though a SQL database like mysql and also manage call routing too? Is this possible since I understand FS uses XML config files. Best regards Sam From jmesquita at freeswitch.org Sun Nov 15 10:10:05 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 15 Nov 2009 15:10:05 -0300 Subject: [Freeswitch-users] FS mod_SQL In-Reply-To: <2584B7AF-4F61-483A-86C6-A9A1961E8EA8@gmail.com> References: <2584B7AF-4F61-483A-86C6-A9A1961E8EA8@gmail.com> Message-ID: This is the final answer: http://wiki.freeswitch.org/wiki/Mod_xml_curl JM On Sun, Nov 15, 2009 at 1:39 PM, Samuel Mukoti wrote: > Hi, > > I'm a newbie to FS, and I wanted to implement a setup where I > provision the sip endpoints though a SQL database like mysql and also > manage call routing too? Is this possible since I understand FS uses > XML config files. > > Best regards > > Sam > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091115/9777fe2e/attachment.html From mike at jerris.com Sun Nov 15 10:20:51 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Nov 2009 13:20:51 -0500 Subject: [Freeswitch-users] FS mod_SQL In-Reply-To: <2584B7AF-4F61-483A-86C6-A9A1961E8EA8@gmail.com> References: <2584B7AF-4F61-483A-86C6-A9A1961E8EA8@gmail.com> Message-ID: <632798A4-9812-42D5-AA73-F4DDE912954F@jerris.com> http://wiki.freeswitch.org/wiki/Mod_xml_curl On Nov 15, 2009, at 11:39 AM, Samuel Mukoti wrote: > Hi, > > I'm a newbie to FS, and I wanted to implement a setup where I > provision the sip endpoints though a SQL database like mysql and also > manage call routing too? Is this possible since I understand FS uses > XML config files. > From vedamaker at netscape.net Sun Nov 15 09:42:16 2009 From: vedamaker at netscape.net (vedamaker at netscape.net) Date: Sun, 15 Nov 2009 12:42:16 -0500 Subject: [Freeswitch-users] Problem with Siemens A580 IP Phones Message-ID: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> I am FS beginner and I have a basic PBX setup using FS with the Siemens A580 IP Phones. I thought everything was working fine since I could make and receive basic calls without any obvious issues. However, recently I wanted to use more advanced functions in FS and discovered that I could not use any of DTMF based functions (e.g. call transfer/record) during calls with the Siemens IP phones. The same functions work fine when I use a softphone. So, I started looking at the log file and I think there is some problem between the Siemens IP phones and FS (log file attached below). It seems that when a call comes in, FS calls the extensions and then the extensions send back confirmation and SIP status codes. With softphone extensions, I see 180 (Ringing) and 200 (OK) as normal status. However, with Siemens IP phone extensions, I see 480 (Temporarily Unavailable) which seems to cause FS to terminate the session. So, FS log shows there is actually no active session which explains why it does not performs DTMF detection for the call session. However, the call to Siemens IP phones actually continues with ringing when an extension handset answers the call is established with the caller with full voice communication. I don't know how FS works but this seems very strange. I would like to know how to get FS to work properly with Siemens IP phones including the DTMF functions during calls. Any help would be appreciated. ---------------------------------------------- 2009-11-14 09:35:43.942450 [NOTICE] switch_channel.c:602 New Channel sofia/internal/4155559999 at 192.168.1.254 [22f8ee00-d144-11de-a41f-e5a6b5425f55] 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_NEW 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:404 (sofia/internal/4155559999 at 192.168.1.254) State NEW 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3289 Channel sofia/internal/4155559999 at 192.168.1.254 entering state [received][100] 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3296 Remote SDP: v=0 o=- 119640485 119640485 IN IP4 192.168.1.97 s=- c=IN IP4 192.168.1.97 t=0 0 m=audio 16430 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20] 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20] 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20] 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:2029 Set Codec sofia/internal/4155559999 at 192.168.1.254 PCMU/8000 20 ms 160 samples 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3455 (sofia/internal/4155559999 at 192.168.1.254) State Change CS_NEW -> CS_INIT 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_INIT 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/4155559999 at 192.168.1.254) State INIT 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:83 sofia/internal/4155559999 at 192.168.1.254 SOFIA INIT 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:111 (sofia/internal/4155559999 at 192.168.1.254) State Change CS_INIT -> CS_ROUTING 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/4155559999 at 192.168.1.254) State INIT going to sleep 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_ROUTING 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/4155559999 at 192.168.1.254) State ROUTING 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:130 sofia/internal/4155559999 at 192.168.1.254 SOFIA ROUTING 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:78 sofia/internal/4155559999 at 192.168.1.254 Standard ROUTING 2009-11-14 09:35:43.951943 [INFO] mod_dialplan_xml.c:315 Processing WIRELESS CALLER->4155553333 in context default Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->unloop] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->tod_example] continue=true Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [global-intercept] destination_number(4155553333) =~ /^886$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [group-intercept] destination_number(4155553333) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [intercept-ext] destination_number(4155553333) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->redial] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [redial] destination_number(4155553333) =~ /^870$|^\*66$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->global] continue=true Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/4155559999 at 192.168.1.254 Absolute Condition [global] Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [snom-demo-2] destination_number(4155553333) =~ /^9001$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [snom-demo-1] destination_number(4155553333) =~ /^9000$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [eavesdrop] destination_number(4155553333) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [eavesdrop] destination_number(4155553333) =~ /^779$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->call_return] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [call_return] destination_number(4155553333) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->del-group] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [del-group] destination_number(4155553333) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->add-group] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [add-group] destination_number(4155553333) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [call-group-simo] destination_number(4155553333) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [call-group-order] destination_number(4155553333) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [extension-intercom] destination_number(4155553333) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [Local_Extension] destination_number(4155553333) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->group_dial_ringables] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [group_dial_ringables] destination_number(4155553333) =~ /^1999$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->mobile_extensions] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [mobile_extensions] destination_number(4155553333) =~ /^(20[01][0-9])$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->vmain] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [vmain] destination_number(4155553333) =~ /^vmain$|^4000$$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->vm1000] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [vm1000] destination_number(4155553333) =~ /^vm1000$|^4100$|^\*98$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->sip_uri] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [sip_uri] destination_number(4155553333) =~ /^sip:(.*)$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->nb_conferences] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [nb_conferences] destination_number(4155553333) =~ /^(30\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->wb_conferences] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [wb_conferences] destination_number(4155553333) =~ /^(31\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->uwb_conferences] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [uwb_conferences] destination_number(4155553333) =~ /^(32\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->cdquality_conferences] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [cdquality_conferences] destination_number(4155553333) =~ /^(33\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->freeswitch_public_conf_via_sip] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(4155553333) =~ /^9(888|1616|3232)$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [mad_boss_intercom] destination_number(4155553333) =~ /^0911$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [mad_boss_intercom] destination_number(4155553333) =~ /^0912$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->mad_boss] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [mad_boss] destination_number(4155553333) =~ /^0913$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ivr_demo] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ivr_demo] destination_number(4155553333) =~ /^5000$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->dynamic_conference] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [dynamic_conference] destination_number(4155553333) =~ /^5001$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->rtp_multicast_page] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [rtp_multicast_page] destination_number(4155553333) =~ /^pagegroup$|^7243$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] destination_number(4155553333) =~ /^5900$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->unpark] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] destination_number(4155553333) =~ /^5901$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] destination_number(4155553333) =~ /park\+(\d+)/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->unpark] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] destination_number(4155553333) =~ /^parking$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] destination_number(4155553333) =~ /callpark/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->unpark] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] destination_number(4155553333) =~ /pickup/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->wait] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [wait] destination_number(4155553333) =~ /^wait$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->fax_receive] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [fax_receive] destination_number(4155553333) =~ /^9978$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->fax_transmit] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [fax_transmit] destination_number(4155553333) =~ /^9979$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ringback_180] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ringback_180] destination_number(4155553333) =~ /^9980$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ringback_183_uk_ring] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ringback_183_uk_ring] destination_number(4155553333) =~ /^9981$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ringback_183_music_ring] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ringback_183_music_ring] destination_number(4155553333) =~ /^9982$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ringback_post_answer_uk_ring] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(4155553333) =~ /^9983$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ringback_post_answer_music] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ringback_post_answer_music] destination_number(4155553333) =~ /^9984$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ClueCon] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ClueCon] destination_number(4155553333) =~ /^9991$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->show_info] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [show_info] destination_number(4155553333) =~ /^9992$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->video_record] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [video_record] destination_number(4155553333) =~ /^9993$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->video_playback] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [video_playback] destination_number(4155553333) =~ /^9994$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->delay_echo] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [delay_echo] destination_number(4155553333) =~ /^9995$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->echo] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [echo] destination_number(4155553333) =~ /^9996$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->milliwatt] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [milliwatt] destination_number(4155553333) =~ /^9997$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->tone_stream] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [tone_stream] destination_number(4155553333) =~ /^9998$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->zrtp_enrollement] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [zrtp_enrollement] destination_number(4155553333) =~ /^9787$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->hold_music] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [hold_music] destination_number(4155553333) =~ /^9999$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->fax] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [fax] destination_number(4155553333) =~ /^fax|9777$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->test-9555] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [test-9555] destination_number(4155553333) =~ /^9555$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->test-9666] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [test-9666] destination_number(4155553333) =~ /^9666$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->pizza_demo] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [pizza_demo] destination_number(4155553333) =~ /^(pizza|74992)$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->Inbound-4155553333] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [Inbound-4155553333] destination_number(4155553333) =~ /^4155553333$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action ring_ready() Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action set(ringback=${us-ring}) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action set(call_timeout=28) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action set(continue_on_fail=true) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bridge(${group_call(ringables@${domain_name})}) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action answer() Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action sleep(1000) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action voicemail(default ${domain_name} 1000) 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/4155559999 at 192.168.1.254) State Change CS_ROUTING -> CS_EXECUTE 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/4155559999 at 192.168.1.254) State ROUTING going to sleep 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_EXECUTE 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/4155559999 at 192.168.1.254) State EXECUTE 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:173 sofia/internal/4155559999 at 192.168.1.254 SOFIA EXECUTE 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:151 sofia/internal/4155559999 at 192.168.1.254 Standard EXECUTE EXECUTE sofia/internal/4155559999 at 192.168.1.254 hash(insert/192.168.1.254-spymap/4155559999/22f8ee00-d144-11de-a41f-e5a6b5425f55) EXECUTE sofia/internal/4155559999 at 192.168.1.254 hash(insert/192.168.1.254-last_dial/4155559999/4155553333) EXECUTE sofia/internal/4155559999 at 192.168.1.254 hash(insert/192.168.1.254-last_dial/global/22f8ee00-d144-11de-a41f-e5a6b5425f55) EXECUTE sofia/internal/4155559999 at 192.168.1.254 ring_ready() 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:415 sofia/internal/4155559999 at 192.168.1.254 receive message [RINGING] 2009-11-14 09:35:43.951943 [NOTICE] mod_sofia.c:1449 Ring-Ready sofia/internal/4155559999 at 192.168.1.254! 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:43.951943 [NOTICE] mod_dptools.c:415 Ring Ready sofia/internal/4155559999 at 192.168.1.254! EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(1 b s execute_extension::dx XML features) 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 execute_extension::dx XML features EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/4155559999.2009-11-14-09-35-43.wav) 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/4155559999.2009-11-14-09-35-43.wav EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(3 b s execute_extension::cf XML features) 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 execute_extension::cf XML features EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(ringback=%(2000,4000,440.0,480.0)) 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/4155559999 at 192.168.1.254 SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(transfer_ringback=local_stream://moh) 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/4155559999 at 192.168.1.254 SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(call_timeout=28) 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/4155559999 at 192.168.1.254 SET [call_timeout]=[28] EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(hangup_after_bridge=true) 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/4155559999 at 192.168.1.254 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(continue_on_fail=true) 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/4155559999 at 192.168.1.254 SET [continue_on_fail]=[true] 2009-11-14 09:35:43.966601 [DEBUG] sofia.c:3289 Channel sofia/internal/4155559999 at 192.168.1.254 entering state [early][180] EXECUTE sofia/internal/4155559999 at 192.168.1.254 bridge([presence_id=1011 at 192.168.1.254]sofia/internal/sip:1011 at 192.168.1.98:5872,[presence_id=1012 at 192.168.1.254]sofia/internal/sip:1012 at 192.168.1.98:5872,[presence_id=1014 at 192.168.1.254]sofia/internal/sip:1014 at 192.168.1.97:5060) 2009-11-14 09:35:43.986485 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:1011 at 192.168.1.98:5872 [22ff385a-d144-11de-a41f-e5a6b5425f55] 2009-11-14 09:35:43.986485 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_NEW -> CS_INIT 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.990495 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:1012 at 192.168.1.98:5872 [22ff6230-d144-11de-a41f-e5a6b5425f55] 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_INIT 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1011 at 192.168.1.98:5872) State INIT 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1011 at 192.168.1.98:5872 SOFIA INIT 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_INIT -> CS_ROUTING 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1011 at 192.168.1.98:5872) State INIT going to sleep 2009-11-14 09:35:43.990495 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1011 at 192.168.1.98:5872 entering state [calling][0] 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_ROUTING 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1011 at 192.168.1.98:5872) State ROUTING 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:130 sofia/internal/sip:1011 at 192.168.1.98:5872 SOFIA ROUTING 2009-11-14 09:35:43.990495 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1011 at 192.168.1.98:5872) State ROUTING going to sleep 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_CONSUME_MEDIA 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1011 at 192.168.1.98:5872) State CONSUME_MEDIA 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_NEW -> CS_INIT 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.994449 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:1014 at 192.168.1.97:5060 [22fffdb2-d144-11de-a41f-e5a6b5425f55] 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_INIT 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1012 at 192.168.1.98:5872) State INIT 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1012 at 192.168.1.98:5872 SOFIA INIT 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_INIT -> CS_ROUTING 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.994449 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1012 at 192.168.1.98:5872 entering state [calling][0] 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1012 at 192.168.1.98:5872) State INIT going to sleep 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_ROUTING 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1012 at 192.168.1.98:5872) State ROUTING 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:130 sofia/internal/sip:1012 at 192.168.1.98:5872 SOFIA ROUTING 2009-11-14 09:35:43.994449 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1012 at 192.168.1.98:5872) State ROUTING going to sleep 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_CONSUME_MEDIA 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1012 at 192.168.1.98:5872) State CONSUME_MEDIA 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_NEW -> CS_INIT 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] 2009-11-14 09:35:43.998457 [DEBUG] switch_ivr_originate.c:1701 sofia/internal/4155559999 at 192.168.1.254 receive message [PROGRESS] 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_INIT 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1014 at 192.168.1.97:5060) State INIT 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1014 at 192.168.1.97:5060 SOFIA INIT 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_INIT -> CS_ROUTING 2009-11-14 09:35:43.998457 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] 2009-11-14 09:35:43.998457 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1014 at 192.168.1.97:5060 entering state [calling][0] 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1014 at 192.168.1.97:5060) State INIT going to sleep 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_ROUTING 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1014 at 192.168.1.97:5060) State ROUTING 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:130 sofia/internal/sip:1014 at 192.168.1.97:5060 SOFIA ROUTING 2009-11-14 09:35:43.998457 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-14 09:35:43.998457 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1014 at 192.168.1.97:5060) State ROUTING going to sleep 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_CONSUME_MEDIA 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1014 at 192.168.1.97:5060) State CONSUME_MEDIA 2009-11-14 09:35:43.998457 [INFO] switch_ivr_originate.c:1701 Sending early media 2009-11-14 09:35:44.2435 [DEBUG] sofia_glue.c:2263 AUDIO RTP [sofia/internal/4155559999 at 192.168.1.254] 192.168.1.254 port 31052 -> 192.168.1.97 port 16430 codec: 0 ms: 20 2009-11-14 09:35:44.2435 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms 2009-11-14 09:35:44.6432 [INFO] mod_sofia.c:1506 Ring SDP: v=0 o=FreeSWITCH 1258189091 1258189092 IN IP4 192.168.1.254 s=FreeSWITCH c=IN IP4 192.168.1.254 t=0 0 m=audio 31052 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-11-14 09:35:44.6432 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/4155559999 at 192.168.1.254! 2009-11-14 09:35:44.6432 [DEBUG] sofia.c:3289 Channel sofia/internal/4155559999 at 192.168.1.254 entering state [early][183] 2009-11-14 09:35:44.6432 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:44.6432 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2009-11-14 09:35:44.6432 [DEBUG] switch_ivr_originate.c:1777 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-14 09:35:44.18430 [DEBUG] switch_core_io.c:649 sofia/internal/4155559999 at 192.168.1.254 receive message [TRANSCODING_NECESSARY] 2009-11-14 09:35:44.22473 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1014 at 192.168.1.97:5060 entering state [proceeding][180] 2009-11-14 09:35:44.22473 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:1014 at 192.168.1.97:5060! 2009-11-14 09:35:52.326423 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1011 at 192.168.1.98:5872 entering state [terminated][480] 2009-11-14 09:35:52.326423 [NOTICE] sofia.c:3849 Hangup sofia/internal/sip:1011 at 192.168.1.98:5872 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-11-14 09:35:52.326423 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [KILL] 2009-11-14 09:35:52.326423 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1011 at 192.168.1.98:5872) State CONSUME_MEDIA going to sleep 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_HANGUP 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1011 at 192.168.1.98:5872) State HANGUP 2009-11-14 09:35:52.330490 [DEBUG] mod_sofia.c:306 sofia/internal/sip:1011 at 192.168.1.98:5872 Overriding SIP cause 408 with 480 from the other leg 2009-11-14 09:35:52.330490 [DEBUG] mod_sofia.c:338 Channel sofia/internal/sip:1011 at 192.168.1.98:5872 hanging up, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1011 at 192.168.1.98:5872 Standard HANGUP, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1011 at 192.168.1.98:5872) State HANGUP going to sleep 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_HANGUP -> CS_REPORTING 2009-11-14 09:35:52.330490 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_REPORTING 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1011 at 192.168.1.98:5872) State REPORTING 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:1011 at 192.168.1.98:5872 Standard REPORTING, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1011 at 192.168.1.98:5872) State REPORTING going to sleep 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_REPORTING -> CS_DESTROY 2009-11-14 09:35:52.330490 [DEBUG] switch_core_session.c:1068 Session 542 (sofia/internal/sip:1011 at 192.168.1.98:5872) Locked, Waiting on external entities 2009-11-14 09:35:52.618414 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1012 at 192.168.1.98:5872 entering state [terminated][480] 2009-11-14 09:35:52.618414 [NOTICE] sofia.c:3849 Hangup sofia/internal/sip:1012 at 192.168.1.98:5872 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-11-14 09:35:52.618414 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [KILL] 2009-11-14 09:35:52.618414 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1012 at 192.168.1.98:5872) State CONSUME_MEDIA going to sleep 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_HANGUP 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1012 at 192.168.1.98:5872) State HANGUP 2009-11-14 09:35:52.626426 [DEBUG] mod_sofia.c:306 sofia/internal/sip:1012 at 192.168.1.98:5872 Overriding SIP cause 408 with 480 from the other leg 2009-11-14 09:35:52.626426 [DEBUG] mod_sofia.c:338 Channel sofia/internal/sip:1012 at 192.168.1.98:5872 hanging up, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1012 at 192.168.1.98:5872 Standard HANGUP, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1012 at 192.168.1.98:5872) State HANGUP going to sleep 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_HANGUP -> CS_REPORTING 2009-11-14 09:35:52.626426 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_REPORTING 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1012 at 192.168.1.98:5872) State REPORTING 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:1012 at 192.168.1.98:5872 Standard REPORTING, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1012 at 192.168.1.98:5872) State REPORTING going to sleep 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_REPORTING -> CS_DESTROY 2009-11-14 09:35:52.626426 [DEBUG] switch_core_session.c:1068 Session 543 (sofia/internal/sip:1012 at 192.168.1.98:5872) Locked, Waiting on external entities 2009-11-14 09:35:59.778421 [DEBUG] sofia.c:3289 Channel sofia/internal/4155559999 at 192.168.1.254 entering state [terminated][487] 2009-11-14 09:35:59.778421 [NOTICE] sofia.c:3849 Hangup sofia/internal/4155559999 at 192.168.1.254 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2009-11-14 09:35:59.778421 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/4155559999 at 192.168.1.254 [KILL] 2009-11-14 09:35:59.778421 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:59.798426 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/internal/sip:1014 at 192.168.1.97:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2009-11-14 09:35:59.798426 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [KILL] 2009-11-14 09:35:59.798426 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] 2009-11-14 09:35:59.798426 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1086 Session 542 (sofia/internal/sip:1011 at 192.168.1.98:5872) Ended 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/sip:1011 at 192.168.1.98:5872 [CS_DESTROY] 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1011 at 192.168.1.98:5872) State DESTROY 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:255 sofia/internal/sip:1011 at 192.168.1.98:5872 SOFIA DESTROY 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1011 at 192.168.1.98:5872 Standard DESTROY 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1011 at 192.168.1.98:5872) State DESTROY going to sleep 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1086 Session 543 (sofia/internal/sip:1012 at 192.168.1.98:5872) Ended 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/sip:1012 at 192.168.1.98:5872 [CS_DESTROY] 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1012 at 192.168.1.98:5872) State DESTROY 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:255 sofia/internal/sip:1012 at 192.168.1.98:5872 SOFIA DESTROY 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1012 at 192.168.1.98:5872 Standard DESTROY 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1012 at 192.168.1.98:5872) State DESTROY going to sleep 2009-11-14 09:35:59.798426 [INFO] mod_dptools.c:2093 Originate Failed. Cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/4155559999 at 192.168.1.254) State EXECUTE going to sleep 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_HANGUP 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/4155559999 at 192.168.1.254) State HANGUP 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:306 sofia/internal/4155559999 at 192.168.1.254 Overriding SIP cause 487 with 487 from the other leg 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:338 Channel sofia/internal/4155559999 at 192.168.1.254 hanging up, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:46 sofia/internal/4155559999 at 192.168.1.254 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/4155559999 at 192.168.1.254) State HANGUP going to sleep 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1014 at 192.168.1.97:5060) State CONSUME_MEDIA going to sleep 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_HANGUP 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1014 at 192.168.1.97:5060) State HANGUP 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:306 sofia/internal/sip:1014 at 192.168.1.97:5060 Overriding SIP cause 487 with 487 from the other leg 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:338 Channel sofia/internal/sip:1014 at 192.168.1.97:5060 hanging up, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:406 Sending CANCEL to sofia/internal/sip:1014 at 192.168.1.97:5060 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1014 at 192.168.1.97:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1014 at 192.168.1.97:5060) State HANGUP going to sleep 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_HANGUP -> CS_REPORTING 2009-11-14 09:35:59.810601 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_REPORTING 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1014 at 192.168.1.97:5060) State REPORTING 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:1014 at 192.168.1.97:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1014 at 192.168.1.97:5060) State REPORTING going to sleep 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_REPORTING -> CS_DESTROY 2009-11-14 09:35:59.810601 [DEBUG] switch_core_session.c:1068 Session 544 (sofia/internal/sip:1014 at 192.168.1.97:5060) Locked, Waiting on external entities 2009-11-14 09:35:59.810601 [NOTICE] switch_core_session.c:1086 Session 544 (sofia/internal/sip:1014 at 192.168.1.97:5060) Ended 2009-11-14 09:35:59.810601 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/sip:1014 at 192.168.1.97:5060 [CS_DESTROY] 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1014 at 192.168.1.97:5060) State DESTROY 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:255 sofia/internal/sip:1014 at 192.168.1.97:5060 SOFIA DESTROY 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1014 at 192.168.1.97:5060 Standard DESTROY 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1014 at 192.168.1.97:5060) State DESTROY going to sleep 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/4155559999 at 192.168.1.254) State Change CS_HANGUP -> CS_REPORTING 2009-11-14 09:35:59.814986 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_REPORTING 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/4155559999 at 192.168.1.254) State REPORTING 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:53 sofia/internal/4155559999 at 192.168.1.254 Standard REPORTING, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/4155559999 at 192.168.1.254) State REPORTING going to sleep 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/4155559999 at 192.168.1.254) State Change CS_REPORTING -> CS_DESTROY 2009-11-14 09:35:59.814986 [DEBUG] switch_core_session.c:1068 Session 541 (sofia/internal/4155559999 at 192.168.1.254) Locked, Waiting on external entities 2009-11-14 09:35:59.814986 [NOTICE] switch_core_session.c:1086 Session 541 (sofia/internal/4155559999 at 192.168.1.254) Ended 2009-11-14 09:35:59.814986 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/4155559999 at 192.168.1.254 [CS_DESTROY] 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/4155559999 at 192.168.1.254) State DESTROY 2009-11-14 09:35:59.814986 [DEBUG] mod_sofia.c:255 sofia/internal/4155559999 at 192.168.1.254 SOFIA DESTROY 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:60 sofia/internal/4155559999 at 192.168.1.254 Standard DESTROY 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/4155559999 at 192.168.1.254) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091115/3f873e0d/attachment-0001.html From abeka at greatiam.com Sun Nov 15 10:53:27 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Sun, 15 Nov 2009 18:53:27 +0000 Subject: [Freeswitch-users] Registration Error 408 In-Reply-To: <34223AC5-699B-499B-A3B9-CED0F9CF1C59@freeswitch.org> References: <4AFF5558.3080408@greatiam.com> <34223AC5-699B-499B-A3B9-CED0F9CF1C59@freeswitch.org> Message-ID: <4B004E27.3040600@greatiam.com> Hi Brian I have FS running on a FC11 box and the clients IDs 1001 and 1002 running off 2 Windows boxes using X-lite3 Thanks Brian West wrote: >
I'm going > to venture to guess you're doing this all on the same machine? > > /b > > On Nov 14, 2009, at 7:11 PM, Samuel Abekah-Mensah wrote: > >> Hello >> >> Please pardon me if the solution to this is somewhere already that I >> have been unable to locate. I have just got a straight out-of-the-box >> build of FS. According to the wiki, I should be able to test using user >> IDs 1001 and 1002. However, I am get the above error. If I, however, >> un-tick register with domain I do net get the error but does not >> communicate either. Is there a conf that I should have done ? >> >> Thanks in advance. >> >> Abeka >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > >
> From mike at jerris.com Sun Nov 15 11:02:37 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Nov 2009 14:02:37 -0500 Subject: [Freeswitch-users] Problem with Siemens A580 IP Phones In-Reply-To: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> References: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> Message-ID: It doesn't look like your call ever gets setup in this trace, if you enable the sip trace you might see a bit more, but it looks like we are receiving a 480 response from the called phone. Mike On Nov 15, 2009, at 12:42 PM, vedamaker at netscape.net wrote: > > I am FS beginner and I have a basic PBX setup using FS with the Siemens A580 IP Phones. I thought everything was working fine since I could make and receive basic calls without any obvious issues. However, recently I wanted to use more advanced functions in FS and discovered that I could not use any of DTMF based functions (e.g. call transfer/record) during calls with the Siemens IP phones. The same functions work fine when I use a softphone. So, I started looking at the log file and I think there is some problem between the Siemens IP phones and FS (log file attached below). It seems that when a call comes in, FS calls the extensions and then the extensions send back confirmation and SIP status codes. With softphone extensions, I see 180 (Ringing) and 200 (OK) as normal status. However, with Siemens IP phone extensions, I see 480 (Temporarily Unavailable) which seems to cause FS to terminate the session. So, FS log shows there is actually no active session which explains why it does not performs DTMF detection for the call session. However, the call to Siemens IP phones actually continues with ringing when an extension handset answers the call is established with the caller with full voice communication. I don't know how FS works but this seems very strange. I would like to know how to get FS to work properly with Siemens IP phones including the DTMF functions during calls. Any help would be appreciated. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091115/66e5c971/attachment.html From jmesquita at freeswitch.org Sun Nov 15 11:07:43 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 15 Nov 2009 17:07:43 -0200 Subject: [Freeswitch-users] Problem with Siemens A580 IP Phones In-Reply-To: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> References: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> Message-ID: I have the same phone and all works fine. Make sure you are setting DTMF to RFC2833 on the phone config page. This log is supposed to go on pastebin (http://pastebin.freeswitch.org), not here. Also, you are making a group call which makes you ring 2 endpoints at the same time. Verify your dialplan and make sure that's what you need/want. Regards, JM On Sun, Nov 15, 2009 at 3:42 PM, wrote: > > I am FS beginner and I have a basic PBX setup using FS with the Siemens > A580 IP Phones. I thought everything was working fine since I could make > and receive basic calls without any obvious issues. However, recently I > wanted to use more advanced functions in FS and discovered that I could not > use any of DTMF based functions (e.g. call transfer/record) during calls > with the Siemens IP phones. The same functions work fine when I use a > softphone. So, I started looking at the log file and I think there is some > problem between the Siemens IP phones and FS (log file attached below). It > seems that when a call comes in, FS calls the extensions and then the > extensions send back confirmation and SIP status codes. With softphone > extensions, I see 180 (Ringing) and 200 (OK) as normal status. However, > with Siemens IP phone extensions, I see 480 (Temporarily Unavailable) which > seems to cause FS to terminate the session. So, FS log shows there is > actually no active session which explains why it does not performs DTMF > detection for the call session. However, the call to Siemens IP phones > actually continues with ringing when an extension handset answers the call > is established with the caller with full voice communication. I don't know > how FS works but this seems very strange. I would like to know how to get > FS to work properly with Siemens IP phones including the DTMF functions > during calls. Any help would be appreciated. > > > ---------------------------------------------- > 2009-11-14 09:35:43.942450 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/4155559999 at 192.168.1.254[22f8ee00-d144-11de-a41f-e5a6b5425f55] > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_NEW > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:404 > (sofia/internal/4155559999 at 192.168.1.254) State NEW > 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3289 Channel sofia/internal/ > 4155559999 at 192.168.1.254 entering state [received][100] > 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3296 Remote SDP: > v=0 > o=- 119640485 119640485 IN IP4 192.168.1.97 > s=- > c=IN IP4 192.168.1.97 > t=0 0 > m=audio 16430 RTP/AVP 0 100 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:115:32000:20] > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:107:16000:20] > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G722:9:8000:20] > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[PCMU:0:8000:20] > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:2029 Set Codec > sofia/internal/4155559999 at 192.168.1.254 PCMU/8000 20 ms 160 samples > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload > to 101 > 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3455 (sofia/internal/ > 4155559999 at 192.168.1.254) State Change CS_NEW -> CS_INIT > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_INIT > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/4155559999 at 192.168.1.254) State INIT > 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:83 sofia/internal/ > 4155559999 at 192.168.1.254 SOFIA INIT > 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 4155559999 at 192.168.1.254) State Change CS_INIT -> CS_ROUTING > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/4155559999 at 192.168.1.254) State INIT going to sleep > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_ROUTING > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/4155559999 at 192.168.1.254) State ROUTING > 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:130 sofia/internal/ > 4155559999 at 192.168.1.254 SOFIA ROUTING > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/4155559999 at 192.168.1.254 Standard ROUTING > 2009-11-14 09:35:43.951943 [INFO] mod_dialplan_xml.c:315 Processing > WIRELESS CALLER->4155553333 in context default > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->unloop] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->tod_example] continue=true > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->global-intercept] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [global-intercept] destination_number(4155553333) =~ /^886$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->group-intercept] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [group-intercept] destination_number(4155553333) =~ /^\*8$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->intercept-ext] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [intercept-ext] destination_number(4155553333) =~ /^\*\*(\d+)$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->redial] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [redial] > destination_number(4155553333) =~ /^870$|^\*66$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->global] continue=true > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [global] > ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ > break=never > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Absolute Condition > [global] > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->snom-demo-2] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [snom-demo-2] destination_number(4155553333) =~ /^9001$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->snom-demo-1] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [snom-demo-1] destination_number(4155553333) =~ /^9000$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->eavesdrop] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [eavesdrop] > destination_number(4155553333) =~ /^88(.*)$|^\*0(.*)$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->eavesdrop] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [eavesdrop] > destination_number(4155553333) =~ /^779$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->call_return] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [call_return] destination_number(4155553333) =~ /^\*69$|^869$|^lcr$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->del-group] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [del-group] > destination_number(4155553333) =~ /^80(\d{2})$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->add-group] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [add-group] > destination_number(4155553333) =~ /^81(\d{2})$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->call-group-simo] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [call-group-simo] destination_number(4155553333) =~ /^82(\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->call-group-order] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [call-group-order] destination_number(4155553333) =~ /^83(\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->extension-intercom] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [extension-intercom] destination_number(4155553333) =~ /^8(10[01][0-9])$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->Local_Extension] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [Local_Extension] destination_number(4155553333) =~ /^(10[01][0-9])$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->group_dial_ringables] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [group_dial_ringables] destination_number(4155553333) =~ /^1999$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->mobile_extensions] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [mobile_extensions] destination_number(4155553333) =~ /^(20[01][0-9])$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->vmain] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [vmain] > destination_number(4155553333) =~ /^vmain$|^4000$$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->vm1000] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [vm1000] > destination_number(4155553333) =~ /^vm1000$|^4100$|^\*98$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->sip_uri] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [sip_uri] > destination_number(4155553333) =~ /^sip:(.*)$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->nb_conferences] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [nb_conferences] destination_number(4155553333) =~ /^(30\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->wb_conferences] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [wb_conferences] destination_number(4155553333) =~ /^(31\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->uwb_conferences] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [uwb_conferences] destination_number(4155553333) =~ /^(32\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->cdquality_conferences] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [cdquality_conferences] destination_number(4155553333) =~ /^(33\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->freeswitch_public_conf_via_sip] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [freeswitch_public_conf_via_sip] destination_number(4155553333) =~ > /^9(888|1616|3232)$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->mad_boss_intercom] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [mad_boss_intercom] destination_number(4155553333) =~ /^0911$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->mad_boss_intercom] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [mad_boss_intercom] destination_number(4155553333) =~ /^0912$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->mad_boss] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [mad_boss] > destination_number(4155553333) =~ /^0913$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ivr_demo] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ivr_demo] > destination_number(4155553333) =~ /^5000$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->dynamic_conference] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [dynamic_conference] destination_number(4155553333) =~ /^5001$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->rtp_multicast_page] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [rtp_multicast_page] destination_number(4155553333) =~ /^pagegroup$|^7243$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] > destination_number(4155553333) =~ /^5900$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->unpark] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] > destination_number(4155553333) =~ /^5901$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] > destination_number(4155553333) =~ /park\+(\d+)/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->unpark] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] > destination_number(4155553333) =~ /^parking$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] > destination_number(4155553333) =~ /callpark/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->unpark] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] > destination_number(4155553333) =~ /pickup/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->wait] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [wait] > destination_number(4155553333) =~ /^wait$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->fax_receive] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [fax_receive] destination_number(4155553333) =~ /^9978$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->fax_transmit] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [fax_transmit] destination_number(4155553333) =~ /^9979$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ringback_180] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [ringback_180] destination_number(4155553333) =~ /^9980$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ringback_183_uk_ring] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [ringback_183_uk_ring] destination_number(4155553333) =~ /^9981$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ringback_183_music_ring] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [ringback_183_music_ring] destination_number(4155553333) =~ /^9982$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ringback_post_answer_uk_ring] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [ringback_post_answer_uk_ring] destination_number(4155553333) =~ /^9983$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ringback_post_answer_music] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [ringback_post_answer_music] destination_number(4155553333) =~ /^9984$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ClueCon] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ClueCon] > destination_number(4155553333) =~ /^9991$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->show_info] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [show_info] > destination_number(4155553333) =~ /^9992$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->video_record] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [video_record] destination_number(4155553333) =~ /^9993$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->video_playback] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [video_playback] destination_number(4155553333) =~ /^9994$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->delay_echo] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [delay_echo] destination_number(4155553333) =~ /^9995$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->echo] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [echo] > destination_number(4155553333) =~ /^9996$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->milliwatt] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [milliwatt] > destination_number(4155553333) =~ /^9997$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->tone_stream] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [tone_stream] destination_number(4155553333) =~ /^9998$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->zrtp_enrollement] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [zrtp_enrollement] destination_number(4155553333) =~ /^9787$/ break=on-false > > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->hold_music] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [hold_music] destination_number(4155553333) =~ /^9999$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->fax] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [fax] > destination_number(4155553333) =~ /^fax|9777$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->test-9555] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [test-9555] > destination_number(4155553333) =~ /^9555$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->test-9666] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [test-9666] > destination_number(4155553333) =~ /^9666$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->pizza_demo] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [pizza_demo] destination_number(4155553333) =~ /^(pizza|74992)$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->Inbound-4155553333] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) > [Inbound-4155553333] destination_number(4155553333) =~ /^4155553333$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action ring_ready() > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(1 b > s execute_extension::dx XML features) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(2 b > s > record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) > > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(3 b > s execute_extension::cf XML features) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > set(ringback=${us-ring}) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > set(transfer_ringback=local_stream://moh) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > set(call_timeout=28) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > set(hangup_after_bridge=true) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > set(continue_on_fail=true) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > bridge(${group_call(ringables@${domain_name})}) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action answer() > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action sleep(1000) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action voicemail(default > ${domain_name} 1000) > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:114 > (sofia/internal/4155559999 at 192.168.1.254) State Change CS_ROUTING -> > CS_EXECUTE > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/4155559999 at 192.168.1.254) State ROUTING going to sleep > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_EXECUTE > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/4155559999 at 192.168.1.254) State EXECUTE > 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:173 sofia/internal/ > 4155559999 at 192.168.1.254 SOFIA EXECUTE > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:151 > sofia/internal/4155559999 at 192.168.1.254 Standard EXECUTE > EXECUTE sofia/internal/4155559999 at 192.168.1.254hash(insert/192.168.1.254-spymap/4155559999/22f8ee00-d144-11de-a41f-e5a6b5425f55) > > EXECUTE sofia/internal/4155559999 at 192.168.1.254hash(insert/192.168.1.254-last_dial/4155559999/4155553333) > EXECUTE sofia/internal/4155559999 at 192.168.1.254hash(insert/192.168.1.254-last_dial/global/22f8ee00-d144-11de-a41f-e5a6b5425f55) > > EXECUTE sofia/internal/4155559999 at 192.168.1.254 ring_ready() > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:415 sofia/internal/ > 4155559999 at 192.168.1.254 receive message [RINGING] > 2009-11-14 09:35:43.951943 [NOTICE] mod_sofia.c:1449 Ring-Ready > sofia/internal/4155559999 at 192.168.1.254! > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:630 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:43.951943 [NOTICE] mod_dptools.c:415 Ring Ready > sofia/internal/4155559999 at 192.168.1.254! > EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(1 b s > execute_extension::dx XML features) > 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 > execute_extension::dx XML features > EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(2 b s > record_session::/usr/local/freeswitch/recordings/4155559999.2009-11-14-09-35-43.wav) > > 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 > record_session::/usr/local/freeswitch/recordings/4155559999.2009-11-14-09-35-43.wav > > EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(3 b s > execute_extension::cf XML features) > 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 > execute_extension::cf XML features > EXECUTE sofia/internal/4155559999 at 192.168.1.254set(ringback=%(2000,4000,440.0,480.0)) > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/ > 4155559999 at 192.168.1.254 SET [ringback]=[%(2000,4000,440.0,480.0)] > EXECUTE sofia/internal/4155559999 at 192.168.1.254set(transfer_ringback=local_stream://moh) > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/ > 4155559999 at 192.168.1.254 SET [transfer_ringback]=[local_stream://moh] > EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(call_timeout=28) > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/ > 4155559999 at 192.168.1.254 SET [call_timeout]=[28] > EXECUTE sofia/internal/4155559999 at 192.168.1.254set(hangup_after_bridge=true) > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/ > 4155559999 at 192.168.1.254 SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(continue_on_fail=true) > > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/ > 4155559999 at 192.168.1.254 SET [continue_on_fail]=[true] > 2009-11-14 09:35:43.966601 [DEBUG] sofia.c:3289 Channel sofia/internal/ > 4155559999 at 192.168.1.254 entering state [early][180] > EXECUTE sofia/internal/4155559999 at 192.168.1.254 bridge([presence_id= > 1011 at 192.168.1.254]sofia/internal/sip:1011 at 192.168.1.98:5872,[presence_id= > 1012 at 192.168.1.254]sofia/internal/sip:1012 at 192.168.1.98:5872,[presence_id= > 1014 at 192.168.1.254]sofia/internal/sip:1014 at 192.168.1.97:5060) > 2009-11-14 09:35:43.986485 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/sip:1011 at 192.168.1.98:5872[22ff385a-d144-11de-a41f-e5a6b5425f55] > 2009-11-14 09:35:43.986485 [DEBUG] mod_sofia.c:2811 (sofia/internal/ > sip:1011 at 192.168.1.98:5872) State Change CS_NEW -> CS_INIT > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.990495 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/sip:1012 at 192.168.1.98:5872[22ff6230-d144-11de-a41f-e5a6b5425f55] > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_INIT > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State INIT > 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:83 sofia/internal/ > sip:1011 at 192.168.1.98:5872 SOFIA INIT > 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:111 (sofia/internal/ > sip:1011 at 192.168.1.98:5872) State Change CS_INIT -> CS_ROUTING > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State INIT going to sleep > 2009-11-14 09:35:43.990495 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1011 at 192.168.1.98:5872 entering state [calling][0] > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change > CS_ROUTING > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State ROUTING > 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:130 sofia/internal/ > sip:1011 at 192.168.1.98:5872 SOFIA ROUTING > 2009-11-14 09:35:43.990495 [DEBUG] switch_ivr_originate.c:63 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State ROUTING going to sleep > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State CONSUME_MEDIA > 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:2811 (sofia/internal/ > sip:1012 at 192.168.1.98:5872) State Change CS_NEW -> CS_INIT > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.994449 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/sip:1014 at 192.168.1.97:5060[22fffdb2-d144-11de-a41f-e5a6b5425f55] > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_INIT > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State INIT > 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:83 sofia/internal/ > sip:1012 at 192.168.1.98:5872 SOFIA INIT > 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:111 (sofia/internal/ > sip:1012 at 192.168.1.98:5872) State Change CS_INIT -> CS_ROUTING > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.994449 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1012 at 192.168.1.98:5872 entering state [calling][0] > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State INIT going to sleep > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change > CS_ROUTING > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State ROUTING > 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:130 sofia/internal/ > sip:1012 at 192.168.1.98:5872 SOFIA ROUTING > 2009-11-14 09:35:43.994449 [DEBUG] switch_ivr_originate.c:63 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State ROUTING going to sleep > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State CONSUME_MEDIA > 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:2811 (sofia/internal/ > sip:1014 at 192.168.1.97:5060) State Change CS_NEW -> CS_INIT > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] > 2009-11-14 09:35:43.998457 [DEBUG] switch_ivr_originate.c:1701 > sofia/internal/4155559999 at 192.168.1.254 receive message [PROGRESS] > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_INIT > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State INIT > 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:83 sofia/internal/ > sip:1014 at 192.168.1.97:5060 SOFIA INIT > 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:111 (sofia/internal/ > sip:1014 at 192.168.1.97:5060) State Change CS_INIT -> CS_ROUTING > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] > 2009-11-14 09:35:43.998457 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1014 at 192.168.1.97:5060 entering state [calling][0] > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State INIT going to sleep > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change > CS_ROUTING > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State ROUTING > 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:130 sofia/internal/ > sip:1014 at 192.168.1.97:5060 SOFIA ROUTING > 2009-11-14 09:35:43.998457 [DEBUG] switch_ivr_originate.c:63 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State ROUTING going to sleep > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State CONSUME_MEDIA > 2009-11-14 09:35:43.998457 [INFO] switch_ivr_originate.c:1701 Sending early > media > 2009-11-14 09:35:44.2435 [DEBUG] sofia_glue.c:2263 AUDIO RTP > [sofia/internal/4155559999 at 192.168.1.254] 192.168.1.254 port 31052 -> > 192.168.1.97 port 16430 codec: 0 ms: 20 > 2009-11-14 09:35:44.2435 [DEBUG] switch_rtp.c:1138 Starting timer [soft] > 160 bytes per 20ms > 2009-11-14 09:35:44.6432 [INFO] mod_sofia.c:1506 Ring SDP: > v=0 > o=FreeSWITCH 1258189091 1258189092 IN IP4 192.168.1.254 > s=FreeSWITCH > c=IN IP4 192.168.1.254 > t=0 0 > m=audio 31052 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2009-11-14 09:35:44.6432 [NOTICE] mod_sofia.c:1509 Pre-Answer > sofia/internal/4155559999 at 192.168.1.254! > 2009-11-14 09:35:44.6432 [DEBUG] sofia.c:3289 Channel sofia/internal/ > 4155559999 at 192.168.1.254 entering state [early][183] > 2009-11-14 09:35:44.6432 [DEBUG] switch_core_session.c:630 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:44.6432 [DEBUG] switch_ivr_originate.c:1718 Raw Codec > Activation Success L16 at 8000hz 1 channel 20ms > 2009-11-14 09:35:44.6432 [DEBUG] switch_ivr_originate.c:1777 Play Ringback > Tone [%(2000,4000,440.0,480.0)] > 2009-11-14 09:35:44.18430 [DEBUG] switch_core_io.c:649 sofia/internal/ > 4155559999 at 192.168.1.254 receive message [TRANSCODING_NECESSARY] > 2009-11-14 09:35:44.22473 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1014 at 192.168.1.97:5060 entering state [proceeding][180] > 2009-11-14 09:35:44.22473 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/ > sip:1014 at 192.168.1.97:5060! > 2009-11-14 09:35:52.326423 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1011 at 192.168.1.98:5872 entering state [terminated][480] > 2009-11-14 09:35:52.326423 [NOTICE] sofia.c:3849 Hangup sofia/internal/ > sip:1011 at 192.168.1.98:5872 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > 2009-11-14 09:35:52.326423 [DEBUG] switch_channel.c:1683 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [KILL] > 2009-11-14 09:35:52.326423 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State CONSUME_MEDIA going to > sleep > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_HANGUP > > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State HANGUP > 2009-11-14 09:35:52.330490 [DEBUG] mod_sofia.c:306 sofia/internal/ > sip:1011 at 192.168.1.98:5872 Overriding SIP cause 408 with 480 from the > other leg > 2009-11-14 09:35:52.330490 [DEBUG] mod_sofia.c:338 Channel sofia/internal/ > sip:1011 at 192.168.1.98:5872 hanging up, cause: NO_USER_RESPONSE > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/sip:1011 at 192.168.1.98:5872 Standard HANGUP, cause: > NO_USER_RESPONSE > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State HANGUP going to sleep > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:476 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_HANGUP -> > CS_REPORTING > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change > CS_REPORTING > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State REPORTING > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/sip:1011 at 192.168.1.98:5872 Standard REPORTING, cause: > NO_USER_RESPONSE > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State REPORTING going to sleep > > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:411 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_REPORTING -> > CS_DESTROY > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_session.c:1068 Session 542 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Locked, Waiting on external > entities > 2009-11-14 09:35:52.618414 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1012 at 192.168.1.98:5872 entering state [terminated][480] > 2009-11-14 09:35:52.618414 [NOTICE] sofia.c:3849 Hangup sofia/internal/ > sip:1012 at 192.168.1.98:5872 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > 2009-11-14 09:35:52.618414 [DEBUG] switch_channel.c:1683 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [KILL] > 2009-11-14 09:35:52.618414 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State CONSUME_MEDIA going to > sleep > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_HANGUP > > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State HANGUP > 2009-11-14 09:35:52.626426 [DEBUG] mod_sofia.c:306 sofia/internal/ > sip:1012 at 192.168.1.98:5872 Overriding SIP cause 408 with 480 from the > other leg > 2009-11-14 09:35:52.626426 [DEBUG] mod_sofia.c:338 Channel sofia/internal/ > sip:1012 at 192.168.1.98:5872 hanging up, cause: NO_USER_RESPONSE > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/sip:1012 at 192.168.1.98:5872 Standard HANGUP, cause: > NO_USER_RESPONSE > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State HANGUP going to sleep > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:476 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_HANGUP -> > CS_REPORTING > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change > CS_REPORTING > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State REPORTING > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/sip:1012 at 192.168.1.98:5872 Standard REPORTING, cause: > NO_USER_RESPONSE > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State REPORTING going to sleep > > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:411 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_REPORTING -> > CS_DESTROY > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_session.c:1068 Session 543 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Locked, Waiting on external > entities > 2009-11-14 09:35:59.778421 [DEBUG] sofia.c:3289 Channel sofia/internal/ > 4155559999 at 192.168.1.254 entering state [terminated][487] > 2009-11-14 09:35:59.778421 [NOTICE] sofia.c:3849 Hangup sofia/internal/ > 4155559999 at 192.168.1.254 [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2009-11-14 09:35:59.778421 [DEBUG] switch_channel.c:1683 Send signal > sofia/internal/4155559999 at 192.168.1.254 [KILL] > 2009-11-14 09:35:59.778421 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:59.798426 [NOTICE] switch_ivr_originate.c:1994 Hangup > sofia/internal/sip:1014 at 192.168.1.97:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2009-11-14 09:35:59.798426 [DEBUG] switch_channel.c:1683 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [KILL] > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] > 2009-11-14 09:35:59.798426 [DEBUG] switch_ivr_originate.c:2134 Originate > Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1086 Session 542 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Ended > 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/sip:1011 at 192.168.1.98:5872 [CS_DESTROY] > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:255 sofia/internal/ > sip:1011 at 192.168.1.98:5872 SOFIA DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/sip:1011 at 192.168.1.98:5872 Standard DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State DESTROY going to sleep > 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1086 Session 543 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Ended > 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/sip:1012 at 192.168.1.98:5872 [CS_DESTROY] > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:255 sofia/internal/ > sip:1012 at 192.168.1.98:5872 SOFIA DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/sip:1012 at 192.168.1.98:5872 Standard DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State DESTROY going to sleep > 2009-11-14 09:35:59.798426 [INFO] mod_dptools.c:2093 Originate Failed. > Cause: ORIGINATOR_CANCEL > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/4155559999 at 192.168.1.254) State EXECUTE going to sleep > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_HANGUP > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/4155559999 at 192.168.1.254) State HANGUP > 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:306 sofia/internal/ > 4155559999 at 192.168.1.254 Overriding SIP cause 487 with 487 from the other > leg > 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:338 Channel sofia/internal/ > 4155559999 at 192.168.1.254 hanging up, cause: ORIGINATOR_CANCEL > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/4155559999 at 192.168.1.254 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/4155559999 at 192.168.1.254) State HANGUP going to sleep > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State CONSUME_MEDIA going to > sleep > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_HANGUP > > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State HANGUP > 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:306 sofia/internal/ > sip:1014 at 192.168.1.97:5060 Overriding SIP cause 487 with 487 from the > other leg > 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:338 Channel sofia/internal/ > sip:1014 at 192.168.1.97:5060 hanging up, cause: ORIGINATOR_CANCEL > 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:406 Sending CANCEL to > sofia/internal/sip:1014 at 192.168.1.97:5060 > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/sip:1014 at 192.168.1.97:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State HANGUP going to sleep > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:476 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_HANGUP -> > CS_REPORTING > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change > CS_REPORTING > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State REPORTING > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/sip:1014 at 192.168.1.97:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State REPORTING going to sleep > > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:411 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_REPORTING -> > CS_DESTROY > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_session.c:1068 Session 544 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Locked, Waiting on external > entities > 2009-11-14 09:35:59.810601 [NOTICE] switch_core_session.c:1086 Session 544 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Ended > 2009-11-14 09:35:59.810601 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/sip:1014 at 192.168.1.97:5060 [CS_DESTROY] > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State DESTROY > 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:255 sofia/internal/ > sip:1014 at 192.168.1.97:5060 SOFIA DESTROY > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/sip:1014 at 192.168.1.97:5060 Standard DESTROY > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State DESTROY going to sleep > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:476 > (sofia/internal/4155559999 at 192.168.1.254) State Change CS_HANGUP -> > CS_REPORTING > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change > CS_REPORTING > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/4155559999 at 192.168.1.254) State REPORTING > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/4155559999 at 192.168.1.254 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/4155559999 at 192.168.1.254) State REPORTING going to sleep > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:411 > (sofia/internal/4155559999 at 192.168.1.254) State Change CS_REPORTING -> > CS_DESTROY > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_session.c:1068 Session 541 > (sofia/internal/4155559999 at 192.168.1.254) Locked, Waiting on external > entities > 2009-11-14 09:35:59.814986 [NOTICE] switch_core_session.c:1086 Session 541 > (sofia/internal/4155559999 at 192.168.1.254) Ended > 2009-11-14 09:35:59.814986 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/4155559999 at 192.168.1.254 [CS_DESTROY] > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/4155559999 at 192.168.1.254) State DESTROY > 2009-11-14 09:35:59.814986 [DEBUG] mod_sofia.c:255 sofia/internal/ > 4155559999 at 192.168.1.254 SOFIA DESTROY > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/4155559999 at 192.168.1.254 Standard DESTROY > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/4155559999 at 192.168.1.254) State DESTROY going to sleep > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091115/7bad9a71/attachment-0001.html From brian at freeswitch.org Sun Nov 15 14:14:08 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Nov 2009 16:14:08 -0600 Subject: [Freeswitch-users] Registration Error 408 In-Reply-To: <4B004E27.3040600@greatiam.com> References: <4AFF5558.3080408@greatiam.com> <34223AC5-699B-499B-A3B9-CED0F9CF1C59@freeswitch.org> <4B004E27.3040600@greatiam.com> Message-ID: <5E9EF7D5-2154-484E-9565-48A276A7D8A3@freeswitch.org> service iptables stop /b On Nov 15, 2009, at 12:53 PM, Samuel Abekah-Mensah wrote: > Hi Brian > > I have FS running on a FC11 box and the clients IDs 1001 and 1002 > running off 2 Windows boxes using X-lite3 > > Thanks > From ahmedmunir007 at gmail.com Sun Nov 15 23:29:50 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Mon, 16 Nov 2009 12:29:50 +0500 Subject: [Freeswitch-users] How to implement mod_lcr + mod_limit Message-ID: Hi, I've worked on setup for carriers routing using mod_lcr + mod_nibble + mod_xml_curl and mod_xml_cdr. The setup is working fine as I desired. Now I want to include mod_limit in to my setup. As I read the wiki pages of mod_limit I want to know how can I limit the calls per destination basis while running mod_lcr? Because LCR is routing to different carriers, how can I call mod_limit in mod_lcr? Kindly advise this issue soon. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/4dd76107/attachment.html From mustafa.pk at gmail.com Sun Nov 15 23:45:49 2009 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Mon, 16 Nov 2009 12:45:49 +0500 Subject: [Freeswitch-users] How to implement mod_lcr + mod_limit In-Reply-To: References: Message-ID: <8213d6070911152345k3f67fb93y730c46f085e386fc@mail.gmail.com> Ahmed, i hope you find the answer here. http://wiki.freeswitch.org/wiki/Mod_limit#limit_hash_execute -m On Mon, Nov 16, 2009 at 12:29 PM, Ahmed Munir wrote: > Hi, > > I've worked on setup for carriers routing using mod_lcr + mod_nibble + > mod_xml_curl and mod_xml_cdr. The setup is working fine as I desired. Now I > want to include mod_limit in to my setup. > > As I read the wiki pages of mod_limit I want to know how can I limit the > calls per destination basis while running mod_lcr? Because LCR is routing to > different carriers, how can I call mod_limit in mod_lcr? > > Kindly advise this issue soon. > > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com From vedamaker at netscape.net Mon Nov 16 00:29:36 2009 From: vedamaker at netscape.net (VedaM) Date: Mon, 16 Nov 2009 00:29:36 -0800 (PST) Subject: [Freeswitch-users] Problem with Siemens A580 IP Phones In-Reply-To: References: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> Message-ID: <26368294.post@talk.nabble.com> I discovered that the problem was due to having the incoming PSTN line connected to FS and the Siemens A580 base station. The Siemens A580 handsets were configured not to accept incoming calls from the PSTN but that does not seem work. In any case, I have disconnected the PSTN line from the Siemens A580 base station but I am still not able to use FS to transfer incoming calls between the Siemens A580 handsets. When I answer an incoming call with one handset (ext 1011) and use the "*1" DTMF command to transfer to another handset (ext 1012) the call is transferred to voicemail because the second handset (ext 1012) returns a 486 (Busy Here) status. Since you are using the same phones, can you tell me if you are able to use FS to transfer calls between handsets? If yes, how are you doing it? Your help is appreciated. Jo?o Mesquita-4 wrote: > > I have the same phone and all works fine. Make sure you are setting DTMF > to > RFC2833 on the phone config page. > > This log is supposed to go on pastebin (http://pastebin.freeswitch.org), > not > here. > > Also, you are making a group call which makes you ring 2 endpoints at the > same time. Verify your dialplan and make sure that's what you need/want. > > Regards, > > JM > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Problem-with-Siemens-A580-IP-Phones-tp26361779p26368294.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Mon Nov 16 00:57:53 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 16 Nov 2009 09:57:53 +0100 Subject: [Freeswitch-users] siptrace/debug log timestamp difference Message-ID: <65d96fc80911160057u1171bc41wc196d4719c10edc4@mail.gmail.com> Hi, just a thing i noticed... the debug log and sip trace have different time ... one hour difference ... looks like UTC/GMT issue. where do i set the time for siptrace correctly ? 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:411 (sofia/external/00010038516659280 at 10.4.5.107:5060) State Change CS_REPORTING -> CS_DESTROY 2009-11-16 09:47:13.779070 [DEBUG] switch_core_session.c:1068 Session 31 (sofia/external/00010038516659280 at 10.4.5.107:5060) Locked, Waiting on external entities 2009-11-16 09:47:13.779070 [NOTICE] switch_core_session.c:1086 Session 31 (sofia/external/00010038516659280 at 10.4.5.107:5060) Ended 2009-11-16 09:47:13.779070 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/00010038516659280 at 10.4.5.107:5060 [CS_DESTROY] 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:564 (sofia/external/00010038516659280 at 10.4.5.107:5060) State DESTROY 2009-11-16 09:47:13.779070 [DEBUG] mod_sofia.c:255 sofia/external/ 00010038516659280 at 10.4.5.107:5060 SOFIA DESTROY 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:60 sofia/external/00010038516659280 at 10.4.5.107:5060 Standard DESTROY 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:564 (sofia/external/00010038516659280 at 10.4.5.107:5060) State DESTROY going to sleep recv 462 bytes from udp/[10.4.5.107]:5060 at 08:47:13.799578: ------------------------------------------------------------------------ ACK sip:30003038515000403 at l01sipindir2.ot.hr:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.4.5.107:5060 ;branch=z9hG4bKterm-13e-30003038515000403-00010038516659280-59521 From: 00010038516659280 ;tag=261638185 To: 30003038515000403 ;tag=9v58macH5mNNH Call-ID: 3189ce3b-3da37db2-3ac943f-140 at 10.4.5.107 CSeq: 1 ACK Max-Forwards: 10 Content-Length: 0 ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/4cd45639/attachment.html From rupa at rupa.com Mon Nov 16 05:39:20 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 16 Nov 2009 07:39:20 -0600 Subject: [Freeswitch-users] Problem with Siemens A580 IP Phones In-Reply-To: <26368294.post@talk.nabble.com> References: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> <26368294.post@talk.nabble.com> Message-ID: Just use the siemens to do the transfer. Hit the menu key, select internal (if you want to do an siemens-siemens transfer), select the extension, hit talk, then menu and choose conference or transfer. If youw ant another extension, then choose external, dial the extension, and the rest is the same. On Mon, Nov 16, 2009 at 2:29 AM, VedaM wrote: > > > I discovered that the problem was due to having the incoming PSTN line > connected to FS and the Siemens A580 base station. ?The Siemens A580 > handsets were configured not to accept incoming calls from the PSTN but that > does not seem work. ?In any case, I have disconnected the PSTN line from the > Siemens A580 base station but I am still not able to use FS to transfer > incoming calls between the Siemens A580 handsets. ?When I answer an incoming > call with one handset (ext 1011) and use the "*1" DTMF command to transfer > to another handset (ext 1012) the call is transferred to voicemail because > the second handset (ext 1012) returns a 486 (Busy Here) status. ?Since you > are using the same phones, can you tell me if you are able to use FS to > transfer calls between handsets? ?If yes, how are you doing it? ?Your help > is appreciated. > > > > Jo?o Mesquita-4 wrote: >> >> I have the same phone and all works fine. Make sure you are setting DTMF >> to >> RFC2833 on the phone config page. >> >> This log is supposed to go on pastebin (http://pastebin.freeswitch.org), >> not >> here. >> >> Also, you are making a group call which makes you ring 2 endpoints at the >> same time. Verify your dialplan and make sure that's what you need/want. >> >> Regards, >> >> JM >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://old.nabble.com/Problem-with-Siemens-A580-IP-Phones-tp26361779p26368294.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From michaelt at voxcore.voxtelecom.co.za Sun Nov 15 23:07:01 2009 From: michaelt at voxcore.voxtelecom.co.za (Michael Toop) Date: Mon, 16 Nov 2009 09:07:01 +0200 Subject: [Freeswitch-users] DTMF Digits Lost when Under Load Message-ID: <330316f60911152307w2800f2e1r87c77d6dcd70be65@mail.gmail.com> Hi All, I have an issue that when my call volumes on my FS IVR box > 30 calls DTMF digits are lost (using RFC2833). It is definitely load related as it all works perfectly under 30 calls. Any pointers or a solution to the problem? Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/be8e93e6/attachment.html From anthony.minessale at gmail.com Mon Nov 16 07:25:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Nov 2009 09:25:51 -0600 Subject: [Freeswitch-users] DTMF Digits Lost when Under Load In-Reply-To: <330316f60911152307w2800f2e1r87c77d6dcd70be65@mail.gmail.com> References: <330316f60911152307w2800f2e1r87c77d6dcd70be65@mail.gmail.com> Message-ID: <191c3a030911160725k38ebcda8ta8c38c36eb80e627@mail.gmail.com> That's a pretty small problem description to be so sure about something. It would probably be better to capture some evidence of the exact problem you are having since we are using computers and we need to see the computers in action doing something specifically incorrect to diagnose any sort of problem. Take the time to describe the origin and destination of your calls, the call flow, the hardware in use on both ends of the call, detailed console logs on debug level, (maybe even uncomment the 2833 debug ifded in switch_rtp.c) and gather something to go on besides "I seem to be losing dtmf) maybe a packect capture of the networking interface on both ends of these calls. Also problems should be reported to http://jira.freeswitch.org not this mailing list. Save us a step if you report a jira and provide all the info above or we will just have to ask for it again. On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop < michaelt at voxcore.voxtelecom.co.za> wrote: > Hi All, > > I have an issue that when my call volumes on my FS IVR box > 30 calls DTMF > digits are lost (using RFC2833). It is definitely load related as it all > works perfectly under 30 calls. > > Any pointers or a solution to the problem? > > Thanks, > > Michael > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/a4927b96/attachment-0001.html From dschwartz at xconnect.net Mon Nov 16 07:45:49 2009 From: dschwartz at xconnect.net (David Schwartz) Date: Mon, 16 Nov 2009 17:45:49 +0200 Subject: [Freeswitch-users] gtalk Message-ID: <6EA53FAD386F9D46B97D49BFE148D5140603A069@ISR-JLM-MAIL1.xconnect.co.il> Thanks to everyone for all the help. I finally got gtalk to work in both directions - well almost. I have the gtalk client indicating that user 1000 1000 is trying to call him only there is no button on the gtalk client to answer the call. (when I call from a gtalk client to gtalk client there is an "answer" button). Has anyone encountered this problem? Only other weird part is that in the debug log I see: 2009-11-16 15:59:01.517850 [DEBUG] libdingaling.c:1228 sasl authentication failed 2009-11-16 15:59:01.517850 [DEBUG] libdingaling.c:1546 io error 2 7 retry in 20 second(s) even though I have connected to the XMPP server (the user I have in the client profile is "online" in the gtalk client) Don't know if these are related. Thanks, David From brian at freeswitch.org Mon Nov 16 08:20:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Nov 2009 10:20:20 -0600 Subject: [Freeswitch-users] gtalk In-Reply-To: <6EA53FAD386F9D46B97D49BFE148D5140603A069@ISR-JLM-MAIL1.xconnect.co.il> References: <6EA53FAD386F9D46B97D49BFE148D5140603A069@ISR-JLM-MAIL1.xconnect.co.il> Message-ID: I think you missed this step http://wiki.freeswitch.org/wiki/Dingaling#TLS So your dingaling is failing to work properly... :P /b On Nov 16, 2009, at 9:45 AM, David Schwartz wrote: > Thanks to everyone for all the help. > > I finally got gtalk to work in both directions - well almost. > > I have the gtalk client indicating that user 1000 1000 is trying to > call him only there is no button on the gtalk client to answer the > call. (when I call from a gtalk client to gtalk client there is an > "answer" button). > > Has anyone encountered this problem? > > Only other weird part is that in the debug log I see: > > 2009-11-16 15:59:01.517850 [DEBUG] libdingaling.c:1228 sasl > authentication failed > > 2009-11-16 15:59:01.517850 [DEBUG] libdingaling.c:1546 io error 2 7 > retry in 20 second(s) > > even though I have connected to the XMPP server (the user I have in > the client profile is "online" in the gtalk client) > > Don't know if these are related. > > Thanks, > > David From michaelt at voxcore.voxtelecom.co.za Mon Nov 16 09:20:11 2009 From: michaelt at voxcore.voxtelecom.co.za (Michael Toop) Date: Mon, 16 Nov 2009 19:20:11 +0200 Subject: [Freeswitch-users] DTMF Digits Lost when Under Load In-Reply-To: <191c3a030911160725k38ebcda8ta8c38c36eb80e627@mail.gmail.com> References: <330316f60911152307w2800f2e1r87c77d6dcd70be65@mail.gmail.com> <191c3a030911160725k38ebcda8ta8c38c36eb80e627@mail.gmail.com> Message-ID: <330316f60911160920n78edd236h82c6c1e7f71de1b6@mail.gmail.com> Hi Anthony, Thanks for the input. I will try & reproduce the problem & give you something more concrete to work with & log it in Jira. Thanks again, Michael On Mon, Nov 16, 2009 at 5:25 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > That's a pretty small problem description to be so sure about something. > It would probably be better to capture some evidence of the exact problem > you are having since we are using computers and we need to see the computers > in action doing something specifically incorrect to diagnose any sort of > problem. Take the time to describe the origin and destination of your > calls, the call flow, the hardware in use on both ends of the call, detailed > console logs on debug level, (maybe even uncomment the 2833 debug ifded in > switch_rtp.c) and gather something to go on besides "I seem to be losing > dtmf) maybe a packect capture of the networking interface on both ends of > these calls. > > Also problems should be reported to http://jira.freeswitch.org not this > mailing list. > Save us a step if you report a jira and provide all the info above or we > will just have to ask for it again. > > > On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop < > michaelt at voxcore.voxtelecom.co.za> wrote: > >> Hi All, >> >> I have an issue that when my call volumes on my FS IVR box > 30 calls >> DTMF digits are lost (using RFC2833). It is definitely load related as it >> all works perfectly under 30 calls. >> >> Any pointers or a solution to the problem? >> >> Thanks, >> >> Michael >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/dca6d724/attachment.html From msc at freeswitch.org Mon Nov 16 09:21:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 09:21:15 -0800 Subject: [Freeswitch-users] Registration Error - 408 timeout In-Reply-To: <4AFF5701.8010508@greatiam.com> References: <4AFF5701.8010508@greatiam.com> Message-ID: <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> On Sat, Nov 14, 2009 at 5:18 PM, Samuel Abekah-Mensah wrote: > Hello > > Please pardon me if the solution to this is somewhere already that I > have been unable to locate. I have just got a straight out-of-the-box > build of FS. According to the wiki, I should be able to test using user > IDs 1001 and 1002. However, I am get the above error. If I, however, > un-tick register with domain I do net get the error but does not > communicate either. Is there a conf that I should have done ? > > I am using X-lite3 > > Is NAT involved or are the x-lite clients on the same LAN? Also, you might want to turn on a SIP trace at the console to see if there are any clues. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/d27593ba/attachment.html From jerry.richards at teotech.com Mon Nov 16 09:36:01 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 16 Nov 2009 09:36:01 -0800 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> Message-ID: <1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> I have a bit of confusion about Lua scripting. When a script is invoked, should it always return an XML string that is used by FS? Or as in the case of dialplan examples, does it actually execute the dialplan (e.g. "session:answer();")? Best Regards, Jerry -----Original Message----- From: Leon de Rooij [mailto:leon at scarlet-internet.nl] Sent: Friday, November 13, 2009 2:29 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Accessing Config Info From Database Hi, You can use mod_xml_curl (generate xml on a webserver): http://wiki.freeswitch.org/wiki/Mod_xml_curl or mod_xml_odbc (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Mod_xml_odbc or LUA together with luasql (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration regards, Leon On Fri, 2009-11-13 at 13:59 -0800, Jerry Richards wrote: > Is there a way to access configuration information from a database (e.g. > SQL) rather than from the XML files? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From Prometheus001 at gmx.net Mon Nov 16 10:26:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 16 Nov 2009 19:26:25 +0100 Subject: [Freeswitch-users] att_xfer and Loopback In-Reply-To: <191c3a030911120750p34d27d44u2fec4015caf2f367@mail.gmail.com> References: <4AFB3A3D.1050602@gmx.net> <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> <191c3a030911112012i63000f3j9867308057c5f318@mail.gmail.com> <4AFBC98C.4070602@gmx.net> <191c3a030911120750p34d27d44u2fec4015caf2f367@mail.gmail.com> Message-ID: <4B019951.7070207@gmx.net> Hello Anthony, I made a console trace today: http://pastebin.freeswitch.org/11125 Different from the mail below, in this case A and C have voice. Best regards Peter Anthony Minessale schrieb: > if you provide a console trace of both situations with console > loglevel debug and put them on pastebin i can tell you what's happening. > > > On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX > wrote: > > Thanks Anthony, > > however this rather deteriorated the situation. > Now it works the following > - A calls B > - B enters *4 gets an announcement and enters digits for C (A get MOH) > - C is called > - As soon as C picks up the call, A and C both have no voice (and B is > dropped) > - When A hangs up, C hangs up > > Before it did: > - A calls B > - B enters *4 gets an announcement and enters digits for C (A get MOH) > - C is called > - As soon as C picks up the call, A and C are connected and B is > dropped > - When A hangs up, C hangs up > > Best regards > Peter > > Anthony Minessale schrieb: > > hit send too soon > > you want to set loopback_bowout=false > > > > This keeps loopback from trying to destroy itself when it sees a > > chance to cut out of the call path. > > > > > > On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale > > > >> wrote: > > > > > > set/export the channel variable loopback_bowout=true so it's on > > the loopback leg > > > > > > > > > > On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX > > > >> wrote: > > > > Hello, > > > > I have some problems with attended transfer and loopback > > > > Scenario how id work > > - A calls B > > - B enters *4 gets an announcement and enter digits for C (A > > get MOH) > > - C is called > > - As soon as C picks up the call, A and C are connected > and B > > is dropped > > > > How it work until here: > > - A calls B > > - B enters *4 gets an announcement and enter digits for C (A > > get MOH) > > - C is called > > - As soon as C picks up the call, B and C are connected (A > > still MOH) > > > > The dial string for C is dynamic and dependent on certain > > parameters, > > therefore C must be called via Loopback in our scenario. > > > > > > Here are the configs: > > In dialplan for calling B: > > > > > > Dialplan for executing the att_xfer: > > > > > expression="^attended_xfer$"> > > data="continue_on_fail=true"/> > > > > data="origination_cancel_key=#"/> > > > data="loopback/${attxfer_callthis}"/> > > > > > > > > So this is pretty standard, except the loopback. SVN is > 15322. > > > > Anybody has a solution for this? > > > > > > Best regards > > Peter > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Nov 16 11:17:04 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 11:17:04 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conf Call - Nov 20 Message-ID: <87f2f3b90911161117k1a0ef20cs80dda5d89beaefe5@mail.gmail.com> FYI, I've added the skeleton of the agenda for this week's call: http://wiki.freeswitch.org/wiki/FS_weekly_2009_11_20 The agendas have been pretty light lately. I would like everyone to think about questions that could be brought up for discussion. Also, I'd like to take this time to say thank you to the many folks who have signed up for the wiki lately and have been adding content. We've had quite a few people join over the past few weeks and they have been doing tweaks and adding content to the wiki. We definitely appreciate your help. If you would like to help out with anything else on the wiki (or other janitorial projects) please contact me off list. We have lots of wiki and JIRA things that folks can help with. Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/00d190fd/attachment.html From abeka at greatiam.com Mon Nov 16 11:31:58 2009 From: abeka at greatiam.com (Samuel 'Otis' Abekah-Mensah) Date: Mon, 16 Nov 2009 19:31:58 +0000 Subject: [Freeswitch-users] Registration Error - 408 timeout In-Reply-To: <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> Message-ID: <4B01A8AE.7070708@greatiam.com> Hello thanks so much. The machines are on the same lan , 2 have static IP with one on DHCP just for variation . I do get there errors on stating FS 1. Error stacksize too large 4194303 offers advise to run ./freeswitch -wate 2. Error checking for PMP [GENERAL ERROR] and 3. [WARNING] sofia_reg.c:1788: Can't register a pointer I do not know if any of this is could help The 2 boxes I run X-lite from are windows 2k service pack 4 Oh I ahve had a go and I am now getting Error 403 - Forbidden on the Xlite clients side. I have also tried using Zoiper but it seems to register but then comes up with an error "bearercapability " Thanks for your time, Michael and may thanks Brian. I am not sure if the iptables bit has caused the change from error 408 to error 403. Thanks; I apperecitae your help . Michael Collins wrote: > > > On Sat, Nov 14, 2009 at 5:18 PM, Samuel Abekah-Mensah > > wrote: > > Hello > > Please pardon me if the solution to this is somewhere already that I > have been unable to locate. I have just got a straight out-of-the-box > build of FS. According to the wiki, I should be able to test using > user > IDs 1001 and 1002. However, I am get the above error. If I, however, > un-tick register with domain I do net get the error but does not > communicate either. Is there a conf that I should have done ? > > I am using X-lite3 > > > Is NAT involved or are the x-lite clients on the same LAN? Also, you > might want to turn on a SIP trace at the console to see if there are > any clues. > -MC > From msc at freeswitch.org Mon Nov 16 11:33:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 11:33:09 -0800 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> <1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> Message-ID: <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards wrote: > > I have a bit of confusion about Lua scripting. When a script is invoked, > should it always return an XML string that is used by FS? Or as in the > case > of dialplan examples, does it actually execute the dialplan (e.g. > "session:answer();")? > > Best Regards, > Jerry > > Jerry, A Lua script that is explicitly called from the dialplan will indeed execute dialplan-ish stuff. For example, let's say you had this in conf/dialplan/default.xml: Then myluascript.lua has something like: --Sample Lua script session:answer() session:sleep(1000) session:streamFile("/path/to/file.wav") session:hangup() Assuming an otherwise default install, the above Lua script would execute when a caller dialed 9876, or if a call was x-ferred to 9876. However, if you're wanting to use Lua to serve up a dialplan then it's totally different. Lua is not called from the dialplan; Lua provides the dialplan to FreeSWITCH. This latter case is the scenario discussed in the wiki section you referenced. ( http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration) Are you trying to use Lua scripting for serving up a dynamic configuration of some sort? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/c73ec21e/attachment.html From leon at scarlet-internet.nl Mon Nov 16 14:02:53 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 16 Nov 2009 23:02:53 +0100 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> <1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> Message-ID: <1258408973.9730.91.camel@desk.bofh.scarlet-internet.nl> Hi, Since recently it's also possible to use lua *as* a dialplan: http://wiki.freeswitch.org/wiki/Mod_lua#For_dialplan regards, Leon On Mon, 2009-11-16 at 11:33 -0800, Michael Collins wrote: > > > On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards > wrote: > > I have a bit of confusion about Lua scripting. When a script > is invoked, > should it always return an XML string that is used by FS? Or > as in the case > of dialplan examples, does it actually execute the dialplan > (e.g. > "session:answer();")? > > Best Regards, > Jerry > > > Jerry, > > A Lua script that is explicitly called from the dialplan will indeed > execute dialplan-ish stuff. For example, let's say you had this in > conf/dialplan/default.xml: > > > > > > > > Then myluascript.lua has something like: > > --Sample Lua script > session:answer() > session:sleep(1000) > session:streamFile("/path/to/file.wav") > session:hangup() > > Assuming an otherwise default install, the above Lua script would > execute when a caller dialed 9876, or if a call was x-ferred to 9876. > > However, if you're wanting to use Lua to serve up a dialplan then it's > totally different. Lua is not called from the dialplan; Lua provides > the dialplan to FreeSWITCH. This latter case is the scenario discussed > in the wiki section you referenced. > (http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration) > > Are you trying to use Lua scripting for serving up a dynamic > configuration of some sort? > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Mon Nov 16 15:05:29 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 17 Nov 2009 10:05:29 +1100 Subject: [Freeswitch-users] TLS support on debian lenny Message-ID: <33c87fa30911161505y6b59312cm2d631dae65cb531d@mail.gmail.com> Hi! I am trying to enable SSL support in FS. I have followed the wiki at http://wiki.freeswitch.org/wiki/SIP_TLS I already had libssl-dev installed, so I thought support should already have been compiled into FS, however enabling Internal_ssl_enable=true in vars.xml results in FS internal profile to not start: 2009-11-17 09:31:48.593240 [NOTICE] sofia.c:3016 Started Profile internal [sofia_reg_internal] 2009-11-17 09:31:48.907740 [ERR] sofia.c:1006 Error Creating SIP UA for profile: internal Checking freeswitch/libs/sofia-sip/config.log I see the following, which I assume means TLS has not been compiled with support: configure:27892: checking openssl/tls1.h usability configure:27909: gcc -c -DSU_DEBUG=0 -g -ggdb conftest.c >&5 conftest.c:156:26: error: openssl/tls1.h: No such file or directory What package should I have installed prior to compiling FS on debian? There is no OpenSSL-Dev. Is it libcurl4-openssl-dev? Thanks From timuckun at gmail.com Mon Nov 16 15:07:18 2009 From: timuckun at gmail.com (Tim Uckun) Date: Tue, 17 Nov 2009 12:07:18 +1300 Subject: [Freeswitch-users] 1.05 Message-ID: <855e4dcf0911161507j4d03ed3fof9bed52926c6bcbf@mail.gmail.com> Where is 1.05? The trunk? Is trunk stable? Thanks. From msc at freeswitch.org Mon Nov 16 15:16:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 15:16:17 -0800 Subject: [Freeswitch-users] 1.05 In-Reply-To: <855e4dcf0911161507j4d03ed3fof9bed52926c6bcbf@mail.gmail.com> References: <855e4dcf0911161507j4d03ed3fof9bed52926c6bcbf@mail.gmail.com> Message-ID: <87f2f3b90911161516s485c055cyd85935295b2c8d32@mail.gmail.com> We are still working on 1.0.5. Right now the best place to be is that latest trunk. More information is forthcoming... -MC On Mon, Nov 16, 2009 at 3:07 PM, Tim Uckun wrote: > Where is 1.05? The trunk? Is trunk stable? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/e2b2bc30/attachment.html From brian at freeswitch.org Mon Nov 16 15:22:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Nov 2009 17:22:15 -0600 Subject: [Freeswitch-users] 1.05 In-Reply-To: <855e4dcf0911161507j4d03ed3fof9bed52926c6bcbf@mail.gmail.com> References: <855e4dcf0911161507j4d03ed3fof9bed52926c6bcbf@mail.gmail.com> Message-ID: <38AA65CA-FCB4-443F-A249-E8CC187356A0@freeswitch.org> Tim, 1.0.5 is coming soon... We were ready to release on the tuesday morning we said but we woke up and Jira was flooded with tons of new issues half of which we asked for more info on and the reporters aren't responding. So the key is if you open a jira be ready to respond because I'm not going to chase people down anymore. On that note we need more people to help out on Jira... All it takes is asking questions and trying to reproduce things that are reported thats the hard part... Once something can be reproduced reliably we can fix it faster. Thanks, /b On Nov 16, 2009, at 5:07 PM, Tim Uckun wrote: > Where is 1.05? The trunk? Is trunk stable? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From tina at a2unlimited.com Mon Nov 16 15:22:01 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Mon, 16 Nov 2009 18:22:01 -0500 Subject: [Freeswitch-users] ESL: No matching function... Message-ID: <673dfcbcbab316d312ea4ae87d13418c.squirrel@emailmg.ipower.com> I have three FreeSWITCH servers currently setup with perl modules using ESL to send call instructions and monitor events. On two of the servers, my modules execute without error, but on a third, I keep getting the following error: No matching function for overloaded 'new_ESLconnection' at /usr/lib64/perl5/site_perl/5.8.8/x86_64-linux-thread-multi/ESL.pm line 116. Is this something in ESL that I'm doing wrong, or is it an issue related to the perl configuration on the server? As far as I can tell, the three servers have been setup the same. - Tina From msc at freeswitch.org Mon Nov 16 15:42:06 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 15:42:06 -0800 Subject: [Freeswitch-users] ESL: No matching function... In-Reply-To: <673dfcbcbab316d312ea4ae87d13418c.squirrel@emailmg.ipower.com> References: <673dfcbcbab316d312ea4ae87d13418c.squirrel@emailmg.ipower.com> Message-ID: <87f2f3b90911161542k4490dc0bx6218122c31728cb@mail.gmail.com> On Mon, Nov 16, 2009 at 3:22 PM, wrote: > I have three FreeSWITCH servers currently setup with perl modules using > ESL to send call instructions and monitor events. On two of the servers, > my modules execute without error, but on a third, I keep getting the > following error: > > No matching function for overloaded 'new_ESLconnection' at > /usr/lib64/perl5/site_perl/5.8.8/x86_64-linux-thread-multi/ESL.pm line > 116. > > Is this something in ESL that I'm doing wrong, or is it an issue related > to the perl configuration on the server? > > As far as I can tell, the three servers have been setup the same. > > - Tina > > Can you cd into libs/esl and do "make && make perlmod" and see if any errors show up? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/61ce07b3/attachment.html From msc at freeswitch.org Mon Nov 16 16:25:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 16:25:09 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.0.5 Status Update Message-ID: <87f2f3b90911161625yb69f604i4b19f7f6a6a31800@mail.gmail.com> Hello folks! I just wanted to let everyone know that there is a status update on the main FreeSWITCH page. Here's a quick link for your convenience: http://bit.ly/3RSY9F The abridged version is this: we're working on it, there are some outstanding JIRA reports that people need to review & test, and we would appreciate more people updating to the latest trunk and making sure all is well. Thanks for helping! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/506f467d/attachment.html From mcampbellsmith at gmail.com Mon Nov 16 19:15:35 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 17 Nov 2009 14:15:35 +1100 Subject: [Freeswitch-users] TLS support on debian lenny In-Reply-To: <33c87fa30911161505y6b59312cm2d631dae65cb531d@mail.gmail.com> References: <33c87fa30911161505y6b59312cm2d631dae65cb531d@mail.gmail.com> Message-ID: <33c87fa30911161915p3a2a082ame20d5427ea0bd4c4@mail.gmail.com> I installed libcurl4-openssl-dev, but this automatically removed libcurl4-gnutls-dev, which is required by mod_dingaling. Now mod_dingaling fails to build with: Compiling mod_dingaling.c ... mod_dingaling.c:309:78: error: macro "switch_odbc_handle_callback_exec" requires 5 arguments, but only 4 given mod_dingaling.c: In function ???mdl_execute_sql_callback???: mod_dingaling.c:309: error: ???switch_odbc_handle_callback_exec??? undeclared (first use in this function) mod_dingaling.c:309: error: (Each undeclared identifier is reported only once mod_dingaling.c:309: error: for each function it appears in.) make[6]: *** [mod_dingaling.lo] Error 1 Anyone know which package should be installed so that TLS works on Debian? On Tue, Nov 17, 2009 at 10:05 AM, Mark Campbell-Smith wrote: > Hi! > > I am trying to enable SSL support in FS. ?I have followed the wiki at > http://wiki.freeswitch.org/wiki/SIP_TLS > > I already had libssl-dev installed, so I thought support should > already have been compiled into FS, however enabling > Internal_ssl_enable=true in vars.xml results in FS internal profile to > not start: > > 2009-11-17 09:31:48.593240 [NOTICE] sofia.c:3016 Started Profile > internal [sofia_reg_internal] > 2009-11-17 09:31:48.907740 [ERR] sofia.c:1006 Error Creating SIP UA > for profile: internal > > Checking freeswitch/libs/sofia-sip/config.log I see the following, > which I assume means TLS has not been compiled with support: > configure:27892: checking openssl/tls1.h usability > configure:27909: gcc -c ?-DSU_DEBUG=0 -g -ggdb ?conftest.c >&5 > conftest.c:156:26: error: openssl/tls1.h: No such file or directory > > What package should I have installed prior to compiling FS on debian? > There is no OpenSSL-Dev. ?Is it libcurl4-openssl-dev? > > Thanks > From brian at freeswitch.org Mon Nov 16 19:24:43 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Nov 2009 21:24:43 -0600 Subject: [Freeswitch-users] TLS support on debian lenny In-Reply-To: <33c87fa30911161915p3a2a082ame20d5427ea0bd4c4@mail.gmail.com> References: <33c87fa30911161505y6b59312cm2d631dae65cb531d@mail.gmail.com> <33c87fa30911161915p3a2a082ame20d5427ea0bd4c4@mail.gmail.com> Message-ID: <2503E4FE-A1DB-486D-B818-CD9B04D3D955@freeswitch.org> Mark update and try again... we did some changes to the core odbc stuff today and waiting on the dust to settle. Thanks, Brian On Nov 16, 2009, at 9:15 PM, Mark Campbell-Smith wrote: > I installed libcurl4-openssl-dev, but this automatically removed > libcurl4-gnutls-dev, which is required by mod_dingaling. Now > mod_dingaling fails to build with: > Compiling mod_dingaling.c ... > mod_dingaling.c:309:78: error: macro > "switch_odbc_handle_callback_exec" requires 5 arguments, but only 4 > given > mod_dingaling.c: In function ???mdl_execute_sql_callback???: > mod_dingaling.c:309: error: ???switch_odbc_handle_callback_exec??? > undeclared (first use in this function) > mod_dingaling.c:309: error: (Each undeclared identifier is reported > only once > mod_dingaling.c:309: error: for each function it appears in.) > make[6]: *** [mod_dingaling.lo] Error 1 > > Anyone know which package should be installed so that TLS works on > Debian? > > On Tue, Nov 17, 2009 at 10:05 AM, Mark Campbell-Smith > wrote: >> Hi! >> >> I am trying to enable SSL support in FS. I have followed the wiki at >> http://wiki.freeswitch.org/wiki/SIP_TLS >> >> I already had libssl-dev installed, so I thought support should >> already have been compiled into FS, however enabling >> Internal_ssl_enable=true in vars.xml results in FS internal profile >> to >> not start: >> >> 2009-11-17 09:31:48.593240 [NOTICE] sofia.c:3016 Started Profile >> internal [sofia_reg_internal] >> 2009-11-17 09:31:48.907740 [ERR] sofia.c:1006 Error Creating SIP UA >> for profile: internal >> >> Checking freeswitch/libs/sofia-sip/config.log I see the following, >> which I assume means TLS has not been compiled with support: >> configure:27892: checking openssl/tls1.h usability >> configure:27909: gcc -c -DSU_DEBUG=0 -g -ggdb conftest.c >&5 >> conftest.c:156:26: error: openssl/tls1.h: No such file or directory >> >> What package should I have installed prior to compiling FS on debian? >> There is no OpenSSL-Dev. Is it libcurl4-openssl-dev? >> >> Thanks >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From elihayun at gmail.com Tue Nov 17 01:03:48 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 17 Nov 2009 11:03:48 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? Message-ID: <4B0266F4.8070602@savion.huji.ac.il> Hi We have more then one profile. To make a call I have to enter : bridge sofia/profile/number at ip The problem is when I use : "${use_profile}" I am getting the caller profile, and I need the destination profile. How do I get this information? Thanks Eli From dujinfang at gmail.com Tue Nov 17 04:17:15 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 17 Nov 2009 20:17:15 +0800 Subject: [Freeswitch-users] prefix Freeswitch-users vs. FreeSWITCH-Users Message-ID: <23f91030911170417y2e857124m5796565d5f24b329@mail.gmail.com> Would it be better to change the list subject prefix from [Freeswitch-users] to FreeSWITCH-Users? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/61b92fe8/attachment.html From mattdfong at gmail.com Tue Nov 17 05:28:46 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 17 Nov 2009 20:28:46 +0700 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel Message-ID: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> I'm trying performing a uuid_record command immediately after a uuid_bridge, but receive a "Can not record session. Media not enabled on channel" error. proxy_media and bypass_media are both set to false. The uuid_record however works if I use sched_api +1 uuid_record... but if I do this, I of course loose the first second of conversation. Does anyone have any ideas on how I might be able to solve this? I've turned on DEBUG mode, but nothing out of the ordinary appears. http://pastebin.freeswitch.org/11141 --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/f0c222f1/attachment.html From regs at kinetix.gr Tue Nov 17 05:42:01 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 17 Nov 2009 15:42:01 +0200 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp Message-ID: <4B02A829.7080708@kinetix.gr> I am trying to use switch_r_sdp to rewrite the SDP. The problem I am facing has to do with the way of doing it. Let's say I have: v=0 o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx s=Opal SIP Session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 5144 RTP/AVP 18 3 101 120 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:120 NSE/8000 a=fmtp:120 192-193 who to I set the switch_r_sdp variable in xml? Obviously this doesn't work : Do I have to escape any special characters and how? I tried using escaped quotes, escaped spaces, escaped tabs etc. Nothing worked. Any suggestions? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From nicolas at medularis.com Tue Nov 17 06:18:16 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 17 Nov 2009 11:18:16 -0300 Subject: [Freeswitch-users] need desperate help with zombie channels In-Reply-To: <27c25bc40911130513u405062c7kb775e14d04761fd4@mail.gmail.com> References: <27c25bc40911130513u405062c7kb775e14d04761fd4@mail.gmail.com> Message-ID: <1b46b4e80911170618x669312cdrc334067eb5c20ec4@mail.gmail.com> Hi Juan, A similar thing happened to me. I was creating channels and bridging them with a JS script. I had to add a session.hangup(); statement at the end of the script. That solved my problem. Cheers, Nico On Fri, Nov 13, 2009 at 10:13 AM, Juan Backson wrote: > Hi, > > I am having difficulty trying to figure out why there are bunch of zombie > channels in my system. It seems to me that these zombies come from > apr_thread pool. > > Does anyone have any idea what may be the cause of these problems? > > > freeswitch at internal> show channels > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure > b789468a-4412-490b-bc66-32f149ba4d1d,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 > ,,,XML,default,,,,, > 7e1ecaaa-b2d8-47a0-9982-25cd44186d4e,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 > ,,,XML,default,,,,, > 01fa2ff6-f807-4ef0-b988-70a9fe8c4536,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_EXCHANGE_MEDIA,a88999001,a88999001,192.168.1.116, > 999100 at 192.168.1.116:9342,incre_call_stat,125 165 182 235 13 3184093 > 0,XML,default,,,,, > 0271541f-f0b5-482c-b05d-b196f85121be,inbound,2009-11-13 > 20:15:35,1258114535,sofia/external/88999001 at 192.168.1.116:7342 > ,CS_EXECUTE,sipp,88999001,192.168.1.116,88999100,hangup,NORMAL_CLEARING,XML,default,,,,, > 7e4ccfec-a4ad-4817-9a82-f1166b34576f,outbound,2009-11-13 > 20:15:35,1258114536,sofia/external/999100 at 192.168.1.116:9342 > ,CS_CONSUME_MEDIA,a88999001,a88999001,192.168.1.116, > 999100 at 192.168.1.116:9342,,,XML,default,,,,, > > 5 total. > > freeswitch at internal> uuid_kill b789468a-4412-490b-bc66-32f149ba4d1d > -ERR No Such Channel! > > These channels actually do not exist in the system! > > > Here is my gcore output with 5 zombies out of 100K test calls : > > > Thread 21 (process 8946): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 128000} > #2 0x00002b3cb3bfb8ca in switch_console_loop () at > src/switch_console.c:819 > arg = 1 > thread = (switch_thread_t *) 0x2aaab00320d0 > thd_attr = (switch_threadattr_t *) 0x2aaab0032070 > pool = (switch_memory_pool_t *) 0x2aaab0031f88 > __func__ = "switch_console_loop" > __PRETTY_FUNCTION__ = "switch_console_loop" > #3 0x0000000000402884 in main (argc=1, argv=) > at src/switch.c:753 > pid_path = "/usr/local/freeswitch/log/freeswitch.pid", '\0' > > pid_buffer = "8946", '\0' > old_pid_buffer = '\0' > pid_len = 4 > old_pid_len = 4198811 > err = 0x2b3cb3cec77d "Success" > ---Type to continue, or q to quit--- > nf = 0 > runas_user = > runas_group = > nc = 0 > pid = > x = > opts = > opts_str = '\0' > local_argv = {0x7ffff6f08c15 "./freeswitch", 0x0 times>} > arg_argv = {0x0 } > alt_dirs = 0 > known_opt = > high_prio = 0 > flags = 65 > ret = > destroy_status = > fd = (switch_file_t *) 0xb6293e0 > pool = (switch_memory_pool_t *) 0xb629368 > rlp = {rlim_cur = 245760, rlim_max = 245760} > waste = 0 > __PRETTY_FUNCTION__ = "main" > > Thread 20 (process 20699): > ---Type to continue, or q to quit--- > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002aaaab35e926 in read_packet (listener=0x2aaae7523d08, > event=0x2aab3b5ab058, timeout=0) at mod_event_socket.c:1255 > do_sleep = 1 '\001' > mlen = 0 > bytes = 0 > mbuf = '\0' > buf = '\0' > len = 123 > status = SWITCH_STATUS_BREAK > count = > start = 1258117263 > pop = (void *) 0x2aaad12f6540 > ptr = 0x2aab3b5a98a0 "" > crcount = 0 '\0' > channel = (switch_channel_t *) 0x0 > clen = > __func__ = "read_packet" > __PRETTY_FUNCTION__ = "read_packet" > ---Type to continue, or q to quit--- > #3 0x00002aaaab36347a in listener_run (thread=, > obj=0x2aaae7523d08) at mod_event_socket.c:2093 > listener = (listener_t *) 0x0 > buf = '\0' > len = 1024 > status = > event = (switch_event_t *) 0x0 > reply = "\000OK log level [7]", '\0' > session = (switch_core_session_t *) 0x0 > channel = > revent = (switch_event_t *) 0x0 > var = > __PRETTY_FUNCTION__ = "listener_run" > __func__ = "listener_run" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 19 (process 14505): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaaf49798, > data=0x7afe0080) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( > thread=, obj=) > at src/switch_event.c:248 > pop = (void *) 0x0 > event = (switch_event_t *) 0x0 > queue = (switch_queue_t *) 0x2aaaaaf49798 > my_id = 1 > __func__ = "switch_event_dispatch_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 18 (process 9334): > #0 0x0000003054e0d2cb in read () from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3cd50c8 in read_char (el=0x2aaab0028180, cp=0x4027002f "") > at read.c:294 > num_read = 1076297860 > tried = 0 > ---Type to continue, or q to quit--- > #2 0x00002b3cb3cd4ceb in el_gets (el=0x2aaab0028180, nread=0x40270084) > at read.c:241 > cmdnum = 112 'p' > num = -1321754256 > ch = 0 '\0' > #3 0x00002b3cb3bfc4bb in console_thread (thread=, > obj=) at src/switch_console.c:464 > arg = 1 > count = 1 > line = 0x2aaab0034e70 "\n" > pool = (switch_memory_pool_t *) 0x2aaab0031f88 > __func__ = "console_thread" > __PRETTY_FUNCTION__ = "console_thread" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 17 (process 9333): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > ---Type to continue, or q to quit--- > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002b3cb3c53895 in softtimer_runtime () at src/switch_time.c:464 > current_ms = 692 > x = 690 > tick = 292 > ts = > last = 1258117283599783 > fwd_errs = 0 > rev_errs = 0 > __func__ = "softtimer_runtime" > #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x0, obj=0x0) > at src/switch_loadable_module.c:94 > status = > ts = (switch_core_thread_session_t *) 0x0 > module = (switch_loadable_module_t *) 0xb6c4e00 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 16 (process 9332): > ---Type to continue, or q to quit--- > #0 0x0000003054e0d4eb in accept () from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c707a4 in apr_socket_accept (new=0x416b4020, > sock=0xbcfde38, > connection_context=0x2aaacda27718) at network_io/unix/sockets.c:187 > No locals. > #2 0x00002aaaab35f889 in mod_event_socket_runtime () > at mod_event_socket.c:2324 > pool = (switch_memory_pool_t *) 0xbcfdc88 > listener_pool = (switch_memory_pool_t *) 0x2aaacda27718 > rv = > sa = (switch_sockaddr_t *) 0xbcfdd68 > inbound_socket = (switch_socket_t *) 0x2aaacda277f8 > listener = > x = > __func__ = "mod_event_socket_runtime" > #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x14f, > obj=0x2aaacda27948) at src/switch_loadable_module.c:94 > status = > ts = (switch_core_thread_session_t *) 0x2aaacda27948 > module = (switch_loadable_module_t *) 0x2aaaac0058c0 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > ---Type to continue, or q to quit--- > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 15 (process 9330): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 55000} > #2 0x00002aaab503cc4c in node_thread_run (thread=, > obj=) at mod_fifo.c:580 > val = (void *) 0x0 > var = (const void *) 0x0 > idle_consumers = > hi = (switch_hash_index_t *) 0x0 > ppl_waiting = 0 > consumer_total = 1087699264 > node = > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > ---Type to continue, or q to quit--- > > Thread 14 (process 9329): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100} > #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( > thread=, obj=) at sofia.c:763 > profile = (sofia_profile_t *) 0xbce2310 > ireg_loops = 18 > gateway_loops = 0 > loops = 72 > qsize = 4294966782 > pop = (void *) 0x0 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 13 (process 9328): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > ---Type to continue, or q to quit--- > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( > thread=, obj=) at sofia.c:763 > profile = (sofia_profile_t *) 0x2aaab000eb10 > ireg_loops = 5 > gateway_loops = 0 > loops = 93 > qsize = 4294966782 > pop = (void *) 0x0 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 12 (process 9327): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce71c0, > tout=1000) > at su_epoll_port.c:495 > ---Type to continue, or q to quit--- > j = 198076976 > n = 0 > events = 0 > index = 10922 > version = 3 > M = 4 > ev = 0x41204ef0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce71c0) > at su_base_port.c:349 > tout = 1000 > tout2 = 0 > __PRETTY_FUNCTION__ = "su_base_port_run" > #3 0x00002aaab45c6c51 in su_port_run (self=0xbce71c0) at su_port.h:326 > base = (su_virtual_port_t *) 0xbce71c0 > #4 0x00002aaab45c6c29 in su_root_run (self=0xbce72a0) at su_root.c:819 > __PRETTY_FUNCTION__ = "su_root_run" > #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404f7ac0) > at su_pthread_port.c:324 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0xbce71c0, sut_root = 0xbce72a0}} > zap = 1 > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > ---Type to continue, or q to quit--- > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 11 (process 9326): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce78b0, > tout=1000) > at su_epoll_port.c:495 > j = -1342070512 > n = 10922 > events = 0 > index = 10922 > version = 3 > M = 4 > ev = 0x411c8ef0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce78b0) > at su_base_port.c:349 > tout = 1000 > tout2 = 0 > __PRETTY_FUNCTION__ = "su_base_port_run" > #3 0x00002aaab45c6c51 in su_port_run (self=0xbce78b0) at su_port.h:326 > ---Type to continue, or q to quit--- > base = (su_virtual_port_t *) 0xbce78b0 > #4 0x00002aaab45c6c29 in su_root_run (self=0x2aaab001a060) at > su_root.c:819 > __PRETTY_FUNCTION__ = "su_root_run" > #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404bbac0) > at su_pthread_port.c:324 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0xbce78b0, sut_root = 0x2aaab001a060}} > zap = 1 > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 10 (process 9325): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce6c30, > tout=1000) > at su_epoll_port.c:495 > j = -1268971119 > n = 10922 > events = 0 > index = 0 > version = 1 > ---Type to continue, or q to quit--- > M = 4 > ev = 0x404f7c40 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbce6c30, tout=1000) > at su_base_port.c:467 > now = {tv_sec = 3467106082, tv_usec = 971475} > __PRETTY_FUNCTION__ = "su_base_port_step" > #3 0x00002aaab45c6d6a in su_port_step (self=0xbce6c30, tout=1000) > at su_port.h:340 > base = (su_virtual_port_t *) 0xbce6c30 > #4 0x00002aaab45c6d32 in su_root_step (self=0xbce4650, tout=1000) > at su_root.c:858 > __PRETTY_FUNCTION__ = "su_root_step" > #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( > thread=, obj=) at sofia.c:973 > profile = (sofia_profile_t *) 0xbce2310 > pool = > node = (sip_alias_node_t *) 0x0 > s_event = (switch_event_t *) 0x0 > sanity = > worker_thread = (switch_thread_t *) 0xbce36a0 > st = SWITCH_STATUS_SUCCESS > __func__ = "sofia_profile_thread_run" > ---Type to continue, or q to quit--- > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 9 (process 9324): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbcdffb0, > tout=1000) > at su_epoll_port.c:495 > j = -1268971119 > n = 10922 > events = 0 > index = 0 > version = 1 > M = 4 > ev = 0x404bbc40 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbcdffb0, tout=1000) > at su_base_port.c:467 > now = {tv_sec = 3467106083, tv_usec = 525146} > __PRETTY_FUNCTION__ = "su_base_port_step" > ---Type to continue, or q to quit--- > #3 0x00002aaab45c6d6a in su_port_step (self=0xbcdffb0, tout=1000) > at su_port.h:340 > base = (su_virtual_port_t *) 0xbcdffb0 > #4 0x00002aaab45c6d32 in su_root_step (self=0xbcdfe00, tout=1000) > at su_root.c:858 > __PRETTY_FUNCTION__ = "su_root_step" > #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( > thread=, obj=) at sofia.c:973 > profile = (sofia_profile_t *) 0x2aaab000eb10 > pool = > node = (sip_alias_node_t *) 0x0 > s_event = (switch_event_t *) 0x0 > sanity = > worker_thread = (switch_thread_t *) 0x2aaab000fea0 > st = SWITCH_STATUS_SUCCESS > __func__ = "sofia_profile_thread_run" > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 8 (process 8999): > ---Type to continue, or q to quit--- > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 444000} > #2 0x00002b3cb3c14e2a in switch_scheduler_task_thread ( > thread=, obj=) > at src/switch_scheduler.c:171 > __func__ = "switch_scheduler_task_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 7 (process 8998): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100} > #2 0x00002b3cb3c054f5 in switch_core_sql_thread ( > thread=, obj=) > at src/switch_core_sqldb.c:220 > ---Type to continue, or q to quit--- > pop = (void *) 0x2aaabf3d6220 > itterations = 0 > trans = 0 '\0' > nothing_in_queue = 1 '\001' > len = 100 > sql_len = 4844546 > sqlbuf = 0x2aab135c7010 "" > sql = > newlen = > lc = 0 > __PRETTY_FUNCTION__ = "switch_core_sql_thread" > __func__ = "switch_core_sql_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 6 (process 8995): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb64c158, data=0x40893088) > at misc/apr_queue.c:276 > ---Type to continue, or q to quit--- > rv = 0 > #2 0x00002b3cb3c48ff1 in log_thread (t=, > obj=) at src/switch_log.c:288 > pop = (void *) 0x0 > node = (switch_log_node_t *) 0x0 > binding = (switch_log_binding_t *) 0x0 > __func__ = "log_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 5 (process 8951): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaac355a8, > data=0x40bec070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > ---Type to continue, or q to quit--- > queue = (switch_queue_t *) 0x2aaaaac355a8 > index = 0 > my_id = 2 > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 4 (process 8950): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaab705a8, > data=0x4060a070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > queue = (switch_queue_t *) 0x2aaaaab705a8 > index = 0 > my_id = 1 > ---Type to continue, or q to quit--- > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 3 (process 8949): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb638fa8, data=0x405ce070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > queue = (switch_queue_t *) 0xb638fa8 > index = 0 > my_id = 0 > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 2 (process 8948): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaacfa5a8, > data=0x40592080) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( > thread=, obj=) > at src/switch_event.c:248 > pop = (void *) 0x0 > event = (switch_event_t *) 0x0 > queue = (switch_queue_t *) 0x2aaaaacfa5a8 > my_id = 0 > __func__ = "switch_event_dispatch_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > ---Type to continue, or q to quit--- > Thread 1 (process 8947): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 451000} > #2 0x00002b3cb3c00c95 in pool_thread (thread=, > obj=) at src/switch_core_memory.c:490 > x = > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > (gdb) > (gdb) > (gdb) > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/75b3fb79/attachment-0001.html From brian at freeswitch.org Tue Nov 17 06:41:19 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 08:41:19 -0600 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp In-Reply-To: <4B02A829.7080708@kinetix.gr> References: <4B02A829.7080708@kinetix.gr> Message-ID: <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> Why are you needing to rewrite it? /b On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: > > I am trying to use switch_r_sdp to rewrite the SDP. > The problem I am facing has to do with the way of doing it. > > Let's say I have: > > v=0 > o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx > s=Opal SIP Session > c=IN IP4 xxx.xxx.xxx.xxx > t=0 0 > m=audio 5144 RTP/AVP 18 3 101 120 > c=IN IP4 xxx.xxx.xxx.xxx > a=rtpmap:18 G729/8000/1 > a=fmtp:18 annexb=no > a=rtpmap:3 gsm/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16,32,36 > a=rtpmap:120 NSE/8000 > a=fmtp:120 192-193 > > who to I set the switch_r_sdp variable in xml? > > Obviously this doesn't work : > > > > Do I have to escape any special characters and how? > I tried using escaped quotes, escaped spaces, escaped tabs etc. > Nothing worked. > > Any suggestions? > > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Tue Nov 17 06:43:15 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 08:43:15 -0600 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: <4B0266F4.8070602@savion.huji.ac.il> References: <4B0266F4.8070602@savion.huji.ac.il> Message-ID: Why do you need to know the destination profile like that? You get to pick that on your own so you should already know that before hand. /b On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote: > Hi > We have more then one profile. To make a call I have to enter : bridge > sofia/profile/number at ip > The problem is when I use : "${use_profile}" I am getting the caller > profile, and I need the destination profile. > > How do I get this information? > > Thanks > > Eli From abeka at greatiam.com Tue Nov 17 07:12:19 2009 From: abeka at greatiam.com (Sam Abekah-Mensah) Date: Tue, 17 Nov 2009 15:12:19 +0000 Subject: [Freeswitch-users] Registration Error - 408 timeout and now 403 In-Reply-To: <4B01A8AE.7070708@greatiam.com> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> <4B01A8AE.7070708@greatiam.com> Message-ID: <4B02BD53.5040203@greatiam.com> Hello I have tried the same setup but this time using a windows build FS1.0.4 on an XP machine and all is fine. The sample 1001 and 1002 IDs work without any tweaking at all. Could the problem be with the linux build 1.0.4.? I am running on an FC11 machine. On the FC11 box I used the svn link to build using the ff: bootstarp.sh configure withoout libcurl to eliinate the spidermonkey lib error make make install Did I miss anything ? In a nutshell I can get the sample test to work using a windows-based version Thanks d Samuel 'Otis' Abekah-Mensah wrote: >
Hello > > thanks so much. The machines are on the same lan , 2 have static IP > with one on DHCP just for variation . I do get there errors on stating FS > 1. Error stacksize too large 4194303 offers advise to run ./freeswitch > -wate > 2. Error checking for PMP [GENERAL ERROR] > and > 3. [WARNING] sofia_reg.c:1788: Can't register a pointer > I do not know if any of this is could help The 2 boxes I run X-lite > from are windows 2k service pack 4 > > Oh I ahve had a go and I am now getting Error 403 - Forbidden on the > Xlite clients side. > > I have also tried using Zoiper but it seems to register but then comes > up with an error "bearercapability " > > Thanks for your time, Michael and may thanks Brian. I am not sure if > the iptables bit has caused the change from error 408 to error 403. > > Thanks; I apperecitae your help > . > > > > Michael Collins wrote: >> >> >> On Sat, Nov 14, 2009 at 5:18 PM, Samuel Abekah-Mensah >> > wrote: >> >> Hello >> >> Please pardon me if the solution to this is somewhere already that I >> have been unable to locate. I have just got a straight >> out-of-the-box >> build of FS. According to the wiki, I should be able to test using >> user >> IDs 1001 and 1002. However, I am get the above error. If I, >> however, >> un-tick register with domain I do net get the error but does not >> communicate either. Is there a conf that I should have done ? >> >> I am using X-lite3 >> >> >> Is NAT involved or are the x-lite clients on the same LAN? Also, you >> might want to turn on a SIP trace at the console to see if there are >> any clues. >> -MC >> > > > >
> From jerry.richards at teotech.com Tue Nov 17 08:22:36 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 17 Nov 2009 08:22:36 -0800 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com><1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl><1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> Message-ID: MC, We would like the dialplan to route the call based on Presence, which is a database lookup. I should be able to do this in Lua, true? Jerry _____ From: Michael Collins [mailto:msc at freeswitch.org] Sent: Monday, November 16, 2009 11:33 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Accessing Config Info From Database On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards wrote: I have a bit of confusion about Lua scripting. When a script is invoked, should it always return an XML string that is used by FS? Or as in the case of dialplan examples, does it actually execute the dialplan (e.g. "session:answer();")? Best Regards, Jerry Jerry, A Lua script that is explicitly called from the dialplan will indeed execute dialplan-ish stuff. For example, let's say you had this in conf/dialplan/default.xml: Then myluascript.lua has something like: --Sample Lua script session:answer() session:sleep(1000) session:streamFile("/path/to/file.wav") session:hangup() Assuming an otherwise default install, the above Lua script would execute when a caller dialed 9876, or if a call was x-ferred to 9876. However, if you're wanting to use Lua to serve up a dialplan then it's totally different. Lua is not called from the dialplan; Lua provides the dialplan to FreeSWITCH. This latter case is the scenario discussed in the wiki section you referenced. (http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration) Are you trying to use Lua scripting for serving up a dynamic configuration of some sort? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/acf65320/attachment.html From mrene_lists at avgs.ca Tue Nov 17 08:43:12 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 08:43:12 -0800 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel In-Reply-To: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> References: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> Message-ID: You can't record until media is present. You could trigger it with execute_on_answer and the record_session application Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: > I'm trying performing a uuid_record command immediately after a > uuid_bridge, but receive a "Can not record session. Media not > enabled on channel" error. proxy_media and bypass_media are both set > to false. > > The uuid_record however works if I use sched_api +1 uuid_record... > but if I do this, I of course loose the first second of conversation. > > Does anyone have any ideas on how I might be able to solve this? > I've turned on DEBUG mode, but nothing out of the ordinary appears. > > http://pastebin.freeswitch.org/11141 > > --matt > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/c460ef84/attachment.html From yehavi.bourvine at gmail.com Tue Nov 17 08:51:00 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 17 Nov 2009 18:51:00 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: References: <4B0266F4.8070602@savion.huji.ac.il> Message-ID: Hello Brian, the situation is as follows: Our PBX machine has more than one interface, each one has a profile. Some phones are registered via one interface and tje others on the other. The call should be sent usinbg the profile of the destination as if not, the IP address of the server in the SIP message is incorrect (the other interface) thus the phone cannot answer. When a call is processed you know the originator profile name; we need also the destination profile name... Thanks! __yehavi: 2009/11/17 Brian West > Why do you need to know the destination profile like that? You get to > pick that on your own so you should already know that before hand. > > > /b > > On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote: > > > Hi > > We have more then one profile. To make a call I have to enter : bridge > > sofia/profile/number at ip > > The problem is when I use : "${use_profile}" I am getting the caller > > profile, and I need the destination profile. > > > > How do I get this information? > > > > Thanks > > > > Eli > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/9461e9bc/attachment-0001.html From mattdfong at gmail.com Tue Nov 17 08:57:54 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 17 Nov 2009 23:57:54 +0700 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel In-Reply-To: References: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> Message-ID: <4256bf830911170857o302b6f41i998cbe974e4d4008@mail.gmail.com> The media should be there, when I uuid_bridge both sessions are parked and should have already had media sent. I'm using ignore_early_media=true --matt On Tue, Nov 17, 2009 at 11:43 PM, Mathieu Rene wrote: > You can't record until media is present. You could trigger it with > execute_on_answer and the record_session application > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: > > I'm trying performing a uuid_record command immediately after a > uuid_bridge, but receive a "Can not record session. Media not enabled on > channel" error. proxy_media and bypass_media are both set to false. > > The uuid_record however works if I use sched_api +1 uuid_record... but if I > do this, I of course loose the first second of conversation. > > Does anyone have any ideas on how I might be able to solve this? I've > turned on DEBUG mode, but nothing out of the ordinary appears. > > http://pastebin.freeswitch.org/11141 > > --matt > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/dac32fc8/attachment.html From krzysiez at go2.pl Tue Nov 17 06:05:07 2009 From: krzysiez at go2.pl (=?UTF-8?Q?Christopher_Z.?=) Date: Tue, 17 Nov 2009 15:05:07 +0100 Subject: [Freeswitch-users] =?utf-8?q?Compilation_problem?= Message-ID: <32c1b333.68a1501e.4b02ad93.bbfcd@go2.pl> Hi, I've got this error after make: http://pastebin.freeswitch.org/11145 Any idea how to fix this error ? Thanks. From krzysiez at go2.pl Tue Nov 17 09:18:32 2009 From: krzysiez at go2.pl (=?UTF-8?Q?Krzysztof_Zimnicki?=) Date: Tue, 17 Nov 2009 18:18:32 +0100 Subject: [Freeswitch-users] =?utf-8?q?=5Bfreeswitch-users=5DCompilation_pr?= =?utf-8?q?oblem?= Message-ID: <3201712c.1d4430a1.4b02dae8.49cdb@go2.pl> Hi, I've got this error after make: http://pastebin.freeswitch.org/11145 Any idea how to fix this error ? Thanks. From freeswitch-users-list at metik.com Tue Nov 17 09:32:05 2009 From: freeswitch-users-list at metik.com (Metik) Date: Tue, 17 Nov 2009 12:32:05 -0500 Subject: [Freeswitch-users] Using uuid_transfer with uuid_hold Message-ID: <4B02DE15.3090207@metik.com> Using the API, any caveats with transferring a call (via uuid_transfer) that has been placed on hold (via uuid_hold) without using "uuild_hold off" before doing so? Is it even possible? -metik From mrene_lists at avgs.ca Tue Nov 17 09:32:52 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 09:32:52 -0800 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel In-Reply-To: <4256bf830911170857o302b6f41i998cbe974e4d4008@mail.gmail.com> References: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> <4256bf830911170857o302b6f41i998cbe974e4d4008@mail.gmail.com> Message-ID: Ah I see what happens, switch_ivr_uuid_bridge will reset the session's read codec (which will be re-initialized as soon as the actual bridge takes place) and uuid_record will think it hasnt been initialized yet. You can set the following vars to execute an application right before bridge starts exchanging audio bridge_pre_execute_aleg_app bridge_pre_execute_aleg_arg Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 8:57 AM, Matthew Fong wrote: > The media should be there, when I uuid_bridge both sessions are > parked and should have already had media sent. I'm using > ignore_early_media=true > > --matt > > On Tue, Nov 17, 2009 at 11:43 PM, Mathieu Rene > wrote: > You can't record until media is present. You could trigger it with > execute_on_answer and the record_session application > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: > >> I'm trying performing a uuid_record command immediately after a >> uuid_bridge, but receive a "Can not record session. Media not >> enabled on channel" error. proxy_media and bypass_media are both >> set to false. >> >> The uuid_record however works if I use sched_api +1 uuid_record... >> but if I do this, I of course loose the first second of conversation. >> >> Does anyone have any ideas on how I might be able to solve this? >> I've turned on DEBUG mode, but nothing out of the ordinary appears. >> >> http://pastebin.freeswitch.org/11141 >> >> --matt >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/f4063684/attachment.html From mrene_lists at avgs.ca Tue Nov 17 09:35:14 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 09:35:14 -0800 Subject: [Freeswitch-users] Using uuid_transfer with uuid_hold In-Reply-To: <4B02DE15.3090207@metik.com> References: <4B02DE15.3090207@metik.com> Message-ID: Shouldnt be a problem, but I think you really want to uuid_park it, not hold it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 9:32 AM, Metik wrote: > Using the API, any caveats with transferring a call (via > uuid_transfer) > that has been placed on hold (via uuid_hold) without using "uuild_hold > off" before doing so? Is it even possible? > > -metik > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Nov 17 09:41:49 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 11:41:49 -0600 Subject: [Freeswitch-users] Using uuid_transfer with uuid_hold In-Reply-To: References: <4B02DE15.3090207@metik.com> Message-ID: <44ABEBCF-FD9C-41C7-9AC1-9FC94FFD509C@freeswitch.org> uuid_hold will send a HOLD indication to the end you're talking to ... it will NOT put the person your talking to on HOLD... I think that is the confusion of uuid_hold. Example: Phone -> FS1 -> FS2(uses uuid_hold) will cause the FS1 box to play hold music to the phone. /b On Nov 17, 2009, at 11:35 AM, Mathieu Rene wrote: > Shouldnt be a problem, but I think you really want to uuid_park it, > not hold it. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/2020cc30/attachment.html From brian at freeswitch.org Tue Nov 17 09:42:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 11:42:35 -0600 Subject: [Freeswitch-users] [freeswitch-users]Compilation problem In-Reply-To: <3201712c.1d4430a1.4b02dae8.49cdb@go2.pl> References: <3201712c.1d4430a1.4b02dae8.49cdb@go2.pl> Message-ID: <68899D9C-6D0C-4576-B6CA-68C04A013C79@freeswitch.org> You 'make current' and stop cross posting.. /b On Nov 17, 2009, at 11:18 AM, Krzysztof Zimnicki wrote: > Hi, > > I've got this error after make: > > http://pastebin.freeswitch.org/11145 > > Any idea how to fix this error ? > > Thanks. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mattdfong at gmail.com Tue Nov 17 09:51:52 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 18 Nov 2009 00:51:52 +0700 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel In-Reply-To: References: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> <4256bf830911170857o302b6f41i998cbe974e4d4008@mail.gmail.com> Message-ID: <4256bf830911170951x643ae03egce257f13982b7af6@mail.gmail.com> Hi Mathieu, This makes sense! Thanks. Since bypass_proxy = false and proxy_media = false, why is it trying to renegotiate a codec on a uuid_bridge? --matt On Wed, Nov 18, 2009 at 12:32 AM, Mathieu Rene wrote: > Ah I see what happens, switch_ivr_uuid_bridge will reset the session's read > codec (which will be re-initialized as soon as the actual bridge takes > place) and uuid_record will think it hasnt been initialized yet. > > You can set the following vars to execute an application right before > bridge starts exchanging audio > bridge_pre_execute_aleg_app > bridge_pre_execute_aleg_arg > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 8:57 AM, Matthew Fong wrote: > > The media should be there, when I uuid_bridge both sessions are parked and > should have already had media sent. I'm using ignore_early_media=true > > --matt > > On Tue, Nov 17, 2009 at 11:43 PM, Mathieu Rene wrote: > >> You can't record until media is present. You could trigger it with >> execute_on_answer and the record_session application >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: >> >> I'm trying performing a uuid_record command immediately after a >> uuid_bridge, but receive a "Can not record session. Media not enabled on >> channel" error. proxy_media and bypass_media are both set to false. >> >> The uuid_record however works if I use sched_api +1 uuid_record... but if >> I do this, I of course loose the first second of conversation. >> >> Does anyone have any ideas on how I might be able to solve this? I've >> turned on DEBUG mode, but nothing out of the ordinary appears. >> >> http://pastebin.freeswitch.org/11141 >> >> --matt >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/c2605d2d/attachment.html From mrene_lists at avgs.ca Tue Nov 17 09:54:11 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 09:54:11 -0800 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel In-Reply-To: <4256bf830911170951x643ae03egce257f13982b7af6@mail.gmail.com> References: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> <4256bf830911170857o302b6f41i998cbe974e4d4008@mail.gmail.com> <4256bf830911170951x643ae03egce257f13982b7af6@mail.gmail.com> Message-ID: Its not re-negotiating it, its re-initializing it so the codec's internal state gets reset. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 9:51 AM, Matthew Fong wrote: > Hi Mathieu, > > This makes sense! Thanks. > > Since bypass_proxy = false and proxy_media = false, why is it trying > to renegotiate a codec on a uuid_bridge? > > --matt > > On Wed, Nov 18, 2009 at 12:32 AM, Mathieu Rene > wrote: > Ah I see what happens, switch_ivr_uuid_bridge will reset the > session's read codec (which will be re-initialized as soon as the > actual bridge takes place) and uuid_record will think it hasnt been > initialized yet. > > You can set the following vars to execute an application right > before bridge starts exchanging audio > bridge_pre_execute_aleg_app > bridge_pre_execute_aleg_arg > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 8:57 AM, Matthew Fong wrote: > >> The media should be there, when I uuid_bridge both sessions are >> parked and should have already had media sent. I'm using >> ignore_early_media=true >> >> --matt >> >> On Tue, Nov 17, 2009 at 11:43 PM, Mathieu Rene >> wrote: >> You can't record until media is present. You could trigger it with >> execute_on_answer and the record_session application >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: >> >>> I'm trying performing a uuid_record command immediately after a >>> uuid_bridge, but receive a "Can not record session. Media not >>> enabled on channel" error. proxy_media and bypass_media are both >>> set to false. >>> >>> The uuid_record however works if I use sched_api +1 uuid_record... >>> but if I do this, I of course loose the first second of >>> conversation. >>> >>> Does anyone have any ideas on how I might be able to solve this? >>> I've turned on DEBUG mode, but nothing out of the ordinary appears. >>> >>> http://pastebin.freeswitch.org/11141 >>> >>> --matt >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/36d677c6/attachment.html From freeswitch-users-list at metik.com Tue Nov 17 10:03:03 2009 From: freeswitch-users-list at metik.com (Metik) Date: Tue, 17 Nov 2009 13:03:03 -0500 Subject: [Freeswitch-users] Using uuid_transfer with uuid_hold In-Reply-To: <44ABEBCF-FD9C-41C7-9AC1-9FC94FFD509C@freeswitch.org> References: <4B02DE15.3090207@metik.com> <44ABEBCF-FD9C-41C7-9AC1-9FC94FFD509C@freeswitch.org> Message-ID: <4B02E557.8050608@metik.com> Brian, That explains what I have been seeing... The console would freeze or the xml-rpc request would never receive a response (trunk rev 15463). -metik Brian West wrote: > uuid_hold will send a HOLD indication to the end you're talking to ... > it will NOT put the person your talking to on HOLD... I think that is > the confusion of uuid_hold. > > > Example: > > Phone -> FS1 -> FS2(uses uuid_hold) will cause the FS1 box to play > hold music to the phone. > > /b > > > On Nov 17, 2009, at 11:35 AM, Mathieu Rene wrote: > >> Shouldnt be a problem, but I think you really want to uuid_park it, >> not hold it. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From tina at a2unlimited.com Tue Nov 17 10:04:23 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Tue, 17 Nov 2009 13:04:23 -0500 Subject: [Freeswitch-users] ESL: No matching function... In-Reply-To: References: Message-ID: MC, Yes, I tried "make && make perlmod", which did not fix the error. Just finished deploying an instance of the application on another server that did not produce the error (exact same configuration). Not sure what is causing it, or how to fix it. Bizarre. - Tina > On Mon, Nov 16, 2009 at 3:22 PM, wrote: > >> I have three FreeSWITCH servers currently setup with perl modules using >> ESL to send call instructions and monitor events. On two of the >> servers, >> my modules execute without error, but on a third, I keep getting the >> following error: >> >> No matching function for overloaded 'new_ESLconnection' at >> /usr/lib64/perl5/site_perl/5.8.8/x86_64-linux-thread-multi/ESL.pm line >> 116. >> >> Is this something in ESL that I'm doing wrong, or is it an issue related >> to the perl configuration on the server? >> >> As far as I can tell, the three servers have been setup the same. >> >> - Tina >> >> > Can you cd into libs/esl and do "make && make perlmod" and see if any > errors > show up? > -MC > From info at daccii.it Tue Nov 17 10:48:30 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Tue, 17 Nov 2009 19:48:30 +0100 Subject: [Freeswitch-users] Help testing a new startskype.sh script Message-ID: <4B02EFFE.1030705@daccii.it> Hi, this morning i've started to work on a new startskype.sh script, for mod_skypiax and, finally, it works as it should! I've done some preliminary testing, can someone help me to test it better with many users? Here jira report http://jira.freeswitch.org/browse/MODSKYPIAX-59 Thank you Best Regards, Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/59bd11dc/attachment.vcf From msc at freeswitch.org Tue Nov 17 10:56:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Nov 2009 10:56:58 -0800 Subject: [Freeswitch-users] ESL: No matching function... In-Reply-To: References: Message-ID: <87f2f3b90911171056j5468436bqa765831665109bec@mail.gmail.com> On Tue, Nov 17, 2009 at 10:04 AM, wrote: > MC, > > Yes, I tried "make && make perlmod", which did not fix the error. > > Just finished deploying an instance of the application on another server > that did not produce the error (exact same configuration). > > Not sure what is causing it, or how to fix it. > > Bizarre. > > - Tina > > Is the offending machine a 32 or 64 bit machine? Just curious if there is something physically different about this machine than the others. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/48d02983/attachment.html From msc at freeswitch.org Tue Nov 17 11:02:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Nov 2009 11:02:35 -0800 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> <1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> Message-ID: <87f2f3b90911171102s553de3b8lb1b8c36155d9fd51@mail.gmail.com> On Tue, Nov 17, 2009 at 8:22 AM, Jerry Richards wrote: > MC, > > We would like the dialplan to route the call based on Presence, which is a > database lookup. I should be able to do this in Lua, true? > > Jerry > > Yes, you can use Lua for this if you wish to do so, HOWEVER, luasql has a bug so tread carefully. Check with Chad/hunmonk for thoughts on doing db lookups from Lua. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/ca588669/attachment.html From msc at freeswitch.org Tue Nov 17 11:05:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Nov 2009 11:05:16 -0800 Subject: [Freeswitch-users] Registration Error - 408 timeout and now 403 In-Reply-To: <4B02BD53.5040203@greatiam.com> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> <4B01A8AE.7070708@greatiam.com> <4B02BD53.5040203@greatiam.com> Message-ID: <87f2f3b90911171105s7fb2fea3l316fc2777cbc051a@mail.gmail.com> Try doing this: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install -MC On Tue, Nov 17, 2009 at 7:12 AM, Sam Abekah-Mensah wrote: > Hello > > I have tried the same setup but this time using a windows build FS1.0.4 > on an XP machine and all is fine. The sample 1001 and 1002 IDs work > without any tweaking at all. Could the problem be with the linux build > 1.0.4.? I am running on an FC11 machine. > > On the FC11 box I used the svn link to build using the ff: > > bootstarp.sh > configure withoout libcurl to eliinate the spidermonkey lib error > make > make install > > Did I miss anything ? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/6dbbd142/attachment.html From kristian.kielhofner at gmail.com Tue Nov 17 11:21:01 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Nov 2009 14:21:01 -0500 Subject: [Freeswitch-users] Build FS without spandsp or libtiff Message-ID: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> Hello everyone, I'm trying to keep my build as small as possible (for AstLinux). Is there anyway to build without spandsp or libtiff? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Tue Nov 17 11:24:23 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 13:24:23 -0600 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> Message-ID: Don't build mod_fax /b On Nov 17, 2009, at 1:21 PM, Kristian Kielhofner wrote: > Hello everyone, > > I'm trying to keep my build as small as possible (for AstLinux). Is > there anyway to build without spandsp or libtiff? > > Thanks! > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/bee1c3cb/attachment.html From kristian.kielhofner at gmail.com Tue Nov 17 11:33:33 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Nov 2009 14:33:33 -0500 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> Message-ID: <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> I'm not... On Tue, Nov 17, 2009 at 2:24 PM, Brian West wrote: > Don't build mod_fax > /b -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Tue Nov 17 11:37:45 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 13:37:45 -0600 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> Message-ID: <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> OH you need spandsp for VoipCodecs. No way around that one. /b On Nov 17, 2009, at 1:33 PM, Kristian Kielhofner wrote: > I'm not... > > On Tue, Nov 17, 2009 at 2:24 PM, Brian West > wrote: >> Don't build mod_fax >> /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/66db7e27/attachment.html From regs at kinetix.gr Tue Nov 17 11:44:48 2009 From: regs at kinetix.gr (regs at kinetix.gr) Date: Tue, 17 Nov 2009 21:44:48 +0200 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp In-Reply-To: <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> References: <4B02A829.7080708@kinetix.gr> <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> Message-ID: <4B02FD30.8050502@kinetix.gr> I am trying to achieve something similar to that : http://wiki.freeswitch.org/wiki/Codec_negotiation#Modifying_the_codec_when_using_proxy_media_mode but I am using xml_curl to create the dialplan (i.e. the web server that serves the dialplan makes the decision about the SDP). So I need a way to write the new SDP in the XML dialplan response. However, in the above example due to the regex manipulation the user is not facing the problem that I am with setting the switch_r_sdp to a complex value that contains =, spaces, new lines etc. Brian West wrote: > Why are you needing to rewrite it? > > /b > > On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: > > >> I am trying to use switch_r_sdp to rewrite the SDP. >> The problem I am facing has to do with the way of doing it. >> >> Let's say I have: >> >> v=0 >> o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx >> s=Opal SIP Session >> c=IN IP4 xxx.xxx.xxx.xxx >> t=0 0 >> m=audio 5144 RTP/AVP 18 3 101 120 >> c=IN IP4 xxx.xxx.xxx.xxx >> a=rtpmap:18 G729/8000/1 >> a=fmtp:18 annexb=no >> a=rtpmap:3 gsm/8000/1 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16,32,36 >> a=rtpmap:120 NSE/8000 >> a=fmtp:120 192-193 >> >> who to I set the switch_r_sdp variable in xml? >> >> Obviously this doesn't work : >> >> >> >> Do I have to escape any special characters and how? >> I tried using escaped quotes, escaped spaces, escaped tabs etc. >> Nothing worked. >> >> Any suggestions? >> >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Tue Nov 17 12:15:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Nov 2009 14:15:02 -0600 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp In-Reply-To: <4B02FD30.8050502@kinetix.gr> References: <4B02A829.7080708@kinetix.gr> <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> <4B02FD30.8050502@kinetix.gr> Message-ID: <191c3a030911171215s336b7d7bk7b1959744da3d2d3@mail.gmail.com> you can do On Tue, Nov 17, 2009 at 1:44 PM, regs at kinetix.gr wrote: > I am trying to achieve something similar to that : > > http://wiki.freeswitch.org/wiki/Codec_negotiation#Modifying_the_codec_when_using_proxy_media_mode > > but I am using xml_curl to create the dialplan (i.e. the web server that > serves the dialplan makes the decision about the SDP). So I need a way > to write > the new SDP in the XML dialplan response. However, in the above example > due to the regex manipulation the user is not facing the problem that I am > with setting the switch_r_sdp to a complex value that contains =, > spaces, new lines etc. > > Brian West wrote: > > Why are you needing to rewrite it? > > > > /b > > > > On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: > > > > > >> I am trying to use switch_r_sdp to rewrite the SDP. > >> The problem I am facing has to do with the way of doing it. > >> > >> Let's say I have: > >> > >> v=0 > >> o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx > >> s=Opal SIP Session > >> c=IN IP4 xxx.xxx.xxx.xxx > >> t=0 0 > >> m=audio 5144 RTP/AVP 18 3 101 120 > >> c=IN IP4 xxx.xxx.xxx.xxx > >> a=rtpmap:18 G729/8000/1 > >> a=fmtp:18 annexb=no > >> a=rtpmap:3 gsm/8000/1 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16,32,36 > >> a=rtpmap:120 NSE/8000 > >> a=fmtp:120 192-193 > >> > >> who to I set the switch_r_sdp variable in xml? > >> > >> Obviously this doesn't work : > >> > >> > >> > >> Do I have to escape any special characters and how? > >> I tried using escaped quotes, escaped spaces, escaped tabs etc. > >> Nothing worked. > >> > >> Any suggestions? > >> > >> > >> > >> > >> -- > >> ------------------------------------------- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: regs at kinetix.gr > >> ------------------------------------------- > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/726426fb/attachment-0001.html From anthony.minessale at gmail.com Tue Nov 17 12:17:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Nov 2009 14:17:04 -0600 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp In-Reply-To: <191c3a030911171215s336b7d7bk7b1959744da3d2d3@mail.gmail.com> References: <4B02A829.7080708@kinetix.gr> <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> <4B02FD30.8050502@kinetix.gr> <191c3a030911171215s336b7d7bk7b1959744da3d2d3@mail.gmail.com> Message-ID: <191c3a030911171217q14acd9c3la4427fcfa7ccc250@mail.gmail.com> I should have said On Tue, Nov 17, 2009 at 2:15 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you can do > > > ]]> > > > > On Tue, Nov 17, 2009 at 1:44 PM, regs at kinetix.gr wrote: > >> I am trying to achieve something similar to that : >> >> http://wiki.freeswitch.org/wiki/Codec_negotiation#Modifying_the_codec_when_using_proxy_media_mode >> >> but I am using xml_curl to create the dialplan (i.e. the web server that >> serves the dialplan makes the decision about the SDP). So I need a way >> to write >> the new SDP in the XML dialplan response. However, in the above example >> due to the regex manipulation the user is not facing the problem that I >> am >> with setting the switch_r_sdp to a complex value that contains =, >> spaces, new lines etc. >> >> Brian West wrote: >> > Why are you needing to rewrite it? >> > >> > /b >> > >> > On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: >> > >> > >> >> I am trying to use switch_r_sdp to rewrite the SDP. >> >> The problem I am facing has to do with the way of doing it. >> >> >> >> Let's say I have: >> >> >> >> v=0 >> >> o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx >> >> s=Opal SIP Session >> >> c=IN IP4 xxx.xxx.xxx.xxx >> >> t=0 0 >> >> m=audio 5144 RTP/AVP 18 3 101 120 >> >> c=IN IP4 xxx.xxx.xxx.xxx >> >> a=rtpmap:18 G729/8000/1 >> >> a=fmtp:18 annexb=no >> >> a=rtpmap:3 gsm/8000/1 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16,32,36 >> >> a=rtpmap:120 NSE/8000 >> >> a=fmtp:120 192-193 >> >> >> >> who to I set the switch_r_sdp variable in xml? >> >> >> >> Obviously this doesn't work : >> >> >> >> >> >> >> >> Do I have to escape any special characters and how? >> >> I tried using escaped quotes, escaped spaces, escaped tabs etc. >> >> Nothing worked. >> >> >> >> Any suggestions? >> >> >> >> >> >> >> >> >> >> -- >> >> ------------------------------------------- >> >> Apostolos Pantsiopoulos >> >> Kinetix Tele.com R & D >> >> email: regs at kinetix.gr >> >> ------------------------------------------- >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >> users >> >> http://www.freeswitch.org >> >> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/cf8ff333/attachment.html From kristian.kielhofner at gmail.com Tue Nov 17 12:19:15 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Nov 2009 15:19:15 -0500 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> Message-ID: <2d9149cd0911171219y744cb81cwf53c5d25ea26c05e@mail.gmail.com> Ah yes, using spandsp instead of libvoipcodecs. I'm not going to question the wisdom of that move but it appears that spandsp (as-is) doesn't cross compile properly (make_at_dictionary is built using the cross compiler and can't run on the host). Once that error is fixed (I hacked it for now) it still bombs as shown here: http://pastebin.freeswitch.org/11149 spandsp + libtiff are almost certainly *much* larger than libvoipcodecs but if that means that I can also build mod_fax I guess it's worth it ;). On Tue, Nov 17, 2009 at 2:37 PM, Brian West wrote: > OH you need spandsp for VoipCodecs. ?No way around that one. > /b -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From tina at a2unlimited.com Tue Nov 17 12:21:22 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Tue, 17 Nov 2009 15:21:22 -0500 Subject: [Freeswitch-users] ESL: No matching function... In-Reply-To: References: Message-ID: <3a74a318fe704f2015db18186913d71d.squirrel@emailmg.ipower.com> Confirmed 64-bit machine. > On Tue, Nov 17, 2009 at 10:04 AM, wrote: > >> MC, >> >> Yes, I tried "make && make perlmod", which did not fix the error. >> >> Just finished deploying an instance of the application on another server >> that did not produce the error (exact same configuration). >> >> Not sure what is causing it, or how to fix it. >> >> Bizarre. >> >> - Tina >> >> > Is the offending machine a 32 or 64 bit machine? Just curious if there is > something physically different about this machine than the others. > -MC > From stevendt at primrosebank.net Tue Nov 17 13:02:19 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 17 Nov 2009 21:02:19 -0000 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 Message-ID: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> Hi, I have just about got FreeSwitch working with a Cisco 7940 Phone. After much reading, I worked out that I needed a TFTP server on the network that would supply the phone with it's SIP personality and config etc. I have been able to get the phone working and realise that the TFTP server needs to be available every time the phone loses power etc. At the moment, I have the TFTP server running on a temporary machine but it would be neater if it ran on the same machine as FreeSwitch. This will be a very small FreeSwitch installation, so, ....... Is there any reason why I should not try to run FreeSwitch and the SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't think the server should put much load on the machine but wondered if there were any other reasons why this is a bad idea ? In addition, while I have the phone working - I get a status message on boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf Can anyone tell me how to locate the errors in this file please ? (I have posted it to the Pastebin) Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/56bad71a/attachment.html From lists at tigertech.com Tue Nov 17 14:18:42 2009 From: lists at tigertech.com (Robert L Mathews) Date: Tue, 17 Nov 2009 14:18:42 -0800 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time Message-ID: <4B032142.1000308@tigertech.com> I'm using FreeSWITCH 1.0.4. When I make a call from a SIP phone to either a conference or an echo test on the FreeSWITCH server, the latency ("lag") starts off very low -- a fraction of a second. But as several minutes of time goes by, the lag increases. After, say, 15 minutes, the lag will reach a couple of seconds, making conference calls unusable. This does not happen on pure SIP-to-SIP calls, even when FreeSWITCH is handling the RTP media. If I hang up and immediately call back in (even to the same conference), the lag is reset to almost zero. If I put the call on "hold" and take it off hold, the lag is also gone. During testing, I've found that this may be related to the freeswitch app on the server not getting all the CPU time it wants. If I suspend the freeswitch process for two seconds and then resume it, the sound stops for two seconds, as I'd expect. But the echo/conference calls that were active are then lagged by two seconds until they hang up (or get put on hold), even after freeswitch is resumed and getting all the CPU time it needs. This is easily reproduced by making a SIP call to the echo test module, then: pkill -STOP freeswitch; sleep 2; pkill -CONT freeswitch Any echo test or conference call that was in progress will then be permanently lagged by two seconds. However, any SIP-to-SIP calls that were in progress will not become lagged. Of course, killing it with -STOP is an artificially nasty thing to do. But it effectively just prevents it from being scheduled on the CPU for a short period of time, and I can duplicate the same behavior (more gradually) by just increasing the load on the machine to the point that the freeswitch app isn't getting much CPU time. Just for the record, I get the same results from several different phones and several different Internet connections, all of which have a ping latency of under 40 ms to the server. This problem does not happen using the same phones and network connections to an asterisk server. Throwing out an even more complicated example that I've encountered: If I have a SIP-to-SIP call going from party A to party B and I stop the process for two seconds, it doesn't permanently introduce lag to that call, as I mentioned. But if a third person (party C) starts eavesdropping on the call and presses "3" to make it a three way call, and then I suspend it for two seconds, the call between A and B isn't lagged, but what party C hears and sends *is* lagged. Any ideas on how to fix this? Do other people see the same thing happening? As I said, the gradual increase in lag over a long period of time makes long conferences unusable, unfortunately. -- Rob From jlenk at frontiernet.net Tue Nov 17 18:38:17 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 17 Nov 2009 18:38:17 -0800 (PST) Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> Message-ID: <1258511897776-4023012.post@n2.nabble.com> Hi I run the SolarWinds TFTP server alongside FS on my small installation - works nicely! Jeff Dave Stevenson wrote: > > Hi, > > I have just about got FreeSwitch working with a Cisco 7940 Phone. After > much reading, I worked out that I needed a TFTP server on the network that > would supply the phone with it's SIP personality and config etc. I have > been able to get the phone working and realise that the TFTP server needs > to be available every time the phone loses power etc. At the moment, I > have the TFTP server running on a temporary machine but it would be neater > if it ran on the same machine as FreeSwitch. This will be a very small > FreeSwitch installation, so, ....... > > Is there any reason why I should not try to run FreeSwitch and the > SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't think > the server should put much load on the machine but wondered if there were > any other reasons why this is a bad idea ? > > In addition, while I have the phone working - I get a status message on > boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf > > Can anyone tell me how to locate the errors in this file please ? (I have > posted it to the Pastebin) > > Regards > Dave > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/TFTP-Server-Cisco-7540-tp4021305p4023012.html Sent from the freeswitch-users mailing list archive at Nabble.com. From elihayun at gmail.com Tue Nov 17 21:36:49 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 18 Nov 2009 07:36:49 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: References: <4B0266F4.8070602@savion.huji.ac.il> Message-ID: <4B0387F1.7070105@savion.huji.ac.il> Brian West wrote: > Why do you need to know the destination profile like that? You get to > pick that on your own so you should already know that before hand. > > > /b > > On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote: > > >> Hi >> We have more then one profile. To make a call I have to enter : bridge >> sofia/profile/number at ip >> The problem is when I use : "${use_profile}" I am getting the caller >> profile, and I need the destination profile. >> >> How do I get this information? >> >> Thanks >> >> Eli >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Thanks for your answer. The problem is when I call to that number that the phone hook to other server, I cannot make the call. Is there is a variable that can tell me the destination profile? Lets say the other profile called "ph1" I have to dial sofia/ph1/xxxxx at host to make the call. Is there other way to do that? Thanks Eli -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/7224c4ea/attachment.html From ujjval at simplesignal.com Tue Nov 17 21:49:02 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Tue, 17 Nov 2009 21:49:02 -0800 Subject: [Freeswitch-users] Changing User-Agent String Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#User_Agent_.5Buser-agent-string.5D As per the above link, we can change the User Agent String, but I added this param name but does not seem to work. [user at freeswitch autoload_configs]$ vi sofia.conf.xml -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/5697880f/attachment.html From mrene_lists at avgs.ca Tue Nov 17 21:52:01 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 21:52:01 -0800 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> Message-ID: It needs to go in the profile, not in sofia's global config. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 9:49 PM, Ujjval Karihaloo wrote: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#User_Agent_.5Buser-agent-string.5D > > As per the above link, we can change the User Agent String, but I > added this param name but does not seem to work. > > [user at freeswitch autoload_configs]$ vi sofia.conf.xml > > > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/c3ad8dcc/attachment.html From mcampbellsmith at gmail.com Tue Nov 17 21:53:27 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 18 Nov 2009 16:53:27 +1100 Subject: [Freeswitch-users] application="info" Message-ID: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> HI All, pretty basic question and I feel a bit stupid asking this, but what are the prerequisites for the INFO to be displayed when is called in a dialplan? ie are there requirements on the loglevel, does the INFO command have to be put at a certain place in the dialplan etc? The reason i ask is that I have a dialplan and the is not getting triggered on the fs_cli output. Is there some other debbugging level that needs to be set? Thanks! From mrene_lists at avgs.ca Tue Nov 17 21:56:09 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 21:56:09 -0800 Subject: [Freeswitch-users] application="info" In-Reply-To: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> References: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> Message-ID: <89E59F4D-6335-4D51-A17B-9A040EE2ACAC@avgs.ca> If you press F8 (or do /log 7), you will see what the dialplan is executing, try to see if you see the info app in there And, your global loglevel has to be <= INFO too... fsctl loglevel debug Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote: > HI All, > > pretty basic question and I feel a bit stupid asking this, but what > are the prerequisites for the INFO to be displayed when application="info"/> is called in a dialplan? > > ie are there requirements on the loglevel, does the INFO command have > to be put at a certain place in the dialplan etc? > > The reason i ask is that I have a dialplan and the application="info"/> is not getting triggered on the fs_cli output. > Is there some other debbugging level that needs to be set? > > Thanks! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Tue Nov 17 22:05:41 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 18 Nov 2009 17:05:41 +1100 Subject: [Freeswitch-users] application="info" In-Reply-To: <89E59F4D-6335-4D51-A17B-9A040EE2ACAC@avgs.ca> References: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> <89E59F4D-6335-4D51-A17B-9A040EE2ACAC@avgs.ca> Message-ID: <33c87fa30911172205m75e5ab91m3e54a4647d098e2@mail.gmail.com> I had console loglevel set to DEBUG, so that should be fine. And I do see that FS is executing the exact extension where I have put the INFO application - still no info on the console.... I'm using FreeSWITCH Version 1.0.trunk (15490) On Wed, Nov 18, 2009 at 4:56 PM, Mathieu Rene wrote: > If you press F8 (or do /log 7), you will see what the dialplan is > executing, try to see if you see the info app in there > > And, your global loglevel has to be <= INFO too... fsctl loglevel debug > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote: > >> HI All, >> >> pretty basic question and I feel a bit stupid asking this, but what >> are the prerequisites for the INFO to be displayed when > application="info"/> is called in a dialplan? >> >> ie are there requirements on the loglevel, does the INFO command have >> to be put at a certain place in the dialplan etc? >> >> The reason i ask is that I have a dialplan and the > application="info"/> is not getting triggered on the fs_cli output. >> Is there some other debbugging level that needs to be set? >> >> Thanks! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mrene_lists at avgs.ca Tue Nov 17 22:08:59 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 22:08:59 -0800 Subject: [Freeswitch-users] application="info" In-Reply-To: <33c87fa30911172205m75e5ab91m3e54a4647d098e2@mail.gmail.com> References: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> <89E59F4D-6335-4D51-A17B-9A040EE2ACAC@avgs.ca> <33c87fa30911172205m75e5ab91m3e54a4647d098e2@mail.gmail.com> Message-ID: Console loglevel only sets the loglevel on the console, not on fs_cli or other event_socket client programs. You have to do /log 7 on fs_cli. fsctl loglevel is the global system loglevel, if its at warning, you wont see anything below warning ANYWHERE (console/event socket log files) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 10:05 PM, Mark Campbell-Smith wrote: > I had console loglevel set to DEBUG, so that should be fine. > > And I do see that FS is executing the exact extension where I have put > the INFO application - still no info on the console.... > > I'm using FreeSWITCH Version 1.0.trunk (15490) > > > > On Wed, Nov 18, 2009 at 4:56 PM, Mathieu Rene > wrote: >> If you press F8 (or do /log 7), you will see what the dialplan is >> executing, try to see if you see the info app in there >> >> And, your global loglevel has to be <= INFO too... fsctl loglevel >> debug >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote: >> >>> HI All, >>> >>> pretty basic question and I feel a bit stupid asking this, but what >>> are the prerequisites for the INFO to be displayed when >> application="info"/> is called in a dialplan? >>> >>> ie are there requirements on the loglevel, does the INFO command >>> have >>> to be put at a certain place in the dialplan etc? >>> >>> The reason i ask is that I have a dialplan and the >> application="info"/> is not getting triggered on the fs_cli output. >>> Is there some other debbugging level that needs to be set? >>> >>> Thanks! >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Tue Nov 17 22:17:31 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 18 Nov 2009 17:17:31 +1100 Subject: [Freeswitch-users] application="info" In-Reply-To: References: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> <89E59F4D-6335-4D51-A17B-9A040EE2ACAC@avgs.ca> <33c87fa30911172205m75e5ab91m3e54a4647d098e2@mail.gmail.com> Message-ID: <33c87fa30911172217p5f22ae24j1721c92e060538d8@mail.gmail.com> ahha... great. Thanks Mathieu. On Wed, Nov 18, 2009 at 5:08 PM, Mathieu Rene wrote: > Console loglevel only sets the loglevel on the console, not on fs_cli > or other event_socket client programs. You have to do /log 7 on fs_cli. > > fsctl loglevel is the global system loglevel, if its at warning, you > wont see anything below warning ANYWHERE (console/event socket log > files) > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 10:05 PM, Mark Campbell-Smith wrote: > >> I had console loglevel set to DEBUG, so that should be fine. >> >> And I do see that FS is executing the exact extension where I have put >> the INFO application - still no info on the console.... >> >> I'm using FreeSWITCH Version 1.0.trunk (15490) >> >> >> >> On Wed, Nov 18, 2009 at 4:56 PM, Mathieu Rene >> wrote: >>> If you press F8 (or do /log 7), you will see what the dialplan is >>> executing, try to see if you see the info app in there >>> >>> And, your global loglevel has to be <= INFO too... fsctl loglevel >>> debug >>> >>> Mathieu Rene >>> Avant-Garde Solutions Inc >>> Office: + 1 (514) 664-1044 x100 >>> Cell: +1 (514) 664-1044 x200 >>> mrene at avgs.ca >>> >>> >>> >>> >>> On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote: >>> >>>> HI All, >>>> >>>> pretty basic question and I feel a bit stupid asking this, but what >>>> are the prerequisites for the INFO to be displayed when >>> application="info"/> is called in a dialplan? >>>> >>>> ie are there requirements on the loglevel, does the INFO command >>>> have >>>> to be put at a certain place in the dialplan etc? >>>> >>>> The reason i ask is that I have a dialplan and the >>> application="info"/> is not getting triggered on the fs_cli output. >>>> Is there some other debbugging level that needs to be set? >>>> >>>> Thanks! >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jonas.gauffin at gmail.com Wed Nov 18 00:05:34 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 18 Nov 2009 09:05:34 +0100 Subject: [Freeswitch-users] acl configuration Message-ID: Hello, What should my acl.conf.xml look like if I want to allow ALL calls on the external profile and use only digest authentication on all other profiles? Thanks, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/8df807b4/attachment.html From regs at kinetix.gr Wed Nov 18 00:13:42 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Wed, 18 Nov 2009 10:13:42 +0200 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp In-Reply-To: <191c3a030911171217q14acd9c3la4427fcfa7ccc250@mail.gmail.com> References: <4B02A829.7080708@kinetix.gr> <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> <4B02FD30.8050502@kinetix.gr> <191c3a030911171215s336b7d7bk7b1959744da3d2d3@mail.gmail.com> <191c3a030911171217q14acd9c3la4427fcfa7ccc250@mail.gmail.com> Message-ID: <4B03ACB6.8060302@kinetix.gr> Thanx! That worked fine. Anthony Minessale wrote: > I should have said > > > ]]> > > > > On Tue, Nov 17, 2009 at 2:15 PM, Anthony Minessale > > wrote: > > you can do > > > ]]> > > > > On Tue, Nov 17, 2009 at 1:44 PM, regs at kinetix.gr > > > wrote: > > I am trying to achieve something similar to that : > http://wiki.freeswitch.org/wiki/Codec_negotiation#Modifying_the_codec_when_using_proxy_media_mode > > but I am using xml_curl to create the dialplan (i.e. the web > server that > serves the dialplan makes the decision about the SDP). So I need > a way > to write > the new SDP in the XML dialplan response. However, in the above > example > due to the regex manipulation the user is not facing the > problem that I am > with setting the switch_r_sdp to a complex value that contains =, > spaces, new lines etc. > > Brian West wrote: > > Why are you needing to rewrite it? > > > > /b > > > > On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: > > > > > >> I am trying to use switch_r_sdp to rewrite the SDP. > >> The problem I am facing has to do with the way of doing it. > >> > >> Let's say I have: > >> > >> v=0 > >> o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx > >> s=Opal SIP Session > >> c=IN IP4 xxx.xxx.xxx.xxx > >> t=0 0 > >> m=audio 5144 RTP/AVP 18 3 101 120 > >> c=IN IP4 xxx.xxx.xxx.xxx > >> a=rtpmap:18 G729/8000/1 > >> a=fmtp:18 annexb=no > >> a=rtpmap:3 gsm/8000/1 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16,32,36 > >> a=rtpmap:120 NSE/8000 > >> a=fmtp:120 192-193 > >> > >> who to I set the switch_r_sdp variable in xml? > >> > >> Obviously this doesn't work : > >> > >> > >> > >> Do I have to escape any special characters and how? > >> I tried using escaped quotes, escaped spaces, escaped tabs etc. > >> Nothing worked. > >> > >> Any suggestions? > >> > >> > >> > >> > >> -- > >> ------------------------------------------- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: regs at kinetix.gr > >> ------------------------------------------- > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From stevendt at primrosebank.net Wed Nov 18 01:03:08 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 18 Nov 2009 09:03:08 -0000 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> <1258511897776-4023012.post@n2.nabble.com> Message-ID: <1015DE92916E4605A896A02360B21E27@bp1.ad.bp.com> Hi Jeff, thanks lot for the feedback, sounds like it'll do just what I want without breaking FreeSwitch, or needing to have a dedicated server just for a few personality and supporting files. I have been messing with some custom ring tones. As I'm sure you know, but others might not, the Cisco phone downloads the sound file from the TFTP server everytime you change the ringtone from one of the default ones so the TFTP server needs to be on all the time so running it on the FreeSwitch machine is ideal for me, regards Dave ----- Original Message ----- From: "Jeff Lenk" To: Sent: Wednesday, November 18, 2009 2:38 AM Subject: Re: [Freeswitch-users] TFTP Server & Cisco 7540 > > Hi > > I run the SolarWinds TFTP server alongside FS on my small installation - > works nicely! > > Jeff > > > > Dave Stevenson wrote: >> >> Hi, >> >> I have just about got FreeSwitch working with a Cisco 7940 Phone. After >> much reading, I worked out that I needed a TFTP server on the network >> that >> would supply the phone with it's SIP personality and config etc. I have >> been able to get the phone working and realise that the TFTP server needs >> to be available every time the phone loses power etc. At the moment, I >> have the TFTP server running on a temporary machine but it would be >> neater >> if it ran on the same machine as FreeSwitch. This will be a very small >> FreeSwitch installation, so, ....... >> >> Is there any reason why I should not try to run FreeSwitch and the >> SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't >> think >> the server should put much load on the machine but wondered if there were >> any other reasons why this is a bad idea ? >> >> In addition, while I have the phone working - I get a status message on >> boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf >> >> Can anyone tell me how to locate the errors in this file please ? (I have >> posted it to the Pastebin) >> >> Regards >> Dave >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/TFTP-Server-Cisco-7540-tp4021305p4023012.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mustafa.pk at gmail.com Wed Nov 18 01:20:45 2009 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Wed, 18 Nov 2009 14:20:45 +0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <1015DE92916E4605A896A02360B21E27@bp1.ad.bp.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> <1258511897776-4023012.post@n2.nabble.com> <1015DE92916E4605A896A02360B21E27@bp1.ad.bp.com> Message-ID: <8213d6070911180120r2f8c2908x25825c4df96c6229@mail.gmail.com> i don't really think you need to erect a dedicated server for serving configs and binaries over tftp protocol, when installation size is not _very_ large. On Wed, Nov 18, 2009 at 2:03 PM, Dave Stevenson wrote: > Hi Jeff, > > thanks ?lot for the feedback, sounds like it'll do just what I want without > breaking FreeSwitch, or needing to have a dedicated server just for a few > personality and supporting files. > > I have been messing with some custom ring tones. As I'm sure you know, but > others might not, the Cisco phone downloads the sound file from the TFTP > server everytime you change the ringtone from one of the default ones so the > TFTP server needs to be on all the time so running it on the FreeSwitch > machine is ideal for me, > > regards > Dave > > > > ----- Original Message ----- > From: "Jeff Lenk" > To: > Sent: Wednesday, November 18, 2009 2:38 AM > Subject: Re: [Freeswitch-users] TFTP Server & Cisco 7540 > > >> >> Hi >> >> I run the SolarWinds TFTP server alongside FS on my small installation - >> works nicely! >> >> Jeff >> >> >> >> Dave Stevenson wrote: >>> >>> Hi, >>> >>> I have just about got FreeSwitch working with a Cisco 7940 Phone. After >>> much reading, I worked out that I needed a TFTP server on the network >>> that >>> would supply the phone with it's SIP personality and config etc. I have >>> been able to get the phone working and realise that the TFTP server needs >>> to be available every time the phone loses power etc. At the moment, I >>> have the TFTP server running on a temporary machine but it would be >>> neater >>> if it ran on the same machine as FreeSwitch. This will be a very small >>> FreeSwitch installation, so, ....... >>> >>> Is there any reason why I should not try to run FreeSwitch and the >>> SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't >>> think >>> the server should put much load on the machine but wondered if there were >>> any other reasons why this is a bad idea ? >>> >>> In addition, while I have the phone working - I get a status message on >>> boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf >>> >>> Can anyone tell me how to locate the errors in this file please ? (I have >>> posted it to the Pastebin) >>> >>> Regards >>> Dave >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/TFTP-Server-Cisco-7540-tp4021305p4023012.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com From abeka at greatiam.com Wed Nov 18 02:43:15 2009 From: abeka at greatiam.com (Sam Abekah-Mensah) Date: Wed, 18 Nov 2009 10:43:15 +0000 Subject: [Freeswitch-users] Registration Error - 408 timeout and now 403 In-Reply-To: <87f2f3b90911171105s7fb2fea3l316fc2777cbc051a@mail.gmail.com> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> <4B01A8AE.7070708@greatiam.com> <4B02BD53.5040203@greatiam.com> <87f2f3b90911171105s7fb2fea3l316fc2777cbc051a@mail.gmail.com> Message-ID: <4B03CFC3.7040501@greatiam.com> Thank you so much for your responses. I have resolved the problem somehow. I copied the default.xml from the root conf folder, the sample 1001 and 1002 .xml on a windows build to the Fedora 11 machine and that worked. I guessed the rejection was with the configuration on theFedora box.even though it was more straight -out-of-the-box. No one is seeking help on this so it must be something I did. I am reinstalling FC11 from scartch and see if I can reproduce the error after FS-1.0.4 install. Thanks folks . Michael Collins wrote: > Try doing this: > http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install > > -MC > > On Tue, Nov 17, 2009 at 7:12 AM, Sam Abekah-Mensah > wrote: > > Hello > > I have tried the same setup but this time using a windows build > FS1.0.4 > on an XP machine and all is fine. The sample 1001 and 1002 IDs work > without any tweaking at all. Could the problem be with the linux > build > 1.0.4.? I am running on an FC11 machine. > > On the FC11 box I used the svn link to build using the ff: > > bootstarp.sh > configure withoout libcurl to eliinate the spidermonkey lib error > make > make install > > Did I miss anything ? > From elihayun at gmail.com Wed Nov 18 04:56:25 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 18 Nov 2009 14:56:25 +0200 Subject: [Freeswitch-users] change event value Message-ID: <4B03EEF9.7070802@savion.huji.ac.il> Hi Is there is a way to intercept an event (for example : REGISTER) and change one of its parameters (for example: the extension number) and fire up the corrected event? I need it to set the speedial of the phone value to be "**xxxxx" but to make it register as "xxxxx" Thanks Eli From siniypin at gmail.com Wed Nov 18 05:07:31 2009 From: siniypin at gmail.com (RobertT) Date: Wed, 18 Nov 2009 16:07:31 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> Message-ID: <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> I've tried to add ;transport=tcp in dialplan bridge application and it has ended up with error on FS with message "can't find registered extension * called_extension*%external_call;transport=tcp" whereas this extension is registered in FS via tcp. Also I tried to reproduce the same scenario with public SIP server and everything worked fine from what I can draw that problem is with my FS configuration or maybe with some FS host's network configuration. Any help will be appretiated. Best regards, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/74eb06b3/attachment.html From brian at freeswitch.org Wed Nov 18 06:24:27 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 08:24:27 -0600 Subject: [Freeswitch-users] acl configuration In-Reply-To: References: Message-ID: <08832B55-1243-498B-88D7-D85E6509AAEF@freeswitch.org> You wouldn't use ACL's at all. You just set auth-calls=false and remove all references to ACL's /b On Nov 18, 2009, at 2:05 AM, Jonas Gauffin wrote: > Hello, > > What should my acl.conf.xml look like if I want to allow ALL calls > on the external profile and use only digest authentication on all > other profiles? > > Thanks, > Jonas > __________ From brian at freeswitch.org Wed Nov 18 06:25:27 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 08:25:27 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> Message-ID: <4EF7A0AD-A76B-4B2D-BA16-EE49C2AB9467@freeswitch.org> AH you are hiding the fact you use FreeSWITCH... Great... but why? /b On Nov 17, 2009, at 11:52 PM, Mathieu Rene wrote: > It needs to go in the profile, not in sofia's global config. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/0e42a604/attachment.html From brian at freeswitch.org Wed Nov 18 06:27:31 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 08:27:31 -0600 Subject: [Freeswitch-users] Registration Error - 408 timeout and now 403 In-Reply-To: <4B03CFC3.7040501@greatiam.com> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> <4B01A8AE.7070708@greatiam.com> <4B02BD53.5040203@greatiam.com> <87f2f3b90911171105s7fb2fea3l316fc2777cbc051a@mail.gmail.com> <4B03CFC3.7040501@greatiam.com> Message-ID: <4271AC30-877E-4E56-975B-77F71C3BD466@freeswitch.org> 403 is Forbidden, So its not really an error you're just getting told NO. You should follow the guide on the wiki on how to debug. 1. Turn on SIP Trace. sofia profile xxx siptrace on 2. Press F8 The error logs are very verbose and usually point to the problem. /b On Nov 18, 2009, at 4:43 AM, Sam Abekah-Mensah wrote: > Thank you so much for your responses. > > I have resolved the problem somehow. > I copied the default.xml from the root conf folder, the sample 1001 > and > 1002 .xml on a windows build to the Fedora 11 machine and that worked. > I guessed the rejection was with the configuration on theFedora > box.even > though it was more straight -out-of-the-box. No one is seeking help > on > this so it must be something I did. > > I am reinstalling FC11 from scartch and see if I can reproduce the > error > after FS-1.0.4 install. > > Thanks folks From brian at freeswitch.org Wed Nov 18 06:27:59 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 08:27:59 -0600 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> Message-ID: <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> How exactly are you doing this? /b On Nov 18, 2009, at 7:07 AM, RobertT wrote: > I've tried to add ;transport=tcp in dialplan bridge application and > it has ended up with error on FS with message "can't find registered > extension called_extension%external_call;transport=tcp" whereas this > extension is registered in FS via tcp. Also I tried to reproduce the > same scenario with public SIP server and everything worked fine from > what I can draw that problem is with my FS configuration or maybe > with some FS host's network configuration. > Any help will be appretiated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/03e9c623/attachment.html From ujjval at simplesignal.com Wed Nov 18 07:17:04 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Wed, 18 Nov 2009 07:17:04 -0800 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> Not sure I am the only one changing User-Agent....but I just want a way for our Customers to know the purpose of the server when they talk to it. There is FreeSwitch written into the SDP "o" line as well...which I don't care about, I want to have something in there that identifies the purpose of the server. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Tuesday, November 17, 2009 10:52 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Changing User-Agent String It needs to go in the profile, not in sofia's global config. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 9:49 PM, Ujjval Karihaloo wrote: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#User_Agent_.5Buser-agent-string.5D As per the above link, we can change the User Agent String, but I added this param name but does not seem to work. [user at freeswitch autoload_configs]$ vi sofia.conf.xml _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/70896d2f/attachment-0001.html From brian at freeswitch.org Wed Nov 18 07:21:55 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 09:21:55 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> Message-ID: <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> you do realize that is NOT the purpose of the user-agent string... changing might break things in some people's configs due to some assumptions made about the user agent on the far side for interop purposes... its your choice to change it but it servers NO purpose doing so. /b On Nov 18, 2009, at 9:17 AM, Ujjval Karihaloo wrote: > Not sure I am the only one changing User-Agent?.but I just want a > way for our Customers to know the purpose of the server when they > talk to it. There is FreeSwitch written into the SDP ?o? line as > well?which I don?t care about, I want to have something in there > that identifies the purpose of the server. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/1fc50096/attachment.html From kristian.kielhofner at gmail.com Wed Nov 18 07:39:24 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 18 Nov 2009 10:39:24 -0500 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> Message-ID: <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> That's *exactly* why I change the User-Agent string. If they're changing their behavior in some way based on the assumption I'm using FreeSWITCH I want to know about it :). On Wed, Nov 18, 2009 at 10:21 AM, Brian West wrote: > you do realize that is NOT the purpose of the user-agent string... changing > might break things in some people's configs due to some assumptions made > about the user agent on the far side for interop purposes... its your choice > to change it but it servers NO purpose doing so. > /b -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Wed Nov 18 08:08:48 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 10:08:48 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> Message-ID: <684091CA-4793-41C6-88D5-BE09146E5713@freeswitch.org> There are a few behaviors in FreeSWITCH that get triggered based on the remote side... you know all about those? :P I'm just saying its usually not wise to change it just in case. /b On Nov 18, 2009, at 9:39 AM, Kristian Kielhofner wrote: > That's *exactly* why I change the User-Agent string. If they're > changing their behavior in some way based on the assumption I'm using > FreeSWITCH I want to know about it :). From oscav at hotmail.fr Wed Nov 18 08:07:42 2009 From: oscav at hotmail.fr (Oscav) Date: Wed, 18 Nov 2009 08:07:42 -0800 (PST) Subject: [Freeswitch-users] sched_broadcast doesn't execute Message-ID: <26408422.post@talk.nabble.com> Hi, I'm writing a script in Javascript that plays a message during a bridge. I'm trying to use a sched_broadcast to do it. The job is scheduled and then deleted but I never hear the wav file and I don't get the "OK Message Scheduled" in the log. It even doesn't display any error message if I specify a wrong file name. Someone could help me on this issue ?? new_session.execute("sched_broadcast", "+20 alloted_timeout ${uuid} playback:ivr-welcome_to_freeswitch.wav"); I already did some posts but I got no answer. This is very difficult to progress without help. Many thanks -- View this message in context: http://old.nabble.com/sched_broadcast-doesn%27t-execute-tp26408422p26408422.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mayamatakeshi at gmail.com Wed Nov 18 08:32:04 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Thu, 19 Nov 2009 01:32:04 +0900 Subject: [Freeswitch-users] mod_fifo and multi-tenancy Message-ID: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> About mod_fifo, it would be safe to use it in multi-tenancy scenarios where domains are created and deleted all the time and in consequence, fifos are created all the time? I mean, fifos are eventually destroyed by mod_fifo itself. Is this correct? br, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/0dfd376c/attachment.html From robert.hadley at teotech.com Wed Nov 18 08:40:16 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Wed, 18 Nov 2009 08:40:16 -0800 Subject: [Freeswitch-users] Anybody interested in helping fix the -srcdir option? Message-ID: <77F5794BDB0A4FD4BD7592D70F4719D4@greyhawk.tonecommander.com> Hi All, Anybody interested in helping fix the -srcdir option? I am trying to build in a subdirectory off the Freeswitch source. I am working on it and finding issues. However, being a newbie at autoconf/automake and shell scripting I sometimes struggle at finding fixes. For example, the script command below is in bootstrap.sh, but might need to be moved or duplicated in configure.* to support using configure -srcdir option, as the modules.conf file also needs to be to the build destination folder. if [ ! -f modules.conf ]; then cp build/modules.conf.in modules.conf fi Thanks, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/80dfcf50/attachment.html From kristian.kielhofner at gmail.com Wed Nov 18 08:42:10 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 18 Nov 2009 11:42:10 -0500 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <684091CA-4793-41C6-88D5-BE09146E5713@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> <684091CA-4793-41C6-88D5-BE09146E5713@freeswitch.org> Message-ID: <2d9149cd0911180842t547bc6fdt526bbc9399485141@mail.gmail.com> Brian, I see some things based on SDP originator (we all know what those are about) but nothing for SIP user agent... More curious than anything else, am I missing something? On Wed, Nov 18, 2009 at 11:08 AM, Brian West wrote: > There are a few behaviors in FreeSWITCH that get triggered based on > the remote side... you know all about those? ?:P ?I'm just saying its > usually not wise to change it just in case. > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From mike at jerris.com Wed Nov 18 08:43:35 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Nov 2009 11:43:35 -0500 Subject: [Freeswitch-users] Compilation problem In-Reply-To: <32c1b333.68a1501e.4b02ad93.bbfcd@go2.pl> References: <32c1b333.68a1501e.4b02ad93.bbfcd@go2.pl> Message-ID: <5377DF41-8B48-443D-AD65-E4C5F82E58D7@jerris.com> This issue is now fixed in trunk. Mike On Nov 17, 2009, at 9:05 AM, Christopher Z. wrote: > Hi, > > I've got this error after make: > > http://pastebin.freeswitch.org/11145 > > Any idea how to fix this error ? From abeka at greatiam.com Wed Nov 18 08:44:10 2009 From: abeka at greatiam.com (Samuel 'Otis' Abekah-Mensah) Date: Wed, 18 Nov 2009 16:44:10 +0000 Subject: [Freeswitch-users] Registration Error - 408 timeout and now 403 In-Reply-To: <4271AC30-877E-4E56-975B-77F71C3BD466@freeswitch.org> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> <4B01A8AE.7070708@greatiam.com> <4B02BD53.5040203@greatiam.com> <87f2f3b90911171105s7fb2fea3l316fc2777cbc051a@mail.gmail.com> <4B03CFC3.7040501@greatiam.com> <4271AC30-877E-4E56-975B-77F71C3BD466@freeswitch.org> Message-ID: <4B04245A.8080000@greatiam.com> Thanks. I will look up how to debug on the wiki. Kinda late now with my setup; got to learn it anyway. Thanks for the direction. Brian West wrote: >
403 is > Forbidden, So its not really an error you're just getting told NO. > You should follow the guide on the wiki on how to debug. > > 1. Turn on SIP Trace. sofia profile xxx siptrace on > 2. Press F8 > > The error logs are very verbose and usually point to the problem. > > /b > > On Nov 18, 2009, at 4:43 AM, Sam Abekah-Mensah wrote: > >> Thank you so much for your responses. >> >> I have resolved the problem somehow. >> I copied the default.xml from the root conf folder, the sample 1001 and >> 1002 .xml on a windows build to the Fedora 11 machine and that worked. >> I guessed the rejection was with the configuration on theFedora box.even >> though it was more straight -out-of-the-box. No one is seeking help on >> this so it must be something I did. >> >> I am reinstalling FC11 from scartch and see if I can reproduce the error >> after FS-1.0.4 install. >> >> Thanks folks > > > >
From brian at freeswitch.org Wed Nov 18 09:14:59 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 11:14:59 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <2d9149cd0911180842t547bc6fdt526bbc9399485141@mail.gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> <684091CA-4793-41C6-88D5-BE09146E5713@freeswitch.org> <2d9149cd0911180842t547bc6fdt526bbc9399485141@mail.gmail.com> Message-ID: <7E7166FF-1824-4F68-957E-898B4AEBB40E@freeswitch.org> We do in sofia_reg and sofia_sla, and some where we do updates so we don't send things to phones we know will piss their pants. /b On Nov 18, 2009, at 10:42 AM, Kristian Kielhofner wrote: > Brian, > > I see some things based on SDP originator (we all know what those > are about) but nothing for SIP user agent... From steveu at coppice.org Wed Nov 18 09:45:37 2009 From: steveu at coppice.org (Steve Underwood) Date: Thu, 19 Nov 2009 01:45:37 +0800 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <2d9149cd0911171219y744cb81cwf53c5d25ea26c05e@mail.gmail.com> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> <2d9149cd0911171219y744cb81cwf53c5d25ea26c05e@mail.gmail.com> Message-ID: <4B0432C1.7000006@coppice.org> On 11/18/2009 04:19 AM, Kristian Kielhofner wrote: > Ah yes, using spandsp instead of libvoipcodecs. I'm not going to > question the wisdom of that move but it appears that spandsp (as-is) > doesn't cross compile properly (make_at_dictionary is built using the > cross compiler and can't run on the host). Once that error is fixed > (I hacked it for now) it still bombs as shown here: > > http://pastebin.freeswitch.org/11149 > > spandsp + libtiff are almost certainly *much* larger than > libvoipcodecs but if that means that I can also build mod_fax I guess > it's worth it ;). > > On Tue, Nov 17, 2009 at 2:37 PM, Brian West wrote: > >> OH you need spandsp for VoipCodecs. No way around that one. >> /b >> > Over time more and more of spandsp will be used by Freeswitch, so its most certainly an integral part of FS going forward. In a few months it might be possible to not use libTIFF, depending how things go with some developments. spandsp builds OK for many cross compile setups. make_at_dictionary and make_modem_filters should be built using the host compiler, not the target compiler. This seems to work in the places I've tried it. The problem in your pastebin log seems to be a broken C99 environment, and not a spandsp problem. Steve From helmut.kuper at ewetel.de Wed Nov 18 10:21:38 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 18 Nov 2009 19:21:38 +0100 Subject: [Freeswitch-users] Question about odbc support Message-ID: <4B043B32.20802@ewetel.de> Hi, does anybody know how to check the affected rows caused by delete, insert or update sql statements in FS? To do this with sqlite3 there is a function called switch_core_db_changes(). regards helmut From kristian.kielhofner at gmail.com Wed Nov 18 10:53:22 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 18 Nov 2009 13:53:22 -0500 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <7E7166FF-1824-4F68-957E-898B4AEBB40E@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> <684091CA-4793-41C6-88D5-BE09146E5713@freeswitch.org> <2d9149cd0911180842t547bc6fdt526bbc9399485141@mail.gmail.com> <7E7166FF-1824-4F68-957E-898B4AEBB40E@freeswitch.org> Message-ID: <2d9149cd0911181053u20644421q725809f0a6a40377@mail.gmail.com> Ah, that explains it. I don't do any registration or presence/sla with FS - yet ;). On Wed, Nov 18, 2009 at 12:14 PM, Brian West wrote: > We do in sofia_reg and sofia_sla, and some where we do updates so we > don't send things to phones we know will piss their pants. > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From kristian.kielhofner at gmail.com Wed Nov 18 11:02:44 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 18 Nov 2009 14:02:44 -0500 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <4B0432C1.7000006@coppice.org> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> <2d9149cd0911171219y744cb81cwf53c5d25ea26c05e@mail.gmail.com> <4B0432C1.7000006@coppice.org> Message-ID: <2d9149cd0911181102h92309e8recc7f46e8f81bd88@mail.gmail.com> On Wed, Nov 18, 2009 at 12:45 PM, Steve Underwood wrote: > Over time more and more of spandsp will be used by Freeswitch, so its > most certainly an integral part of FS going forward. In a few months it > might be possible to not use libTIFF, depending how things go with some > developments. > > spandsp builds OK for many cross compile setups. make_at_dictionary and > make_modem_filters should be built using the host compiler, not the > target compiler. This seems to work in the places I've tried it. > > The problem in your pastebin log seems to be a broken C99 environment, > and not a spandsp problem. > > Steve > make_at_dictionary is not build using the host compiler. I had to hack it (manually passing CC and LIBTOOL to make) to get the build to proceed to the next error... This may be specific to the integration with the rest of the FreeSWITCH build system. I'm using uClibc (as are most other embedded environments) and I've had other C99 issues before. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Wed Nov 18 11:28:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Nov 2009 13:28:44 -0600 Subject: [Freeswitch-users] sched_broadcast doesn't execute In-Reply-To: <26408422.post@talk.nabble.com> References: <26408422.post@talk.nabble.com> Message-ID: <191c3a030911181128g35ba0652w9fc575d5586367dc@mail.gmail.com> is that your exact code? ${uuid} will not be expanded by javascript var uuid = session.getVariable(uuid); new_session.execute("sched_broadcast", "+20 alloted_timeout " + uuid + " playback:ivr-welcome_to_freeswitch.wav"); On Wed, Nov 18, 2009 at 10:07 AM, Oscav wrote: > > Hi, > > I'm writing a script in Javascript that plays a message during a bridge. > I'm > trying to use a sched_broadcast to do it. The job is scheduled and then > deleted but I never hear the wav file and I don't get the "OK Message > Scheduled" in the log. It even doesn't display any error message if I > specify a wrong file name. Someone could help me on this issue ?? > > new_session.execute("sched_broadcast", "+20 alloted_timeout ${uuid} > playback:ivr-welcome_to_freeswitch.wav"); > > I already did some posts but I got no answer. This is very difficult to > progress without help. > > Many thanks > -- > View this message in context: > http://old.nabble.com/sched_broadcast-doesn%27t-execute-tp26408422p26408422.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/eb531511/attachment.html From william.nishio at voicetechnology.com.br Wed Nov 18 11:38:24 2009 From: william.nishio at voicetechnology.com.br (William Kendi ...) Date: Wed, 18 Nov 2009 17:38:24 -0200 Subject: [Freeswitch-users] mod dptools record problem - hangup channel with invalid file path Message-ID: <67d615ac0911181138m30f3064ci1a2dad6732354e35@mail.gmail.com> While I was testing the "mod dptools record" application using invalid file paths, i noted that the "mod dptools record" application terminated the call. I am currently looking for a way to change this behaviour. Any suggestions? Can this be done? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/3d073d11/attachment.html From mike at jerris.com Wed Nov 18 11:40:11 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Nov 2009 14:40:11 -0500 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <2d9149cd0911181102h92309e8recc7f46e8f81bd88@mail.gmail.com> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> <2d9149cd0911171219y744cb81cwf53c5d25ea26c05e@mail.gmail.com> <4B0432C1.7000006@coppice.org> <2d9149cd0911181102h92309e8recc7f46e8f81bd88@mail.gmail.com> Message-ID: <9E79AB11-C3CB-4418-9360-69497550F053@jerris.com> Kristian, catch up with me somewhere that I can get remote access to this build environment so that we can sort this out. Mike On Nov 18, 2009, at 2:02 PM, Kristian Kielhofner wrote: > On Wed, Nov 18, 2009 at 12:45 PM, Steve Underwood wrote: >> Over time more and more of spandsp will be used by Freeswitch, so its >> most certainly an integral part of FS going forward. In a few months it >> might be possible to not use libTIFF, depending how things go with some >> developments. >> >> spandsp builds OK for many cross compile setups. make_at_dictionary and >> make_modem_filters should be built using the host compiler, not the >> target compiler. This seems to work in the places I've tried it. >> >> The problem in your pastebin log seems to be a broken C99 environment, >> and not a spandsp problem. >> >> Steve >> > > make_at_dictionary is not build using the host compiler. I had to > hack it (manually passing CC and LIBTOOL to make) to get the build to > proceed to the next error... This may be specific to the integration > with the rest of the FreeSWITCH build system. > > I'm using uClibc (as are most other embedded environments) and I've > had other C99 issues before. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Nov 18 11:45:43 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 13:45:43 -0600 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <4B032142.1000308@tigertech.com> References: <4B032142.1000308@tigertech.com> Message-ID: <4256FA16-DE2A-4871-A398-30CC0E8E6D4C@freeswitch.org> Can you tell us what kind of phone you're using? And have you tried this on SVN trunk? /b On Nov 17, 2009, at 4:18 PM, Robert L Mathews wrote: > I'm using FreeSWITCH 1.0.4. > > When I make a call from a SIP phone to either a conference or an echo > test on the FreeSWITCH server, the latency ("lag") starts off very low > -- a fraction of a second. But as several minutes of time goes by, > the > lag increases. After, say, 15 minutes, the lag will reach a couple of > seconds, making conference calls unusable. > > This does not happen on pure SIP-to-SIP calls, even when FreeSWITCH is > handling the RTP media. > > If I hang up and immediately call back in (even to the same > conference), > the lag is reset to almost zero. If I put the call on "hold" and > take it > off hold, the lag is also gone. > > During testing, I've found that this may be related to the freeswitch > app on the server not getting all the CPU time it wants. > > If I suspend the freeswitch process for two seconds and then resume > it, > the sound stops for two seconds, as I'd expect. But the echo/ > conference > calls that were active are then lagged by two seconds until they > hang up > (or get put on hold), even after freeswitch is resumed and getting all > the CPU time it needs. > > This is easily reproduced by making a SIP call to the echo test > module, > then: > > pkill -STOP freeswitch; sleep 2; pkill -CONT freeswitch > > Any echo test or conference call that was in progress will then be > permanently lagged by two seconds. However, any SIP-to-SIP calls that > were in progress will not become lagged. > > Of course, killing it with -STOP is an artificially nasty thing to do. > But it effectively just prevents it from being scheduled on the CPU > for > a short period of time, and I can duplicate the same behavior (more > gradually) by just increasing the load on the machine to the point > that > the freeswitch app isn't getting much CPU time. > > Just for the record, I get the same results from several different > phones and several different Internet connections, all of which have a > ping latency of under 40 ms to the server. This problem does not > happen > using the same phones and network connections to an asterisk server. > > Throwing out an even more complicated example that I've encountered: > If > I have a SIP-to-SIP call going from party A to party B and I stop the > process for two seconds, it doesn't permanently introduce lag to that > call, as I mentioned. But if a third person (party C) starts > eavesdropping on the call and presses "3" to make it a three way call, > and then I suspend it for two seconds, the call between A and B isn't > lagged, but what party C hears and sends *is* lagged. > > Any ideas on how to fix this? Do other people see the same thing > happening? As I said, the gradual increase in lag over a long period > of > time makes long conferences unusable, unfortunately. > > -- > Rob > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Wed Nov 18 11:46:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Nov 2009 13:46:47 -0600 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <4B032142.1000308@tigertech.com> References: <4B032142.1000308@tigertech.com> Message-ID: <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> Have you tried SVN trunk? 1.0.4 is several months old and will soon be replaced with 1.0.5 so if you can find similar problems on trunk it would be more helpful than possibly pointing out issues that we have already fixed. On Tue, Nov 17, 2009 at 4:18 PM, Robert L Mathews wrote: > I'm using FreeSWITCH 1.0.4. > > When I make a call from a SIP phone to either a conference or an echo > test on the FreeSWITCH server, the latency ("lag") starts off very low > -- a fraction of a second. But as several minutes of time goes by, the > lag increases. After, say, 15 minutes, the lag will reach a couple of > seconds, making conference calls unusable. > > This does not happen on pure SIP-to-SIP calls, even when FreeSWITCH is > handling the RTP media. > > If I hang up and immediately call back in (even to the same conference), > the lag is reset to almost zero. If I put the call on "hold" and take it > off hold, the lag is also gone. > > During testing, I've found that this may be related to the freeswitch > app on the server not getting all the CPU time it wants. > > If I suspend the freeswitch process for two seconds and then resume it, > the sound stops for two seconds, as I'd expect. But the echo/conference > calls that were active are then lagged by two seconds until they hang up > (or get put on hold), even after freeswitch is resumed and getting all > the CPU time it needs. > > This is easily reproduced by making a SIP call to the echo test module, > then: > > pkill -STOP freeswitch; sleep 2; pkill -CONT freeswitch > > Any echo test or conference call that was in progress will then be > permanently lagged by two seconds. However, any SIP-to-SIP calls that > were in progress will not become lagged. > > Of course, killing it with -STOP is an artificially nasty thing to do. > But it effectively just prevents it from being scheduled on the CPU for > a short period of time, and I can duplicate the same behavior (more > gradually) by just increasing the load on the machine to the point that > the freeswitch app isn't getting much CPU time. > > Just for the record, I get the same results from several different > phones and several different Internet connections, all of which have a > ping latency of under 40 ms to the server. This problem does not happen > using the same phones and network connections to an asterisk server. > > Throwing out an even more complicated example that I've encountered: If > I have a SIP-to-SIP call going from party A to party B and I stop the > process for two seconds, it doesn't permanently introduce lag to that > call, as I mentioned. But if a third person (party C) starts > eavesdropping on the call and presses "3" to make it a three way call, > and then I suspend it for two seconds, the call between A and B isn't > lagged, but what party C hears and sends *is* lagged. > > Any ideas on how to fix this? Do other people see the same thing > happening? As I said, the gradual increase in lag over a long period of > time makes long conferences unusable, unfortunately. > > -- > Rob > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/cf37b5ba/attachment-0001.html From mike at jerris.com Wed Nov 18 11:48:44 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Nov 2009 14:48:44 -0500 Subject: [Freeswitch-users] Anybody interested in helping fix the -srcdir option? In-Reply-To: <77F5794BDB0A4FD4BD7592D70F4719D4@greyhawk.tonecommander.com> References: <77F5794BDB0A4FD4BD7592D70F4719D4@greyhawk.tonecommander.com> Message-ID: Fixed in svn r15526 and other fixes in svn r15527. mike On Nov 18, 2009, at 11:40 AM, Robert Hadley wrote: > Hi All, > > Anybody interested in helping fix the ?srcdir option? I am trying to build in a subdirectory off the Freeswitch source. I am working on it and finding issues. However, being a newbie at autoconf/automake and shell scripting I sometimes struggle at finding fixes. > > For example, the script command below is in bootstrap.sh, but might need to be moved or duplicated in configure.* to support using configure ?srcdir option, as the modules.conf file also needs to be to the build destination folder. > > if [ ! -f modules.conf ]; then > cp build/modules.conf.in modules.conf > fi > > Thanks, > Robert > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/07a46d7e/attachment.html From mike at jerris.com Wed Nov 18 11:53:05 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Nov 2009 14:53:05 -0500 Subject: [Freeswitch-users] mod dptools record problem - hangup channel with invalid file path In-Reply-To: <67d615ac0911181138m30f3064ci1a2dad6732354e35@mail.gmail.com> References: <67d615ac0911181138m30f3064ci1a2dad6732354e35@mail.gmail.com> Message-ID: <994A83CB-7069-4808-9055-30B8BD3CEA75@jerris.com> Okay, I'll ask the obvious question. Why are you passing record invalid file paths and why should it not fail if you do? Mike On Nov 18, 2009, at 2:38 PM, William Kendi ... wrote: > While I was testing the "mod dptools record" application using invalid file paths, i noted that the "mod dptools record" application terminated the call. > I am currently looking for a way to change this behaviour. > Any suggestions? Can this be done? From william.nishio at voicetechnology.com.br Wed Nov 18 12:26:00 2009 From: william.nishio at voicetechnology.com.br (William Kendi ...) Date: Wed, 18 Nov 2009 18:26:00 -0200 Subject: [Freeswitch-users] mod dptools record problem - hangup channel with invalid file path In-Reply-To: <994A83CB-7069-4808-9055-30B8BD3CEA75@jerris.com> References: <67d615ac0911181138m30f3064ci1a2dad6732354e35@mail.gmail.com> <994A83CB-7069-4808-9055-30B8BD3CEA75@jerris.com> Message-ID: <67d615ac0911181226y22b4fec6ndb8e622a24db101c@mail.gmail.com> Actually, I am integrating FreeSWITCH with a weird IVR Framework, and the current behaviour of the "mod dptools record" application breaks some rules of the weird IVR Framework that must be integrated with FreeSWITCH. In order to integrate FreeSWITCH with the weird IVR Framework, the "mod dptools record" application mustn't terminate the call when invalid file paths are passed, and a notification of the invalid file path through the event system of FreeSWITCH should be enough for me, like the behaviour of the "mod dptools playback" application when invalid file paths are passed. Thanks in advance. ** 2009/11/18 Michael Jerris > Okay, I'll ask the obvious question. Why are you passing record invalid > file paths and why should it not fail if you do? > > Mike > > On Nov 18, 2009, at 2:38 PM, William Kendi ... wrote: > > > While I was testing the "mod dptools record" application using invalid > file paths, i noted that the "mod dptools record" application terminated the > call. > > I am currently looking for a way to change this behaviour. > > Any suggestions? Can this be done? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/2d9c412e/attachment.html From timuckun at gmail.com Wed Nov 18 12:39:18 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 19 Nov 2009 09:39:18 +1300 Subject: [Freeswitch-users] Hardware echo cancellation. Message-ID: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> I am about to build a new machine as a VOIP server. I am going to get either a quad core intel or a six core AMD processor with at least eight gigabytes of RAM in it. Given that much horsepower I am wondering if there is any need to purchase hardware with echo cancellation (I am thinking about redfone devices).. I can save some money by not getting the echo cancellation. So is it worth saving that money? Is it always better to have hardware echo cancellation? Is a quad core capable of dealing with echo cancellation needs of an IVR which is going to take lots of simultaneous calls? Thanks. From mrene_lists at avgs.ca Wed Nov 18 12:52:29 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 18 Nov 2009 12:52:29 -0800 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> Message-ID: <6A7DC321-F2E6-493F-ACFD-0373950D9659@avgs.ca> If you have TDM hardware, Buy the echo canceller. It's worth it. It's not required for standard SIP calls though, only when dealing with analog or TDM circuits. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 18-Nov-09, at 12:39 PM, Tim Uckun wrote: > I am about to build a new machine as a VOIP server. I am going to get > either a quad core intel or a six core AMD processor with at least > eight gigabytes of RAM in it. Given that much horsepower I am > wondering if there is any need to purchase hardware with echo > cancellation (I am thinking about redfone devices).. I can save some > money by not getting the echo cancellation. > > So is it worth saving that money? Is it always better to have hardware > echo cancellation? Is a quad core capable of dealing with echo > cancellation needs of an IVR which is going to take lots of > simultaneous calls? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sicfslist at gmail.com Wed Nov 18 13:00:09 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 18 Nov 2009 15:00:09 -0600 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> Message-ID: <4B046059.3090104@gmail.com> Given that it's an IVR system I think you'll find the DTMF results better with hardware based echo cans ... and to be frank the hardware based echo cancellations are really not that much more expensive on cards from folks like Sangoma. SDR Tim Uckun wrote: > I am about to build a new machine as a VOIP server. I am going to get > either a quad core intel or a six core AMD processor with at least > eight gigabytes of RAM in it. Given that much horsepower I am > wondering if there is any need to purchase hardware with echo > cancellation (I am thinking about redfone devices).. I can save some > money by not getting the echo cancellation. > > So is it worth saving that money? Is it always better to have hardware > echo cancellation? Is a quad core capable of dealing with echo > cancellation needs of an IVR which is going to take lots of > simultaneous calls? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dave at 3c.co.uk Wed Nov 18 13:04:09 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 18 Nov 2009 14:04:09 -0700 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> Message-ID: <1258578249.12820.264.camel@localhost.localdomain> Hi Tim, Here you go: http://old.nabble.com/echo-cancellation-on-PRI-cards-td22552605.html > I am about to build a new machine as a VOIP server. I am going to get > either a quad core intel or a six core AMD processor with at least > eight gigabytes of RAM in it. Given that much horsepower I am > wondering if there is any need to purchase hardware with echo > cancellation (I am thinking about redfone devices).. I can save some > money by not getting the echo cancellation. > > So is it worth saving that money? Is it always better to have hardware > echo cancellation? Is a quad core capable of dealing with echo > cancellation needs of an IVR which is going to take lots of > simultaneous calls? In (very) brief: maybe, no, and depends on the definition of 'lots'. --Dave From nicolas at medularis.com Wed Nov 18 13:13:33 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Wed, 18 Nov 2009 18:13:33 -0300 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> Message-ID: <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> I had a voip provider which wouldn't accept calls from Freeswitch because of the user-agent string. I had to change it to "Asterisk" and then everything worked. Nico On Wed, Nov 18, 2009 at 12:21 PM, Brian West wrote: > you do realize that is NOT the purpose of the user-agent string... changing > might break things in some people's configs due to some assumptions made > about the user agent on the far side for interop purposes... its your choice > to change it but it servers NO purpose doing so. > > /b > > On Nov 18, 2009, at 9:17 AM, Ujjval Karihaloo wrote: > > Not sure I am the only one changing *User-Agent*?.but I just want a way > for our Customers to know the purpose of the server when they talk to it. > There is FreeSwitch written into the SDP ?o? line as well?which I don?t care > about, I want to have something in there that identifies the purpose of the > server. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/bfd09469/attachment-0001.html From brian at freeswitch.org Wed Nov 18 13:20:37 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 15:20:37 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> Message-ID: Sounds like you need to take a baseball bat to their forehead. /b On Nov 18, 2009, at 3:13 PM, Nicolas Brenner wrote: > I had a voip provider which wouldn't accept calls from Freeswitch > because of the user-agent string. I had to change it to "Asterisk" > and then everything worked. > > Nico From rob4manhere at gmail.com Wed Nov 18 13:30:32 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Wed, 18 Nov 2009 15:30:32 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> Message-ID: lol! we have to play nice in the wiki but the mailing list is another story. On Nov 18, 2009, at 3:20 PM, Brian West wrote: > Sounds like you need to take a baseball bat to their forehead. > > /b > > On Nov 18, 2009, at 3:13 PM, Nicolas Brenner wrote: > >> I had a voip provider which wouldn't accept calls from Freeswitch >> because of the user-agent string. I had to change it to "Asterisk" >> and then everything worked. >> >> Nico > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lists at tigertech.com Wed Nov 18 13:33:30 2009 From: lists at tigertech.com (Robert L Mathews) Date: Wed, 18 Nov 2009 13:33:30 -0800 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> References: <4B032142.1000308@tigertech.com> <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> Message-ID: <4B04682A.6000309@tigertech.com> Anthony Minessale wrote: > Have you tried SVN trunk? No, I haven't. I'll try it. (But also, if others want to try seeing if it happens, it's trivially duplicated by calling the echo test app and sending the freeswitch server process a "-STOP" signal, sleeping for a second, then sending it a "-CONT" signal.) In any case, though, I have partially found the cause of the problem -- at least in the echo test module in 1.0.4. It's the same problem reported here: http://www.mail-archive.com/freeswitch-users%40lists.freeswitch.org/msg15800.html The two suggestions there, explicitly setting "rtp-autoflush-during-bridge" true and "rtp_timer_name=none", didn't fix it for me. (The first is no surprise because that's the default anyway.) However, setting the (undocumented?) parameter "rtp-autoflush" to true *did* fix it in the echo test (but not the conference). I think what's happening is that FreeSWITCH contains code to detect when we've "fallen behind" the RTP audio. It looks like this happens in rtp_common_read() of switch_rtp.c: if the code detects that there are "extra" RTP packets waiting during several consecutive rtp_common_read() calls, switch_core_timer_sync() is called to catch up. This code is apparently used during bridged calls when "rtp-autoflush-during-bridge" is true (the default), which explains why I don't have the problem during SIP-to-SIP calls. However, I'm guessing that echo test calls somehow aren't considered "bridged" by that code. Therefore the code isn't being used unless "rtp-autoflush" is true. That thread suggests that this is probably a phone or network problem, but it seems to me that even if all the timing is perfect, this problem will happen if the freeswitch server thread doesn't get enough CPU time to retrieve a packet before the next one arrives. For example, if packets arrive every 20 ms but high load on the server causes the process to sleep for 25 ms, so that two packets are waiting the next time the process runs, it will never catch up that extra packet -- the echo test or conference will now be permanently 20 ms behind. And if that happens again, it will now be 40 ms behind, and so on. That explains why the lag slowly increases over time. Does that make sense? I don't quite understand why the "catch up" code isn't always used for all RTP streams: if an RTP packet poll repeatedly shows that there are extra audio packets waiting, it seems to make sense that we'd always want to catch up. Anyway, as I said, I'm still having the conference problem, even with "rtp-autoflush" enabled. So I need to try the svn trunk version and see if it still happens, then track it down further if so. I will report back. -- Robert L Mathews, Tiger Technologies From timuckun at gmail.com Wed Nov 18 13:36:12 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 19 Nov 2009 10:36:12 +1300 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258578249.12820.264.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> Message-ID: <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> On Thu, Nov 19, 2009 at 10:04 AM, David Knell wrote: > Hi Tim, > > Here you go: > http://old.nabble.com/echo-cancellation-on-PRI-cards-td22552605.html > Thanks. That's almost exactly the same situation as the one I am going to find myself in. > > In (very) brief: maybe, no, and depends on the definition of 'lots'. > By lots I mean somewhere between 50 to a 100 but it's mostly an IVR application so all it will be doing is either playing prompts or recording messages. Almost no live conversations. From msc at freeswitch.org Wed Nov 18 13:36:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Nov 2009 13:36:34 -0800 Subject: [Freeswitch-users] mod_fifo and multi-tenancy In-Reply-To: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> References: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> Message-ID: <87f2f3b90911181336w2b0f6cb8s211bc235a6f9084d@mail.gmail.com> On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi wrote: > About mod_fifo, it would be safe to use it in multi-tenancy scenarios where > domains are created and deleted all the time and in consequence, fifos are > created all the time? I mean, fifos are eventually destroyed by mod_fifo > itself. Is this correct? > > br, > takeshi > > No, FIFOs are not "destroyed" automatically just because the last member goes away. Tony says that an empty FIFO takes up almost no memory so performance shouldn't be an issue. You can always issue the API command: *fifo reparse del_all* to clean everything out if you feel like things are getting out of hand. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/5d38dbaa/attachment.html From msc at freeswitch.org Wed Nov 18 13:39:33 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Nov 2009 13:39:33 -0800 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> Message-ID: <87f2f3b90911181339w114f9bffl7837e03d30de1cf4@mail.gmail.com> On Wed, Nov 18, 2009 at 1:36 PM, Tim Uckun wrote: > On Thu, Nov 19, 2009 at 10:04 AM, David Knell wrote: > > Hi Tim, > > > > Here you go: > > http://old.nabble.com/echo-cancellation-on-PRI-cards-td22552605.html > > > Thanks. That's almost exactly the same situation as the one I am going > to find myself in. > > > > > In (very) brief: maybe, no, and depends on the definition of 'lots'. > > > > By lots I mean somewhere between 50 to a 100 but it's mostly an IVR > application so all it will be doing is either playing prompts or > recording messages. Almost no live conversations. > > Just get the HW echo can - all the COOL kids are doing it! :P -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/249b92cb/attachment.html From brian at freeswitch.org Wed Nov 18 14:05:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 16:05:12 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> Message-ID: <51686C79-3122-4B3B-AD06-F9CEF175E023@freeswitch.org> Well this is a bit more informal vs the wiki where people take it as fact! :) Plus its a little humpday humor! /b On Nov 18, 2009, at 3:30 PM, Rob Forman wrote: > lol! > > we have to play nice in the wiki but the mailing list is another > story. > > > On Nov 18, 2009, at 3:20 PM, Brian West wrote: > >> Sounds like you need to take a baseball bat to their forehead. >> >> /b From hads at nice.net.nz Wed Nov 18 14:06:17 2009 From: hads at nice.net.nz (Hadley Rich) Date: Thu, 19 Nov 2009 11:06:17 +1300 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258578249.12820.264.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> Message-ID: <1258581977.6280.5.camel@sodium> On Wed, 2009-11-18 at 14:04 -0700, David Knell wrote: > In (very) brief: maybe, no, and depends on the definition of 'lots'. Most people say yes. -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier From dfansler at dv-fansler.com Wed Nov 18 13:49:02 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Wed, 18 Nov 2009 16:49:02 -0500 Subject: [Freeswitch-users] APT Utility Message-ID: <005a01ca6898$f16d99d0$d448cd70$@com> Greetings - I am trying to startup a freeSwitch on a P4 running Ubuntu 9.04 "Jaunty". I know very little about Linux. I decided to try this after reading the article in Linux Pro Magazine. I have been following the detailed instructions in the wiki for using Ubuntu Jaunty, however I have run into an unknown - "Use your favorite APT utility to get the needed packages". I am good at following direct instructions - but this statement is too vague for my minimal minimal - did I mention minimal - knowledge of Linux. Could someone please give me detailed instructions on how to use APT utility to get the needed packages - and what are the needed packages? Thanks kindly, David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/e357620b/attachment-0001.html From rob4manhere at gmail.com Wed Nov 18 14:52:51 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Wed, 18 Nov 2009 16:52:51 -0600 Subject: [Freeswitch-users] APT Utility In-Reply-To: <005a01ca6898$f16d99d0$d448cd70$@com> References: <005a01ca6898$f16d99d0$d448cd70$@com> Message-ID: Hi David, When using Apt, you would install packages with: apt-get install Or search for packages with apt-cache search If you're not root, you'll need to stick "sudo " in front of those command. Honestly, you might want to find a better tutorial with explicit command-by-command instructions. Good luck! Rob On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: > Greetings ? I am trying to startup a freeSwitch on a P4 running > Ubuntu 9.04 ?Jaunty?. I know very little about Linux. I decided to > try this after reading the article in Linux Pro Magazine. I have > been following the detailed instructions in the wiki for using > Ubuntu Jaunty, however I have run into an unknown ? ?Use your > favorite APT utility to get the needed packages?. > I am good at following direct instructions ? but this statement is > too vague for my minimal minimal ? did I mention minimal - knowledge > of Linux. > > Could someone please give me detailed instructions on how to use APT > utility to get the needed packages ? and what are the needed packages? > Thanks kindly, > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/e3da4325/attachment.html From sicfslist at gmail.com Wed Nov 18 14:58:05 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 18 Nov 2009 16:58:05 -0600 Subject: [Freeswitch-users] APT Utility In-Reply-To: <005a01ca6898$f16d99d0$d448cd70$@com> References: <005a01ca6898$f16d99d0$d448cd70$@com> Message-ID: <4B047BFD.8000007@gmail.com> David, apt is the pack management system for the debian line of distros (including Ubuntu). Google apt-get or aptitude for more information on the utility. This probably isn't the best list for topics not related to FS (apt is a linux utility) ... but here is a brief rundown: apt-get install make flex patch gcc g++ autoconf automake libtool libncurses5-dev ncurses-dev python-MySQLdb subversion -y cd /usr/src ## DO ONE OF THE FOLLOWING TO GET THE SRC (TRUNK IS BEST FOR NOW -- NO STABLE RELEASE)## svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk or svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch cd freeswitch/ ./bootstrap.sh ./configure make make install Type that as you see it on the command line (do sudo -i and type in the root password first). I will say that FS requires some basic knowledge of Linux to get running ... and certainly to manage / maintain. Try digging around for some Ubuntu how to's for some basic info. Hope this helps. SDR David V. Fansler wrote: > > Greetings ? I am trying to startup a freeSwitch on a P4 running Ubuntu > 9.04 ?Jaunty?. I know very little about Linux. I decided to try this > after reading the article in Linux Pro Magazine. I have been following > the detailed instructions in the wiki for using Ubuntu Jaunty, however > I have run into an unknown ? ?Use your favorite *APT* utility to get > the needed packages?. > > I am good at following direct instructions ? but this statement is too > vague for my minimal minimal ? did I mention minimal - knowledge of Linux. > > Could someone please give me detailed instructions on how to use APT > utility to get the needed packages ? and what are the needed packages? > > Thanks kindly, > > David > > David V. Fansler > > s/v Annabelle > > dfansler at dv-fansler.com > > www.dv-fansler.com > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Wed Nov 18 15:10:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Nov 2009 17:10:27 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <51686C79-3122-4B3B-AD06-F9CEF175E023@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> <51686C79-3122-4B3B-AD06-F9CEF175E023@freeswitch.org> Message-ID: <191c3a030911181510i4c8d36brc03fa4063b088c93@mail.gmail.com> maybe you could send them 183 then 4 180's or send them an invite and pretend to deadlock and not send any more sip traffic as a way of identifying yourself On Wed, Nov 18, 2009 at 4:05 PM, Brian West wrote: > Well this is a bit more informal vs the wiki where people take it as > fact! :) Plus its a little humpday humor! > > /b > > On Nov 18, 2009, at 3:30 PM, Rob Forman wrote: > > > lol! > > > > we have to play nice in the wiki but the mailing list is another > > story. > > > > > > On Nov 18, 2009, at 3:20 PM, Brian West wrote: > > > >> Sounds like you need to take a baseball bat to their forehead. > >> > >> /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/3e07c844/attachment.html From anthony.minessale at gmail.com Wed Nov 18 15:28:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Nov 2009 17:28:30 -0600 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <4B04682A.6000309@tigertech.com> References: <4B032142.1000308@tigertech.com> <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> <4B04682A.6000309@tigertech.com> Message-ID: <191c3a030911181528j7a38ce32gb2fc6fdd585932a9@mail.gmail.com> I can promise you that much of your problems will be solved with latest SVN. There was a small change in the timer sync that was causing your symptoms in specific cases and it's been resolved for months. Did you answer the question about what phones? I'm going to guess Cisco based on the symptoms. non bridge calls use a timer to make sure the audio is coming in at a steady rate to ensure bursting RTP is played at the correct rate. Stopping it and restarting 2 seconds later will cause a delay by design because you have suspended the process but not the UDP stack. If you don't want to use rtp-timer you can disable it in the profile settings for rtp-timer-name by setting it to "none" or the channel variable rtp_timer_name=none on a per call basis, this is not necessarily recommended for everyone because it's a blocking read on the rtp socket which is usually undesirable in an asynchronous thing like a phone call. BTW, Conference counts as a bridge because it has 2 threads one for each direction On Wed, Nov 18, 2009 at 3:33 PM, Robert L Mathews wrote: > Anthony Minessale wrote: > > Have you tried SVN trunk? > > No, I haven't. I'll try it. (But also, if others want to try seeing if > it happens, it's trivially duplicated by calling the echo test app and > sending the freeswitch server process a "-STOP" signal, sleeping for a > second, then sending it a "-CONT" signal.) > > In any case, though, I have partially found the cause of the problem -- > at least in the echo test module in 1.0.4. It's the same problem > reported here: > > > http://www.mail-archive.com/freeswitch-users%40lists.freeswitch.org/msg15800.html > > The two suggestions there, explicitly setting > "rtp-autoflush-during-bridge" true and "rtp_timer_name=none", didn't fix > it for me. (The first is no surprise because that's the default anyway.) > > However, setting the (undocumented?) parameter "rtp-autoflush" to true > *did* fix it in the echo test (but not the conference). > > I think what's happening is that FreeSWITCH contains code to detect when > we've "fallen behind" the RTP audio. It looks like this happens in > rtp_common_read() of switch_rtp.c: if the code detects that there are > "extra" RTP packets waiting during several consecutive rtp_common_read() > calls, switch_core_timer_sync() is called to catch up. > > This code is apparently used during bridged calls when > "rtp-autoflush-during-bridge" is true (the default), which explains why > I don't have the problem during SIP-to-SIP calls. > > However, I'm guessing that echo test calls somehow aren't considered > "bridged" by that code. Therefore the code isn't being used unless > "rtp-autoflush" is true. > > That thread suggests that this is probably a phone or network problem, > but it seems to me that even if all the timing is perfect, this problem > will happen if the freeswitch server thread doesn't get enough CPU time > to retrieve a packet before the next one arrives. For example, if > packets arrive every 20 ms but high load on the server causes the > process to sleep for 25 ms, so that two packets are waiting the next > time the process runs, it will never catch up that extra packet -- the > echo test or conference will now be permanently 20 ms behind. And if > that happens again, it will now be 40 ms behind, and so on. That > explains why the lag slowly increases over time. > > Does that make sense? I don't quite understand why the "catch up" code > isn't always used for all RTP streams: if an RTP packet poll repeatedly > shows that there are extra audio packets waiting, it seems to make sense > that we'd always want to catch up. > > Anyway, as I said, I'm still having the conference problem, even with > "rtp-autoflush" enabled. So I need to try the svn trunk version and see > if it still happens, then track it down further if so. I will report back. > > -- > Robert L Mathews, Tiger Technologies > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/39ece30a/attachment-0001.html From dave at 3c.co.uk Wed Nov 18 15:39:02 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 18 Nov 2009 16:39:02 -0700 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> Message-ID: <1258587542.12820.275.camel@localhost.localdomain> Hi Tim, > > In (very) brief: maybe, no, and depends on the definition of 'lots'. > > > > By lots I mean somewhere between 50 to a 100 but it's mostly an IVR > application so all it will be doing is either playing prompts or > recording messages. Almost no live conversations. For the sort of box you're talking about (quad core++), this isn't lots; it's hardly any.. --Dave From brian at freeswitch.org Wed Nov 18 15:47:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 17:47:04 -0600 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258587542.12820.275.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> Message-ID: <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> It just doesn't belong in user space or kernel space in the machine for true performance you should do it in hardware... I'm pretty sure the poor box would die if you tried it on 32 E1's at the same time. /b On Nov 18, 2009, at 5:39 PM, David Knell wrote: > For the sort of box you're talking about (quad core++), this isn't > lots; > it's hardly any.. > > --Dave From anthony.minessale at gmail.com Wed Nov 18 15:47:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Nov 2009 17:47:51 -0600 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258587542.12820.275.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> Message-ID: <191c3a030911181547t7c92f306l4d8c1f2920b28688@mail.gmail.com> The important thing is that if you are using wanpipe native interface there is no software echo canceler. Sangoma only supports a hardware echo canceler so its not a matter of the cpu compensating for it, if you don't get the cards with the echo can you won't have any unless you run the card in zaptel mode, something we have aspired to avoid. On Wed, Nov 18, 2009 at 5:39 PM, David Knell wrote: > Hi Tim, > > > > In (very) brief: maybe, no, and depends on the definition of 'lots'. > > > > > > > By lots I mean somewhere between 50 to a 100 but it's mostly an IVR > > application so all it will be doing is either playing prompts or > > recording messages. Almost no live conversations. > > For the sort of box you're talking about (quad core++), this isn't lots; > it's hardly any.. > > --Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/aa4e8ff4/attachment.html From mayamatakeshi at gmail.com Wed Nov 18 17:09:37 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Thu, 19 Nov 2009 10:09:37 +0900 Subject: [Freeswitch-users] mod_fifo and multi-tenancy In-Reply-To: <87f2f3b90911181336w2b0f6cb8s211bc235a6f9084d@mail.gmail.com> References: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> <87f2f3b90911181336w2b0f6cb8s211bc235a6f9084d@mail.gmail.com> Message-ID: <15b9404e0911181709i1057e2co9a06c24e2db1d267@mail.gmail.com> On Thu, Nov 19, 2009 at 6:36 AM, Michael Collins wrote: > > > On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi wrote: > >> About mod_fifo, it would be safe to use it in multi-tenancy scenarios >> where domains are created and deleted all the time and in consequence, fifos >> are created all the time? I mean, fifos are eventually destroyed by mod_fifo >> itself. Is this correct? >> >> br, >> takeshi >> >> > No, FIFOs are not "destroyed" automatically just because the last member > goes away. Tony says that an empty FIFO takes up almost no memory so > performance shouldn't be an issue. You can always issue the API command: *fifo > reparse del_all* to clean everything out if you feel like things are > getting out of hand. > Thanks, I have updated the mod_fifo wiki page. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/d1e43720/attachment.html From msc at freeswitch.org Wed Nov 18 17:45:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Nov 2009 17:45:45 -0800 Subject: [Freeswitch-users] mod_fifo and multi-tenancy In-Reply-To: <15b9404e0911181709i1057e2co9a06c24e2db1d267@mail.gmail.com> References: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> <87f2f3b90911181336w2b0f6cb8s211bc235a6f9084d@mail.gmail.com> <15b9404e0911181709i1057e2co9a06c24e2db1d267@mail.gmail.com> Message-ID: <87f2f3b90911181745n15b89330wdc14f2c5e1dd4048@mail.gmail.com> On Wed, Nov 18, 2009 at 5:09 PM, mayamatakeshi wrote: > > > On Thu, Nov 19, 2009 at 6:36 AM, Michael Collins wrote: > >> >> >> On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi wrote: >> >>> About mod_fifo, it would be safe to use it in multi-tenancy scenarios >>> where domains are created and deleted all the time and in consequence, fifos >>> are created all the time? I mean, fifos are eventually destroyed by mod_fifo >>> itself. Is this correct? >>> >>> br, >>> takeshi >>> >>> >> No, FIFOs are not "destroyed" automatically just because the last member >> goes away. Tony says that an empty FIFO takes up almost no memory so >> performance shouldn't be an issue. You can always issue the API command: >> *fifo reparse del_all* to clean everything out if you feel like things >> are getting out of hand. >> > > Thanks, > I have updated the mod_fifo wiki page. > > FYI, I was doing some other research and I noticed this on the mod_fifo wiki page: *fifo_destroy_after_use*: FreeSWITCH automatically create a new FIFO when the first time use it, and keep in the memory hash. This var tell FreeSWITCH destroy it to save memory. Using for a one time FIFO. So... you do have that option as well. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/4230bb05/attachment.html From mcampbellsmith at gmail.com Wed Nov 18 18:36:52 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 19 Nov 2009 13:36:52 +1100 Subject: [Freeswitch-users] Call from Secure RTP to non-secure RTP Message-ID: <33c87fa30911181836i75ec2945gc7e5782b38c14415@mail.gmail.com> Hi! How do I setup FS so that placing a call from an extension that only support SRTP (1002) to an extension that only supports RTP (1000)? I put this dialstring, from the wiki http://wiki.freeswitch.org/wiki/Tls, into the users xml file under directory/default I have also put a when 1000 is dialing 1002. However I never see crytpo sent in the RTP to 1002 and it responds with Bad Security Level What have I missed? Thanks From jim at evolutiontel.net Wed Nov 18 21:12:35 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 19 Nov 2009 16:12:35 +1100 Subject: [Freeswitch-users] Call from Secure RTP to non-secure RTP In-Reply-To: <33c87fa30911181836i75ec2945gc7e5782b38c14415@mail.gmail.com> References: <33c87fa30911181836i75ec2945gc7e5782b38c14415@mail.gmail.com> Message-ID: Does 1002 use TLS to transport SIP signalling? My experience is that TLS is required on some phones otherwise they will not do srtp and will reply with the responce you have mentioned. Sent from my iPhone On 19/11/2009, at 1:36 PM, Mark Campbell-Smith wrote: > Hi! > > How do I setup FS so that placing a call from an extension that only > support SRTP (1002) to an extension that only supports RTP (1000)? > > I put this dialstring, from the wiki > http://wiki.freeswitch.org/wiki/Tls, into the users xml file under > directory/default > > value="{sip_secure_media=${regex(${sofia_contact(${dialed_user}@$ > {dialed_domain})}|transport=tls)}, > presence_id=${dialed_user}@${dialed_domain}}${sofia_contact($ > {dialed_user}@${dialed_domain})}" > /> > > I have also put a data="sip_secure_media=true"/> when 1000 is dialing 1002. > > > > > > > However I never see crytpo sent in the RTP to 1002 and it responds > with Bad Security Level > > What have I missed? > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From dujinfang at gmail.com Wed Nov 18 21:22:04 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 19 Nov 2009 13:22:04 +0800 Subject: [Freeswitch-users] mod_fifo and multi-tenancy In-Reply-To: <87f2f3b90911181745n15b89330wdc14f2c5e1dd4048@mail.gmail.com> References: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> <87f2f3b90911181336w2b0f6cb8s211bc235a6f9084d@mail.gmail.com> <15b9404e0911181709i1057e2co9a06c24e2db1d267@mail.gmail.com> <87f2f3b90911181745n15b89330wdc14f2c5e1dd4048@mail.gmail.com> Message-ID: <23f91030911182122h4b6c94b9hab74132425b6b006@mail.gmail.com> I once wrote a patch for "fifo delete", but didn't submit to jira. If someone think it's useful to merge into trunk, I think I can still find the code, but sure need to test with the current trunk. 2009/11/19 Michael Collins > > > On Wed, Nov 18, 2009 at 5:09 PM, mayamatakeshi wrote: > >> >> >> On Thu, Nov 19, 2009 at 6:36 AM, Michael Collins wrote: >> >>> >>> >>> On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi wrote: >>> >>>> About mod_fifo, it would be safe to use it in multi-tenancy scenarios >>>> where domains are created and deleted all the time and in consequence, fifos >>>> are created all the time? I mean, fifos are eventually destroyed by mod_fifo >>>> itself. Is this correct? >>>> >>>> br, >>>> takeshi >>>> >>>> >>> No, FIFOs are not "destroyed" automatically just because the last member >>> goes away. Tony says that an empty FIFO takes up almost no memory so >>> performance shouldn't be an issue. You can always issue the API command: >>> *fifo reparse del_all* to clean everything out if you feel like things >>> are getting out of hand. >>> >> >> Thanks, >> I have updated the mod_fifo wiki page. >> >> > FYI, I was doing some other research and I noticed this on the mod_fifo > wiki page: > > *fifo_destroy_after_use*: FreeSWITCH automatically create a new FIFO when > the first time use it, and keep in the memory hash. This var tell FreeSWITCH > destroy it to save memory. Using for a one time FIFO. > > So... you do have that option as well. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/432e1b76/attachment-0001.html From mcampbellsmith at gmail.com Wed Nov 18 21:22:32 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 19 Nov 2009 16:22:32 +1100 Subject: [Freeswitch-users] Call from Secure RTP to non-secure RTP In-Reply-To: References: <33c87fa30911181836i75ec2945gc7e5782b38c14415@mail.gmail.com> Message-ID: <33c87fa30911182122g79354754t49d16f35db1f0d26@mail.gmail.com> Thanks Jim, Yep, 1002 does TLS and SRTP, 1000 does UDP and RTP. Cheers On Thu, Nov 19, 2009 at 4:12 PM, Jim Burke wrote: > Does 1002 use TLS to transport SIP signalling? My experience is that > TLS is required on some phones otherwise they will not do srtp and > will reply with the responce you have mentioned. > > Sent from my iPhone > > On 19/11/2009, at 1:36 PM, Mark Campbell-Smith > wrote: > >> Hi! >> >> How do I setup FS so that placing a call from an extension that only >> support SRTP (1002) to an extension that only supports RTP (1000)? >> >> I put this dialstring, from the wiki >> http://wiki.freeswitch.org/wiki/Tls, into the users xml file under >> directory/default >> >> > value="{sip_secure_media=${regex(${sofia_contact(${dialed_user}@$ >> {dialed_domain})}|transport=tls)}, >> ?presence_id=${dialed_user}@${dialed_domain}}${sofia_contact($ >> {dialed_user}@${dialed_domain})}" >> /> >> >> I have also put a > data="sip_secure_media=true"/> when 1000 is dialing 1002. >> ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> >> However I never see crytpo sent in the RTP to 1002 and it responds >> with Bad Security Level >> >> What have I missed? >> >> Thanks >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Wed Nov 18 21:28:39 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 19 Nov 2009 13:28:39 +0800 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <191c3a030911181510i4c8d36brc03fa4063b088c93@mail.gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> <51686C79-3122-4B3B-AD06-F9CEF175E023@freeswitch.org> <191c3a030911181510i4c8d36brc03fa4063b088c93@mail.gmail.com> Message-ID: <23f91030911182128g11441fcasf9aff6e49e997f3@mail.gmail.com> lol: 2009/11/19 Anthony Minessale > maybe you could send them 183 then 4 180's or send them an invite and > pretend to deadlock and not send any more sip traffic as a way of > identifying yourself > > On Wed, Nov 18, 2009 at 4:05 PM, Brian West wrote: > >> Well this is a bit more informal vs the wiki where people take it as >> fact! :) Plus its a little humpday humor! >> >> /b >> >> On Nov 18, 2009, at 3:30 PM, Rob Forman wrote: >> >> > lol! >> > >> > we have to play nice in the wiki but the mailing list is another >> > story. >> > >> > >> > On Nov 18, 2009, at 3:20 PM, Brian West wrote: >> > >> >> Sounds like you need to take a baseball bat to their forehead. >> >> >> >> /b >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/b56af2c2/attachment.html From ujjval at simplesignal.com Wed Nov 18 21:34:02 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Wed, 18 Nov 2009 21:34:02 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> I have used the following setting in ivr.conf.xml to setup conferencing with moderator. However, the issue I have is - the user enters 123456 and then say if it's a moderator they enter wrong Moderator PIN 3 times then it takes the user back to the main menu..."conference_menu" and asks for main conf pin (123456) once again. I would like the caller to be disconnected if they get into the Moderator menu and enter wrong Moderator PIN 3 times. Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO? 80112 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Thursday, November 05, 2009 7:52 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hi UK, From what I've done and read, the caller-controls (in conference.conf.xml) can be modified to almost anything you can think of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a caller control just for those with the moderator flag. So unless you want everyone able to mute/kick everyone then you can't do it. The wiki seems to indicate this as well: "Be aware that the caller-controls are applied across the entire conference. You cannot enter one member of the conference using caller- controls ABC and then enter a second member using caller-controls XYZ." http://wiki.freeswitch.org/wiki/Mod_conference I think this might be a limitation of mod_conference. Perhaps one of the pros can chime in if I'm off-base or there's some nifty way to accomplish this. Cheers, Rob On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > Any ideas on the below...has anyone implemented the below: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Ujjval Karihaloo > Sent: Monday, November 02, 2009 12:52 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Rob: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 9:34 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hm, strange. I haven't seen that before. Can you pastebin your logs > at debug level? > > On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > >> It's strange... a tcpdump tells me that there is no DTMF from my >> provider when using IVR, but when I call into a TN that goes >> directly into the Conference App, I see DTMF from the provider. >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 7:23 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I've never had any problem with that. Is your logging at debug level >> so you can see the RECV DTFM in the log/fs_cli? Are you calling from >> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >> isn't passing them through. >> >> Make sure your logging is turned up then try something simpler, like >> calling the echo application, and see if DTFM comes through. >> >> Rob >> >> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >> >>> Rob: >>> >>> For some reason, I don't see the DTMF appear on the fs_CLI when >>> using the below configuration....so it basically timesout. >>> >>> UK >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, October 26, 2009 9:21 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Thx a lot Rob, reading the wiki your way or using IVR seems >>> correct.. >>> =============== >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> ====================== >>> >>> I will try it out. >>> >>> Again thx a lot for your help. Will keep everyone posted. >>> >>> ________________________________________ >>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>> Sent: Friday, October 23, 2009 12:22 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I just re-tested with the pin in my dial plan: >>> >>> >>> >>> And it doesn't challenge me for the pin. I just drop right in. I >>> figured this is how it was intended, since the wiki says the pin is >>> set initially and only challenged in later attempts [by future >>> callers]: >>> >>> "The first time a conference name (confname) is used, it will be >>> created on demand, and the pin will be set to what ever is specified >>> at that time: the pin in the data string if specified, or if not, >>> the >>> "pin" setting in the conference profile, and if that is also >>> unspecified, then there is no pin protection. Any later attempt to >>> join the conference must specify the same pin number, if one existed >>> when it was created. " >>> >>> >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> >>> I'm sure there's a scenario where its used and useful, the wiki just >>> doesn't explain it. >>> >>> Rob >>> >>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>> >>>> Well first off you're not defining a pine here... >>>> >>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>> pin >>>> number] >>>> >>>> That might be why its not asking for a pin. >>>> >>>> /b >>>> >>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From rob4manhere at gmail.com Wed Nov 18 22:02:03 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 19 Nov 2009 00:02:03 -0600 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> Message-ID: <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> Hi again UK, IVR is designed to naturally return to previous or top menus. I don't think there's a way to change this default behavior. Maybe its time to move to a script-based pin validation system so you have the full control you need. http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR Rob On Nov 18, 2009, at 11:34 PM, Ujjval Karihaloo wrote: > I have used the following setting in ivr.conf.xml to setup > conferencing with moderator. > > However, the issue I have is - the user enters 123456 and then say > if it's a moderator they enter wrong Moderator PIN 3 times then it > takes the user back to the main menu..."conference_menu" and asks > for main conf pin (123456) once again. > > I would like the caller to be disconnected if they get into the > Moderator menu and enter wrong Moderator PIN 3 times. > > greet-long="welcome_please_enter_conference_pin.wav" > greet-short="check_and_try_again.wav" > invalid-sound="passcode_invalid.wav" > exit-sound="voicemail/vm-goodbye.wav" > timeout="10000" > inter-digit-timeout="5000" > max-failures="3" > max-timeouts="3" > digit-len="7"> > param="conference_123456_moderator_menu" /> > > > greet- > long > = > "conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant > .wav" > greet-short="check_moderator_pin_or_1_to_join.wav" > invalid-sound="invalid_moderator_pin.wav" > exit-sound="voicemail/vm-goodbye.wav" > timeout="10000" > inter-digit-timeout="5000" > max-failures="3" > max-timeouts="3" > digit-len="5"> > > > > > > > > > Ujjval Karihaloo > VP Voice Engineering > IP Phone: +13032428610 > E-Fax: +17202391690 > > SimpleSignal Inc. > 88 Inverness Circle East > Suite K105 > Englewood, CO 80112 > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Thursday, November 05, 2009 7:52 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hi UK, > > From what I've done and read, the caller-controls (in > conference.conf.xml) can be modified to almost anything you can think > of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a > caller control just for those with the moderator flag. So unless you > want everyone able to mute/kick everyone then you can't do it. > > The wiki seems to indicate this as well: > > "Be aware that the caller-controls are applied across the entire > conference. You cannot enter one member of the conference using > caller- > controls ABC and then enter a second member using caller-controls > XYZ." > > http://wiki.freeswitch.org/wiki/Mod_conference > > > I think this might be a limitation of mod_conference. Perhaps one of > the pros can chime in if I'm off-base or there's some nifty way to > accomplish this. > > Cheers, > Rob > > On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > >> Any ideas on the below...has anyone implemented the below: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, November 02, 2009 12:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Rob: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 9:34 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hm, strange. I haven't seen that before. Can you pastebin your logs >> at debug level? >> >> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >> >>> It's strange... a tcpdump tells me that there is no DTMF from my >>> provider when using IVR, but when I call into a TN that goes >>> directly into the Conference App, I see DTMF from the provider. >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 7:23 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I've never had any problem with that. Is your logging at debug >>> level >>> so you can see the RECV DTFM in the log/fs_cli? Are you calling >>> from >>> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >>> isn't passing them through. >>> >>> Make sure your logging is turned up then try something simpler, like >>> calling the echo application, and see if DTFM comes through. >>> >>> Rob >>> >>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>> >>>> Rob: >>>> >>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>> using the below configuration....so it basically timesout. >>>> >>>> UK >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Ujjval Karihaloo >>>> Sent: Monday, October 26, 2009 9:21 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>> Moderator >>>> >>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>> correct.. >>>> =============== >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> ====================== >>>> >>>> I will try it out. >>>> >>>> Again thx a lot for your help. Will keep everyone posted. >>>> >>>> ________________________________________ >>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>> Sent: Friday, October 23, 2009 12:22 PM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>> Moderator >>>> >>>> I just re-tested with the pin in my dial plan: >>>> >>>> >>>> >>>> And it doesn't challenge me for the pin. I just drop right in. I >>>> figured this is how it was intended, since the wiki says the pin is >>>> set initially and only challenged in later attempts [by future >>>> callers]: >>>> >>>> "The first time a conference name (confname) is used, it will be >>>> created on demand, and the pin will be set to what ever is >>>> specified >>>> at that time: the pin in the data string if specified, or if not, >>>> the >>>> "pin" setting in the conference profile, and if that is also >>>> unspecified, then there is no pin protection. Any later attempt to >>>> join the conference must specify the same pin number, if one >>>> existed >>>> when it was created. " >>>> >>>> >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> >>>> I'm sure there's a scenario where its used and useful, the wiki >>>> just >>>> doesn't explain it. >>>> >>>> Rob >>>> >>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>> >>>>> Well first off you're not defining a pine here... >>>>> >>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>> pin >>>>> number] >>>>> >>>>> That might be why its not asking for a pin. >>>>> >>>>> /b >>>>> >>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From elihayun at gmail.com Wed Nov 18 22:36:22 2009 From: elihayun at gmail.com (Eli Hayun) Date: Thu, 19 Nov 2009 08:36:22 +0200 Subject: [Freeswitch-users] change event value Message-ID: <4B04E766.8070706@savion.huji.ac.il> Hi Is there is a way to intercept an event (for example : REGISTER) and change one of its parameters (for example: the extension number) and fire up the corrected event? I need it to set the speedial of the phone value to be "**xxxxx" but to make it register as "xxxxx" Thanks Eli From lists at tigertech.com Wed Nov 18 23:09:22 2009 From: lists at tigertech.com (Robert L Mathews) Date: Wed, 18 Nov 2009 23:09:22 -0800 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <191c3a030911181528j7a38ce32gb2fc6fdd585932a9@mail.gmail.com> References: <4B032142.1000308@tigertech.com> <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> <4B04682A.6000309@tigertech.com> <191c3a030911181528j7a38ce32gb2fc6fdd585932a9@mail.gmail.com> Message-ID: <4B04EF22.1030404@tigertech.com> Anthony Minessale wrote: > I can promise you that much of your problems will be solved with > latest SVN. Okay, thanks! And in fact, I tried today's SVN, and it has fixed the problem with the conference, even without setting "rtp-autoflush". Conferences now discard packets and "catch up" when they gets behind, even with only the default "rtp-autoflush-during-bridge" set. The echo test still suffers from the same problem unless "rtp-autoflush" is used, which I assume is simply because it's not considered a bridged call. Eavesdropping on an existing bridged call, then pressing "3" to turn it into a conference call, also requires "rtp-autoflush" to avoid persistent lag on the third leg. > Did you answer the question about what phones? I'm going to guess Cisco > based on the symptoms. It happens with all phones, as far as I can tell. I've tried at least Grandstream GXP2000, Grandstream BT102, SJPhone, Twinkle, and Express Talk (none of them Cisco). I'm fairly positive the problem is unrelated to phones; it's caused by delays in CPU scheduling of the server process. > non bridge calls use a timer to make sure the audio is coming in at a > steady rate to ensure bursting RTP > is played at the correct rate. Stopping it and restarting 2 seconds > later will cause a delay by design because you have suspended the > process but not the UDP stack. Ummm.... well, a copy of FreeSWITCH running on any non-realtime operating system will occasionally not get scheduled for all the CPU time it wants. For example, it wouldn't be unusual for a thread to ask to sleep for 20 milliseconds but actually not wake up for 21, 25, or even 40 milliseconds because the server is busy with other things. Each time that happens, it's a smaller version of my artificial suspend test: the operating system has, of course "suspended the process but not the UDP stack". It doesn't necessarily mean there's bursty network traffic or phone timing issues. Should FreeSWITCH really lag by that much for the rest of the call? 20 milliseconds here, 20 milliseconds there, and pretty soon you're talking about real seconds. I'm assuming the reason for making it catch up on bridged calls, but not non-bridged calls, is that people talking to each other can't tolerate high latency, but people listening to an IVR or something can. But is that still true if it gets seconds behind? And should the third leg of an eavesdrop-converted-to-three-way-call be considered non-bridged for this purpose? Anyway, given that current svn trunk fixes the problem by default in conferences and any other bridged call, I'm satisfied. And if anyone complains about this problem for non-bridged calls, I suppose they can enable "rtp-autoflush" to get the same "catch-up" behavior there. I took a shot at documenting these parameters in the wiki on: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#rtp-autoflush-during-bridge Thanks for the help! -- Robert L Mathews, Tiger Technologies From helmut.kuper at ewetel.de Thu Nov 19 00:33:03 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 19 Nov 2009 09:33:03 +0100 Subject: [Freeswitch-users] Question about odbc support In-Reply-To: <4B043B32.20802@ewetel.de> References: <4B043B32.20802@ewetel.de> Message-ID: <4B0502BF.6050800@ewetel.de> Hello, hm kind of unclear Question. So I'm looking for a way to get the affected number of rows after executing a delete statement via ODBC. There is a function called "SQLRowCount()", but I didn't found a switch_odbc_* function in FS which allows me to call it. On 18.11.2009 19:21, Helmut Kuper wrote: > Hi, > > > does anybody know how to check the affected rows caused by delete, > insert or update sql statements in FS? > > To do this with sqlite3 there is a function called switch_core_db_changes(). > > > regards > helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Mit freundlichen Gr??en Helmut Kuper Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From helmut.kuper at ewetel.de Thu Nov 19 00:33:11 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 19 Nov 2009 09:33:11 +0100 Subject: [Freeswitch-users] Question about odbc support In-Reply-To: <4B043B32.20802@ewetel.de> References: <4B043B32.20802@ewetel.de> Message-ID: <4B0502C7.1060105@ewetel.de> Hello, hm kind of unclear Question. So I'm looking for a way to get the affected number of rows after executing a delete statement via ODBC. There is a function called "SQLRowCount()", but I didn't found a switch_odbc_* function in FS which allows me to call it. On 18.11.2009 19:21, Helmut Kuper wrote: > Hi, > > > does anybody know how to check the affected rows caused by delete, > insert or update sql statements in FS? > > To do this with sqlite3 there is a function called switch_core_db_changes(). > > > regards > helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From woodydickson at gmail.com Thu Nov 19 00:40:58 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Thu, 19 Nov 2009 16:40:58 +0800 Subject: [Freeswitch-users] store registration info in memcache Message-ID: Hi, Is there anyway to store registration info in memcache instead of sqlite? By doing that, it is possible for multiple freeswitch to share the same user registration info. Is there anyway I can intercept the registration success/failure event and write stuff to memcache myself? thanks, woody -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/623d720c/attachment.html From samuelmukoti at gmail.com Thu Nov 19 00:48:03 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Thu, 19 Nov 2009 10:48:03 +0200 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> Message-ID: <2A0CB328-8C3A-4CD0-B2EC-D6952E7539C0@gmail.com> Greetings, I'm a new freeswitch user and am wondering what people do when setting options in the freeswitch config files. Do people use special tools, XML editors etc or is it just vi/emacs/Kate? I'm a developer and was thinking of putting together a small editor to manage my freeswitch server, something like freepbx, I know it's not an easy undertaking but I'm sure it's well worth it. Regards, Samuel Mukoti CEO Melivo Business Systems Mobile: +263912739405 Email: sam at melivo.com Skype: samuelmukoti Twitter: twitter.com/samuelmukoti On 19 Nov,2009, at 8:02 AM, Rob Forman wrote: > Hi again UK, > > IVR is designed to naturally return to previous or top menus. I don't > think there's a way to change this default behavior. Maybe its time > to move to a script-based pin validation system so you have the full > control you need. > > http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR > > Rob > > On Nov 18, 2009, at 11:34 PM, Ujjval Karihaloo wrote: > >> I have used the following setting in ivr.conf.xml to setup >> conferencing with moderator. >> >> However, the issue I have is - the user enters 123456 and then say >> if it's a moderator they enter wrong Moderator PIN 3 times then it >> takes the user back to the main menu..."conference_menu" and asks >> for main conf pin (123456) once again. >> >> I would like the caller to be disconnected if they get into the >> Moderator menu and enter wrong Moderator PIN 3 times. >> >> > greet-long="welcome_please_enter_conference_pin.wav" >> greet-short="check_and_try_again.wav" >> invalid-sound="passcode_invalid.wav" >> exit-sound="voicemail/vm-goodbye.wav" >> timeout="10000" >> inter-digit-timeout="5000" >> max-failures="3" >> max-timeouts="3" >> digit-len="7"> >> > param="conference_123456_moderator_menu" /> >> >> >> > greet- >> long >> = >> "conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant >> .wav" >> greet-short="check_moderator_pin_or_1_to_join.wav" >> invalid-sound="invalid_moderator_pin.wav" >> exit-sound="voicemail/vm-goodbye.wav" >> timeout="10000" >> inter-digit-timeout="5000" >> max-failures="3" >> max-timeouts="3" >> digit-len="5"> >> >> >> >> >> >> >> >> >> Ujjval Karihaloo >> VP Voice Engineering >> IP Phone: +13032428610 >> E-Fax: +17202391690 >> >> SimpleSignal Inc. >> 88 Inverness Circle East >> Suite K105 >> Englewood, CO 80112 >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Thursday, November 05, 2009 7:52 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hi UK, >> >> From what I've done and read, the caller-controls (in >> conference.conf.xml) can be modified to almost anything you can think >> of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a >> caller control just for those with the moderator flag. So unless you >> want everyone able to mute/kick everyone then you can't do it. >> >> The wiki seems to indicate this as well: >> >> "Be aware that the caller-controls are applied across the entire >> conference. You cannot enter one member of the conference using >> caller- >> controls ABC and then enter a second member using caller-controls >> XYZ." >> >> http://wiki.freeswitch.org/wiki/Mod_conference >> >> >> I think this might be a limitation of mod_conference. Perhaps one of >> the pros can chime in if I'm off-base or there's some nifty way to >> accomplish this. >> >> Cheers, >> Rob >> >> On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: >> >>> Any ideas on the below...has anyone implemented the below: >>> >>> Once I have the Moderator and Participants logged on, how do I >>> invoke the moderator previlidges, LIk esay muting everyone/someone >>> or kicking someone out of the Conf and the like? >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, November 02, 2009 12:52 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Rob: >>> >>> Once I have the Moderator and Participants logged on, how do I >>> invoke the moderator previlidges, LIk esay muting everyone/someone >>> or kicking someone out of the Conf and the like? >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 9:34 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Hm, strange. I haven't seen that before. Can you pastebin your >>> logs >>> at debug level? >>> >>> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >>> >>>> It's strange... a tcpdump tells me that there is no DTMF from my >>>> provider when using IVR, but when I call into a TN that goes >>>> directly into the Conference App, I see DTMF from the provider. >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman >>>> Sent: Friday, October 30, 2009 7:23 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>> Moderator >>>> >>>> I've never had any problem with that. Is your logging at debug >>>> level >>>> so you can see the RECV DTFM in the log/fs_cli? Are you calling >>>> from >>>> a SIP phone on the pbx, or via a PSTN provider? Maybe your >>>> provider >>>> isn't passing them through. >>>> >>>> Make sure your logging is turned up then try something simpler, >>>> like >>>> calling the echo application, and see if DTFM comes through. >>>> >>>> Rob >>>> >>>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>>> >>>>> Rob: >>>>> >>>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>>> using the below configuration....so it basically timesout. >>>>> >>>>> UK >>>>> >>>>> >>>>> >>>>> -----Original Message----- >>>>> From: freeswitch-users-bounces at lists.freeswitch.org >>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>>> ] On Behalf Of Ujjval Karihaloo >>>>> Sent: Monday, October 26, 2009 9:21 AM >>>>> To: freeswitch-users at lists.freeswitch.org >>>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>>> Moderator >>>>> >>>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>>> correct.. >>>>> =============== >>>>> The wiki also says that the wait-mod might be "used in >>>>> conjunction >>>>> with an IVR where the moderators are authenticated with an extra >>>>> pass- >>>>> code", which is what I did. I guess that's why I didn't >>>>> understand >>>>> the point of the +pin. >>>>> ====================== >>>>> >>>>> I will try it out. >>>>> >>>>> Again thx a lot for your help. Will keep everyone posted. >>>>> >>>>> ________________________________________ >>>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch- >>>>> users-bounces at lists.freeswitch.org >>>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>>> Sent: Friday, October 23, 2009 12:22 PM >>>>> To: freeswitch-users at lists.freeswitch.org >>>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>>> Moderator >>>>> >>>>> I just re-tested with the pin in my dial plan: >>>>> >>>>> >>>>> >>>>> And it doesn't challenge me for the pin. I just drop right in. I >>>>> figured this is how it was intended, since the wiki says the pin >>>>> is >>>>> set initially and only challenged in later attempts [by future >>>>> callers]: >>>>> >>>>> "The first time a conference name (confname) is used, it will be >>>>> created on demand, and the pin will be set to what ever is >>>>> specified >>>>> at that time: the pin in the data string if specified, or if not, >>>>> the >>>>> "pin" setting in the conference profile, and if that is also >>>>> unspecified, then there is no pin protection. Any later attempt to >>>>> join the conference must specify the same pin number, if one >>>>> existed >>>>> when it was created. " >>>>> >>>>> >>>>> The wiki also says that the wait-mod might be "used in >>>>> conjunction >>>>> with an IVR where the moderators are authenticated with an extra >>>>> pass- >>>>> code", which is what I did. I guess that's why I didn't >>>>> understand >>>>> the point of the +pin. >>>>> >>>>> I'm sure there's a scenario where its used and useful, the wiki >>>>> just >>>>> doesn't explain it. >>>>> >>>>> Rob >>>>> >>>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>>> >>>>>> Well first off you're not defining a pine here... >>>>>> >>>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>>> pin >>>>>> number] >>>>>> >>>>>> That might be why its not asking for a pin. >>>>>> >>>>>> /b >>>>>> >>>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>>> freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From jason at jasonjgw.net Thu Nov 19 01:07:09 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 19 Nov 2009 20:07:09 +1100 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <2A0CB328-8C3A-4CD0-B2EC-D6952E7539C0@gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> <2A0CB328-8C3A-4CD0-B2EC-D6952E7539C0@gmail.com> Message-ID: <20091119090709.GA26604@jdc.jasonjgw.net> Samuel Mukoti wrote: > I'm a new freeswitch user and am wondering what people do when setting > options in the freeswitch config files. Do people use special tools, > XML editors etc or is it just vi/emacs/Kate? Emacs has an XML editing mode; Vim may have extensions for handling XML as well. However, I have not found it necessary to invoke the XML features of an editor; just treating the configuration files as plain text is sufficient. From leon at scarlet-internet.nl Thu Nov 19 01:14:22 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 19 Nov 2009 10:14:22 +0100 Subject: [Freeswitch-users] store registration info in memcache In-Reply-To: References: Message-ID: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> Hi, Not that I know of, but you can use odbc to store registrations and share it that way.. regards, Leon On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: > Hi, > > Is there anyway to store registration info in memcache instead of > sqlite? > > By doing that, it is possible for multiple freeswitch to share the > same user registration info. > > Is there anyway I can intercept the registration success/failure > event and write stuff to memcache myself? > > thanks, > woody > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jaybinks at gmail.com Thu Nov 19 01:33:12 2009 From: jaybinks at gmail.com (jay binks) Date: Thu, 19 Nov 2009 19:33:12 +1000 Subject: [Freeswitch-users] store registration info in memcache In-Reply-To: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> References: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> Message-ID: I believe OBDC is the official way.. however id love look at doing this in a higher performance way, without the single point of failure.. local memcache, in front of OBDC or something ?? not 100% sure of it, but just using a single central database is a little bit of a concern in a carrier environment. Jay On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij wrote: > Hi, > > Not that I know of, but you can use odbc to store registrations and > share it that way.. > > regards, > > Leon > > On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: > > > Hi, > > > > Is there anyway to store registration info in memcache instead of > > sqlite? > > > > By doing that, it is possible for multiple freeswitch to share the > > same user registration info. > > > > Is there anyway I can intercept the registration success/failure > > event and write stuff to memcache myself? > > > > thanks, > > woody > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/2852b7b7/attachment.html From leon at scarlet-internet.nl Thu Nov 19 02:07:30 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 19 Nov 2009 11:07:30 +0100 Subject: [Freeswitch-users] store registration info in memcache In-Reply-To: References: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> Message-ID: <72D493D6-F194-495A-8028-41362870305C@scarlet-internet.nl> Well, you can of course easily have a loadbalancer with failover in front of your sql servers and have them replicate to each other. Freeswitch will reconnect if a connection goes down. Perhaps failover is also possible directly through odbc ? Does anyone know if that's possible ? regards, Leon On Nov 19, 2009, at 10:33 AM, jay binks wrote: > I believe OBDC is the official way.. > however id love look at doing this in a higher performance way, > without the single point of failure.. > > local memcache, in front of OBDC or something ?? > > not 100% sure of it, but just using a single central database is a > little bit of a concern in a carrier environment. > > Jay > > > > On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij > wrote: > Hi, > > Not that I know of, but you can use odbc to store registrations and > share it that way.. > > regards, > > Leon > > On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: > > > Hi, > > > > Is there anyway to store registration info in memcache instead of > > sqlite? > > > > By doing that, it is possible for multiple freeswitch to share the > > same user registration info. > > > > Is there anyway I can intercept the registration success/failure > > event and write stuff to memcache myself? > > > > thanks, > > woody > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Sincerely > > Jay > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/55f40424/attachment.html From lon at kickasspixels.com Thu Nov 19 02:23:08 2009 From: lon at kickasspixels.com (Lon Baker) Date: Thu, 19 Nov 2009 02:23:08 -0800 Subject: [Freeswitch-users] store registration info in memcache In-Reply-To: <72D493D6-F194-495A-8028-41362870305C@scarlet-internet.nl> References: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> <72D493D6-F194-495A-8028-41362870305C@scarlet-internet.nl> Message-ID: <5d3e0dc60911190223j8a32cf0m4d7ed2983a47987a@mail.gmail.com> If we could access mod_memcache for registration information that would be ideal and highly robust, since you can share memcache with external applications. Lon On Thu, Nov 19, 2009 at 2:07 AM, Leon de Rooij wrote: > Well, you can of course easily have a loadbalancer with failover in front of > your sql servers and have them replicate to each other.?Freeswitch will > reconnect if a connection goes down. Perhaps failover is also possible > directly through odbc ? Does anyone know if that's possible ? > regards, > Leon > > > On Nov 19, 2009, at 10:33 AM, jay binks wrote: > > I believe OBDC is the official way.. > however id love look at doing this in a higher performance way, without the > single point of failure.. > local memcache, in front of OBDC or something ?? > not 100% sure of it, but just using a single central database is a little > bit of a concern in a carrier?environment. > Jay > > > On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij > wrote: >> >> Hi, >> >> Not that I know of, but you can use odbc to store registrations and >> share it that way.. >> >> regards, >> >> Leon >> >> On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: >> >> > Hi, >> > >> > Is there anyway to store registration info in memcache instead of >> > sqlite? >> > >> > By doing that, it is possible for multiple freeswitch to share the >> > same user registration info. >> > >> > Is there anyway I can intercept the registration success/failure >> > event and write stuff to memcache myself? >> > >> > thanks, >> > woody >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Sincerely > > Jay > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From stevendt at primrosebank.net Thu Nov 19 03:33:46 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 11:33:46 -0000 Subject: [Freeswitch-users] Extension Configuration - XML File Entries for Group configuration Message-ID: <73DD76AE07884A1D9535EF27C5841DAD@bp1.ad.bp.com> Hi, Can someone please help me understand a little more about Group configuration ? I believe that Group Membership is configured in the \conf\directory\default.xml file I've done this and the caller groups seem to work fine. However, each extension in the \conf\directory\default directory, e.g., 111.xml also has an entry for "callgroup" Can someone explain what the difference in these two options is please ? regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/0e1d4b31/attachment.html From rupa at rupa.com Thu Nov 19 04:52:40 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 19 Nov 2009 06:52:40 -0600 Subject: [Freeswitch-users] store registration info in memcache In-Reply-To: <5d3e0dc60911190223j8a32cf0m4d7ed2983a47987a@mail.gmail.com> References: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> <72D493D6-F194-495A-8028-41362870305C@scarlet-internet.nl> <5d3e0dc60911190223j8a32cf0m4d7ed2983a47987a@mail.gmail.com> Message-ID: I'd have to double check all the sql used for registration, but I doubt memcache is expressive enough to act as the registration store. For instance, you can't get a list of registrations from it (sofia status profile internal). memcache is a keystore only. That being said, one could use memcache as a umm.. well cache like it is designed as a front end to the real odbc database. Consult memcache first then hit the db. Doing anything like that would require moving much of mod_memcache up into core, something I promised I would do at one point but never got around to doing -- lack of time and motivation and no strong use case IMO. On Thu, Nov 19, 2009 at 4:23 AM, Lon Baker wrote: > If we could access mod_memcache for registration information that > would be ideal and highly robust, since you can share memcache with > external applications. > > Lon > > On Thu, Nov 19, 2009 at 2:07 AM, Leon de Rooij wrote: >> Well, you can of course easily have a loadbalancer with failover in front of >> your sql servers and have them replicate to each other.?Freeswitch will >> reconnect if a connection goes down. Perhaps failover is also possible >> directly through odbc ? Does anyone know if that's possible ? >> regards, >> Leon >> >> >> On Nov 19, 2009, at 10:33 AM, jay binks wrote: >> >> I believe OBDC is the official way.. >> however id love look at doing this in a higher performance way, without the >> single point of failure.. >> local memcache, in front of OBDC or something ?? >> not 100% sure of it, but just using a single central database is a little >> bit of a concern in a carrier?environment. >> Jay >> >> >> On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij >> wrote: >>> >>> Hi, >>> >>> Not that I know of, but you can use odbc to store registrations and >>> share it that way.. >>> >>> regards, >>> >>> Leon >>> >>> On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: >>> >>> > Hi, >>> > >>> > Is there anyway to store registration info in memcache instead of >>> > sqlite? >>> > >>> > By doing that, it is possible for multiple freeswitch to share the >>> > same user registration info. >>> > >>> > Is there anyway I can intercept the registration success/failure >>> > event and write stuff to memcache myself? >>> > >>> > thanks, >>> > woody >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Sincerely >> >> Jay >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From rupa at rupa.com Thu Nov 19 05:03:51 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 19 Nov 2009 07:03:51 -0600 Subject: [Freeswitch-users] How to implement mod_lcr + mod_limit In-Reply-To: References: Message-ID: Using lcr_auto_route + limit isn't really possible at this point. It is on the list of things to do but is more complex than it seems on it's surface. mod_lcr just constructs dial strings, it doesn't do any call control. It does provide enough information to do what you want via a scripting language like lua. mod_lcr sets channel vars lcr_route_count which tells you how many routes there are. It also sets lcr_route_N (where N is 1 to lcr_route_count) which contains each lcr route. You can then iterate over the routes, set limit try to bridge and loop until success. Arguably this should be done from within FS so that you could just use lcr_auto_route (assuming mod_lcr can pull limit info from the routes db). That is "the plan" but a workable solution hasn't magically appeared yet. On Mon, Nov 16, 2009 at 1:29 AM, Ahmed Munir wrote: > Hi, > > I've worked on setup for carriers routing using mod_lcr + mod_nibble + > mod_xml_curl and mod_xml_cdr. The setup is working fine as I desired. Now I > want to include mod_limit in to my setup. > > As I read the wiki pages of mod_limit I want to know how can I limit the > calls per destination basis while running mod_lcr? Because LCR is routing to > different carriers, how can I call mod_limit in mod_lcr? > > Kindly advise this issue soon. > > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From samuelmukoti at gmail.com Thu Nov 19 05:41:16 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Thu, 19 Nov 2009 15:41:16 +0200 Subject: [Freeswitch-users] XML config file parsing Message-ID: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> Thx Jason for the reply, I realise i was quite unclear in what i'm hoping to achieve. I wanted to make a control panel for our office so that we can provision extensions at the same time as we do users. We have a system much like the "ubuntu ebox" that allows use to manage users for our organization and for virtual domains - it uses postgresql as a backend. I'm not aware of freeswitch's abilities or features when it comes to databases. Can freeswitch lookup SQL tables in realtime? I would love the ability to manage dialplans, voicemail accounts, and extensions/endpoints thru a database much like mysql or postgresql The reason i was discussing XML is for this very same purpose, i though i could write helper scripts that would 'spit' out some XML configuration files thus dynamically updating Freeswitch configuration from a web frontend.. almost similar to what the freepbx.org guys have done. regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/50eaa216/attachment.html From rob4manhere at gmail.com Thu Nov 19 06:03:45 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 19 Nov 2009 08:03:45 -0600 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> References: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> Message-ID: <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> Hi Sam, Take a look at mod_xml_curl. Pretty sure it'll do everything you're looking for. http://wiki.freeswitch.org/wiki/Mod_xml_curl Also, I would browse the modules and look for other nifty functionality that already exists before setting out to write something new. http://wiki.freeswitch.org/wiki/Modules Good luck! Rob On Nov 19, 2009, at 7:41 AM, Samuel Mukoti wrote: > Thx Jason for the reply, > > I realise i was quite unclear in what i'm hoping to achieve. I > wanted to make a control panel for our office so that we can > provision extensions at the same time as we do users. We have a > system much like the "ubuntu ebox" that allows use to manage users > for our organization and for virtual domains - it uses postgresql as > a backend. > > I'm not aware of freeswitch's abilities or features when it comes to > databases. Can freeswitch lookup SQL tables in realtime? > > I would love the ability to manage dialplans, voicemail accounts, > and extensions/endpoints thru a database much like mysql or postgresql > > The reason i was discussing XML is for this very same purpose, i > though i could write helper scripts that would 'spit' out some XML > configuration files thus dynamically updating Freeswitch > configuration from a web frontend.. almost similar to what the > freepbx.org guys have done. > > regards, > > Sam > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/001f3e54/attachment.html From john_platts at hotmail.com Wed Nov 18 18:54:43 2009 From: john_platts at hotmail.com (John Platts) Date: Wed, 18 Nov 2009 20:54:43 -0600 Subject: [Freeswitch-users] Need help configuring our FreeSWITCH instance Message-ID: I have installed FreeSWITCH on our server, and need some help configuring our FreeSWITCH instance. All of the numbers associated with our FreeSWITCH instance are in the format: 1NPANXXXXXX (where NPA is the area code, and NXXXXXX are the last 7 digits of the phone number). I need the following configuration: Calls coming from our IP to IP gateway into our FreeSWITCH instance needs to be routed to the endpoint that is registered with FreeSWITCHCalls coming from any of the registered SIP endpoints need to be sent to the appropriate destination. The appropriate destination for any number that is not registered with FreeSWITCH is our IP to IP gateway.Our IP to IP gateway does not require any SIP registration or authentication.G.729 (but not G.729 Annex B), G.711 mu-law, and G.711 A-law need to be enabledSIP registrar enabled for registering endpoints other than our IP-IP gatewaySIP traffic needs to be accepted to and from both the IP-IP gateway and from the registered SIP endpoints. How do I get the above configured in FreeSWITCH? _________________________________________________________________ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/d3082fed/attachment.html From dave at 3c.co.uk Thu Nov 19 07:54:55 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 19 Nov 2009 08:54:55 -0700 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> Message-ID: <1258646095.12820.300.camel@localhost.localdomain> Hi Brian, > It just doesn't belong in user space or kernel space in the machine > for true performance you should do it in hardware... I'm pretty sure > the poor box would die if you tried it on 32 E1's at the same time. Disagree somewhat. The challenge that echo cancellers further from the hardware face is having some idea of the size of the buffers between the canceller and the wire; provided that this is known, or is small in comparison to the canceller's tail length, it can, in principle, go anywhere. All other things being equal, the right place for a software EC is in user space: can be done in a cross-platform way, can use FPU/MMX/SSE without guilt and voodoo, etc. And there is no reason why the same algorithm would perform differently if implemented in "hardware" or on the host CPU. And the OP only needed four E1s.. --Dave > > /b > > On Nov 18, 2009, at 5:39 PM, David Knell wrote: > > > For the sort of box you're talking about (quad core++), this isn't > > lots; > > it's hardly any.. > > > > --Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch-users-list at metik.com Thu Nov 19 08:01:11 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 19 Nov 2009 11:01:11 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <1258511897776-4023012.post@n2.nabble.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> <1258511897776-4023012.post@n2.nabble.com> Message-ID: <4B056BC7.6030009@metik.com> If you are using Windows XP (or Vista for that matter), you may want to look at tftpd32. Its more compact and uses less memory than Solarwinds yet provides not only a tftp server but a dhcp and syslog server as well. In the past, I've use it to upgrade, install, and troubleshoot a variety of gear (dslams, routers, softswitches, SIP endpoints, etc.) when a dedicated server was not available. -metik Jeff Lenk wrote: > Hi > > I run the SolarWinds TFTP server alongside FS on my small installation - > works nicely! > > Jeff > > > > Dave Stevenson wrote: > >> Hi, >> >> I have just about got FreeSwitch working with a Cisco 7940 Phone. After >> much reading, I worked out that I needed a TFTP server on the network that >> would supply the phone with it's SIP personality and config etc. I have >> been able to get the phone working and realise that the TFTP server needs >> to be available every time the phone loses power etc. At the moment, I >> have the TFTP server running on a temporary machine but it would be neater >> if it ran on the same machine as FreeSwitch. This will be a very small >> FreeSwitch installation, so, ....... >> >> Is there any reason why I should not try to run FreeSwitch and the >> SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't think >> the server should put much load on the machine but wondered if there were >> any other reasons why this is a bad idea ? >> >> In addition, while I have the phone working - I get a status message on >> boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf >> >> Can anyone tell me how to locate the errors in this file please ? (I have >> posted it to the Pastebin) >> >> Regards >> Dave >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > From anthony.minessale at gmail.com Thu Nov 19 08:11:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Nov 2009 10:11:56 -0600 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <4B04EF22.1030404@tigertech.com> References: <4B032142.1000308@tigertech.com> <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> <4B04682A.6000309@tigertech.com> <191c3a030911181528j7a38ce32gb2fc6fdd585932a9@mail.gmail.com> <4B04EF22.1030404@tigertech.com> Message-ID: <191c3a030911190811w267162a2p35cf85bb7e62be40@mail.gmail.com> Like I said, The timer by default is designed to make sure that none of the audio is lost for situations like FAX etc. There are parameters you can configure to disable the timers that I mentioned in the last email which will cause all of the audio to be jammed into your ear like twiddlebugs if you did you sleep test and brought it back. We do not use sleep for the timers we have timer objects into the code derived from a high priority thread sending conditional broadcasts to the timer objects. There is certainly a place where this can begin to break down but it has proven to provide reliable timing to thousands of concurrent channels. The auto-flush can get false positives in jittery situations is not always the best answer. What kind of CPU are you using and what kind of hardware that you suspect you are getting delayed cpu scheduling on a small number of calls? I appreciate your theory and I am willing to investigate a little for you but you must be aware we have put more than a few hours of thought into the architecture here. There may be a bigger problem with the eavesdropping which we can have a look at today because that does not sound right. On Thu, Nov 19, 2009 at 1:09 AM, Robert L Mathews wrote: > Anthony Minessale wrote: > > > I can promise you that much of your problems will be solved with > > latest SVN. > > Okay, thanks! > > And in fact, I tried today's SVN, and it has fixed the problem with the > conference, even without setting "rtp-autoflush". Conferences now > discard packets and "catch up" when they gets behind, even with only the > default "rtp-autoflush-during-bridge" set. > > The echo test still suffers from the same problem unless "rtp-autoflush" > is used, which I assume is simply because it's not considered a bridged > call. > > Eavesdropping on an existing bridged call, then pressing "3" to turn it > into a conference call, also requires "rtp-autoflush" to avoid > persistent lag on the third leg. > > > > Did you answer the question about what phones? I'm going to guess Cisco > > based on the symptoms. > > It happens with all phones, as far as I can tell. I've tried at least > Grandstream GXP2000, Grandstream BT102, SJPhone, Twinkle, and Express > Talk (none of them Cisco). I'm fairly positive the problem is unrelated > to phones; it's caused by delays in CPU scheduling of the server process. > > > > non bridge calls use a timer to make sure the audio is coming in at a > > steady rate to ensure bursting RTP > > is played at the correct rate. Stopping it and restarting 2 seconds > > later will cause a delay by design because you have suspended the > > process but not the UDP stack. > > Ummm.... well, a copy of FreeSWITCH running on any non-realtime > operating system will occasionally not get scheduled for all the CPU > time it wants. For example, it wouldn't be unusual for a thread to ask > to sleep for 20 milliseconds but actually not wake up for 21, 25, or > even 40 milliseconds because the server is busy with other things. > > Each time that happens, it's a smaller version of my artificial suspend > test: the operating system has, of course "suspended the process but not > the UDP stack". It doesn't necessarily mean there's bursty network > traffic or phone timing issues. > > Should FreeSWITCH really lag by that much for the rest of the call? 20 > milliseconds here, 20 milliseconds there, and pretty soon you're talking > about real seconds. > > I'm assuming the reason for making it catch up on bridged calls, but not > non-bridged calls, is that people talking to each other can't tolerate > high latency, but people listening to an IVR or something can. But is > that still true if it gets seconds behind? And should the third leg of > an eavesdrop-converted-to-three-way-call be considered non-bridged for > this purpose? > > Anyway, given that current svn trunk fixes the problem by default in > conferences and any other bridged call, I'm satisfied. And if anyone > complains about this problem for non-bridged calls, I suppose they can > enable "rtp-autoflush" to get the same "catch-up" behavior there. > > I took a shot at documenting these parameters in the wiki on: > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#rtp-autoflush-during-bridge > > Thanks for the help! > > -- > Robert L Mathews, Tiger Technologies > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/2d51c0cc/attachment-0001.html From steveu at coppice.org Thu Nov 19 08:15:15 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 20 Nov 2009 00:15:15 +0800 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258646095.12820.300.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> <1258646095.12820.300.camel@localhost.localdomain> Message-ID: <4B056F13.6050106@coppice.org> On 11/19/2009 11:54 PM, David Knell wrote: > Hi Brian, > > >> It just doesn't belong in user space or kernel space in the machine >> for true performance you should do it in hardware... I'm pretty sure >> the poor box would die if you tried it on 32 E1's at the same time. >> > Disagree somewhat. The challenge that echo cancellers further from the > hardware face is having some idea of the size of the buffers between the > canceller and the wire; provided that this is known, or is small in > comparison to the canceller's tail length, it can, in principle, go > anywhere. All other things being equal, the right place for a software > EC is in user space: can be done in a cross-platform way, can use > FPU/MMX/SSE without guilt and voodoo, etc. And there is no reason why > the same algorithm would perform differently if implemented in > "hardware" or on the host CPU. > > And the OP only needed four E1s.. > The audio path between kernel and user space is not stable with any current PC based telephony system. At some point in the day the odd chunk of data is lost here and there, whether you use asterisk, callweaver, yate or FS, with dahdi or sangoma. This is the key problem for user space echo cancellation. When the path hiccups, the EC goes crazy, and howls. So far kernel space EC has been the only way to keep the path length rock solid. There is an Intel development platform which tries to do EC with OSLEC in user space. That's the only delivered system I know that tries to do this. Its very quirky. Steve From stevendt at primrosebank.net Thu Nov 19 08:46:50 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 16:46:50 -0000 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> Message-ID: Metik, thanks a lot for the tip, I will certainly look at it, particularly if it does DHCP too. At the moment, I use my ADSL Router to provide DHCP to the network but I've just discovered that you can't configure options in its DHCP server to point to the TFTP server for the phone. At the moment, I have to have the phone set to a static IP address to be able to configure the TFTP server address which is not as flexible as using DHCP. I had thought about changing over to use Windows Server DHCP services but it sounds like ttpd32 would do the trick. I just need to decide whether I want all of my machines to rely on getting their IP address from another PC - it feels like having DHCP in the router is more robust. Regards Dave ----- Original Message ----- From: "Metik" To: Sent: Thursday, November 19, 2009 4:01 PM Subject: Re: [Freeswitch-users] TFTP Server & Cisco 7540 > If you are using Windows XP (or Vista for that matter), you may want to > look at tftpd32. Its more compact and uses less memory than Solarwinds > yet provides not only a tftp server but a dhcp and syslog server as well. > > In the past, I've use it to upgrade, install, and troubleshoot a variety > of gear (dslams, routers, softswitches, SIP endpoints, etc.) when a > dedicated server was not available. > > -metik > > > Jeff Lenk wrote: >> Hi >> >> I run the SolarWinds TFTP server alongside FS on my small installation - >> works nicely! >> >> Jeff >> >> >> >> Dave Stevenson wrote: >> >>> Hi, >>> >>> I have just about got FreeSwitch working with a Cisco 7940 Phone. After >>> much reading, I worked out that I needed a TFTP server on the network >>> that >>> would supply the phone with it's SIP personality and config etc. I have >>> been able to get the phone working and realise that the TFTP server >>> needs >>> to be available every time the phone loses power etc. At the moment, I >>> have the TFTP server running on a temporary machine but it would be >>> neater >>> if it ran on the same machine as FreeSwitch. This will be a very small >>> FreeSwitch installation, so, ....... >>> >>> Is there any reason why I should not try to run FreeSwitch and the >>> SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't >>> think >>> the server should put much load on the machine but wondered if there >>> were >>> any other reasons why this is a bad idea ? >>> >>> In addition, while I have the phone working - I get a status message on >>> boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf >>> >>> Can anyone tell me how to locate the errors in this file please ? (I >>> have >>> posted it to the Pastebin) >>> >>> Regards >>> Dave >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Thu Nov 19 08:55:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 10:55:28 -0600 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> Message-ID: <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> Some Cisco phones need DHCP option 150. /b On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: > Metik, > > thanks a lot for the tip, I will certainly look at it, particularly > if it > does DHCP too. > > At the moment, I use my ADSL Router to provide DHCP to the network > but I've > just discovered that you can't configure options in its DHCP server > to point > to the TFTP server for the phone. At the moment, I have to have the > phone > set to a static IP address to be able to configure the TFTP server > address > which is not as flexible as using DHCP. I had thought about changing > over to > use Windows Server DHCP services but it sounds like ttpd32 would do > the > trick. > > I just need to decide whether I want all of my machines to rely on > getting > their IP address from another PC - it feels like having DHCP in the > router > is more robust. > > Regards > Dave From kjv at ken-ton.com Thu Nov 19 10:11:00 2009 From: kjv at ken-ton.com (Karl J. Vesterling) Date: Thu, 19 Nov 2009 13:11:00 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> Message-ID: <868A4E38-D947-4291-BBD7-4F4C9E5B239E@ken-ton.com> Yeah, roger that... Here is an excerpt from the page I did on the Cisco 7960G HowTo: http://wiki.freeswitch.org/wiki/Freeswitch_Cisco_7960G_Howto It's for Linux, but you'll get some good pointers on the TFTP option you're looking for. I haven't provisioned any 7540's... Good luck! Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Nov 19, 2009, at 11:55 AM, Brian West wrote: > Some Cisco phones need DHCP option 150. > > /b > > On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: > >> Metik, >> >> thanks a lot for the tip, I will certainly look at it, particularly >> if it >> does DHCP too. >> >> At the moment, I use my ADSL Router to provide DHCP to the network >> but I've >> just discovered that you can't configure options in its DHCP server >> to point >> to the TFTP server for the phone. At the moment, I have to have the >> phone >> set to a static IP address to be able to configure the TFTP server >> address >> which is not as flexible as using DHCP. I had thought about changing >> over to >> use Windows Server DHCP services but it sounds like ttpd32 would do >> the >> trick. >> >> I just need to decide whether I want all of my machines to rely on >> getting >> their IP address from another PC - it feels like having DHCP in the >> router >> is more robust. >> >> Regards >> Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Nov 19 10:17:48 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 12:17:48 -0600 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <868A4E38-D947-4291-BBD7-4F4C9E5B239E@ken-ton.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> <868A4E38-D947-4291-BBD7-4F4C9E5B239E@ken-ton.com> Message-ID: I don't think a 7540 exists. /b On Nov 19, 2009, at 12:11 PM, Karl J. Vesterling wrote: > I haven't provisioned any 7540's... Good luck! From stevendt at primrosebank.net Thu Nov 19 10:25:42 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 18:25:42 -0000 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com><4B056BC7.6030009@metik.com><921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> <868A4E38-D947-4291-BBD7-4F4C9E5B239E@ken-ton.com> Message-ID: Thanks Guys, I had not realised until the last couple of days that DHCP did more than just providing the IP address to the client. I have been happily just doing that for a few years now without anything other than my Router providing the DHCP function. It's only now that I have taken the plunge into IP telephony that I realise that it can do more and for Cisco phones, should provide the address of the TFTP server. My work-around at the moment is to used fixed IP addresses in the phone for it's own IP address and the TFTP server - not as neat as I would like, but it works. I will look at a better long term solution with a different DHCP server (as already mentioned earlier in this thread). Looking on the bright side, I have got the phone provisioned - though I'm still working out what all the options are, but it is working. As Brian has spotted - my reference to a 7540 was an error - I got in right in the body of the original post, but not when I edited the subject line - oooops - sorry. The phone is a 7940 ! regards Dave ----- Original Message ----- From: "Karl J. Vesterling" To: Sent: Thursday, November 19, 2009 6:11 PM Subject: Re: [Freeswitch-users] TFTP Server & Cisco 7540 > Yeah, roger that... > Here is an excerpt from the page I did on the Cisco 7960G HowTo: > > http://wiki.freeswitch.org/wiki/Freeswitch_Cisco_7960G_Howto > > It's for Linux, but you'll get some good pointers on the TFTP option > you're looking for. > I haven't provisioned any 7540's... Good luck! > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 > > On Nov 19, 2009, at 11:55 AM, Brian West wrote: > >> Some Cisco phones need DHCP option 150. >> >> /b >> >> On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: >> >>> Metik, >>> >>> thanks a lot for the tip, I will certainly look at it, particularly >>> if it >>> does DHCP too. >>> >>> At the moment, I use my ADSL Router to provide DHCP to the network >>> but I've >>> just discovered that you can't configure options in its DHCP server >>> to point >>> to the TFTP server for the phone. At the moment, I have to have the >>> phone >>> set to a static IP address to be able to configure the TFTP server >>> address >>> which is not as flexible as using DHCP. I had thought about changing >>> over to >>> use Windows Server DHCP services but it sounds like ttpd32 would do >>> the >>> trick. >>> >>> I just need to decide whether I want all of my machines to rely on >>> getting >>> their IP address from another PC - it feels like having DHCP in the >>> router >>> is more robust. >>> >>> Regards >>> Dave >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shiyanov at gmail.com Thu Nov 19 11:25:57 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 19 Nov 2009 22:25:57 +0300 Subject: [Freeswitch-users] mod dptools record problem - hangup channel with invalid file path In-Reply-To: <67d615ac0911181226y22b4fec6ndb8e622a24db101c@mail.gmail.com> References: <67d615ac0911181138m30f3064ci1a2dad6732354e35@mail.gmail.com> <994A83CB-7069-4808-9055-30B8BD3CEA75@jerris.com> <67d615ac0911181226y22b4fec6ndb8e622a24db101c@mail.gmail.com> Message-ID: I had almost the same problem- it was needed to record everything, even if the record path doesn't exist - it was requested to create the needed path. For this purpose I've used event_socket command "api system ...", precisely, api system mkdir -p path And after this command I've started recording. So, you may the same approach. On Wed, Nov 18, 2009 at 11:26 PM, William Kendi ... < william.nishio at voicetechnology.com.br> wrote: > Actually, I am integrating FreeSWITCH with a weird IVR Framework, and the > current behaviour of the "mod dptools record" application breaks some rules > of the weird IVR Framework that must be integrated with FreeSWITCH. > In order to integrate FreeSWITCH with the weird IVR Framework, the "mod > dptools record" application mustn't terminate the call when invalid file > paths are passed, and a notification of the invalid file path through the > event system of FreeSWITCH should be enough for me, like the behaviour of > the "mod dptools playback" application when invalid file paths are passed. > > Thanks in advance. > > ** > 2009/11/18 Michael Jerris > > Okay, I'll ask the obvious question. Why are you passing record invalid >> file paths and why should it not fail if you do? >> >> Mike >> >> On Nov 18, 2009, at 2:38 PM, William Kendi ... wrote: >> >> > While I was testing the "mod dptools record" application using invalid >> file paths, i noted that the "mod dptools record" application terminated the >> call. >> > I am currently looking for a way to change this behaviour. >> > Any suggestions? Can this be done? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/7e83bf30/attachment-0001.html From info at daccii.it Thu Nov 19 11:29:43 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Thu, 19 Nov 2009 20:29:43 +0100 Subject: [Freeswitch-users] Call doesn't work while registration work for a VOIP provider Message-ID: <4B059CA7.3040201@daccii.it> Hi, i'm trying to configure freeswitch with a VOIP provider, exsorsa, that uses OpenSER. Exsorsa use as own gateway, another provider, Eutelia, that it uses Cisco (or, at least, this appears in headers). Short story: ------------ If i try to setup my Eutelia account all works perfectly while if i try to setup Exsorsa account registration works fine while calling not: when fs send the ACK, as answer to a OK (sip code 200), that is sended from exsorsa as answer to an INVITE, exsorsa send back a BYE. Long story: ----------- I put call log on pastebin with debug and sip_trace enabled for external sip_profile and with log level on debug on fs console. Registration log, here all is ok (or at least it seems to be ok) http://pastebin.freeswitch.org/11176 Annoyng message that comes up every 30 seconds http://pastebin.freeswitch.org/11177 Call log http://pastebin.freeswitch.org/11178 As you can see from call log all works fine until fs send back the acknowledgment message (line 451 on last log). Can this depend on the annoyng message that comes up every 30 seconds? Here my external sip profile config http://pastebin.freeswitch.org/11180 while here exsorsa gateway config http://pastebin.freeswitch.org/11181 Any helps is really appreciated! I'm fought with it all the day!!! Best Regards, Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/712d6ef2/attachment.vcf From brian at freeswitch.org Thu Nov 19 11:38:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 13:38:53 -0600 Subject: [Freeswitch-users] Call doesn't work while registration work for a VOIP provider In-Reply-To: <4B059CA7.3040201@daccii.it> References: <4B059CA7.3040201@daccii.it> Message-ID: I'm going to guess gw+exsorsa is what they don't like. try extensions- in-contact=true on the gateway config. /b On Nov 19, 2009, at 1:29 PM, Albano Daniele Salvatore - Lavoro wrote: > Hi, > > i'm trying to configure freeswitch with a VOIP provider, exsorsa, > that uses OpenSER. Exsorsa use as own gateway, another provider, > Eutelia, that it uses Cisco (or, at least, this appears in headers). > > Short story: > ------------ > If i try to setup my Eutelia account all works perfectly while if i > try to setup Exsorsa account registration works fine while calling > not: when fs send the ACK, as answer to a OK (sip code 200), that is > sended from exsorsa as answer to an INVITE, exsorsa send back a BYE. > > > Long story: > ----------- > I put call log on pastebin with debug and sip_trace enabled for > external sip_profile and with log level on debug on fs console. > > Registration log, here all is ok (or at least it seems to be ok) > http://pastebin.freeswitch.org/11176 > > Annoyng message that comes up every 30 seconds > http://pastebin.freeswitch.org/11177 > > Call log > http://pastebin.freeswitch.org/11178 > > As you can see from call log all works fine until fs send back the > acknowledgment message (line 451 on last log). > > Can this depend on the annoyng message that comes up every 30 seconds? > > Here my external sip profile config > http://pastebin.freeswitch.org/11180 > > while here exsorsa gateway config > http://pastebin.freeswitch.org/11181 > > > Any helps is really appreciated! I'm fought with it all the day!!! > > Best Regards, > Daniele > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From shiyanov at gmail.com Thu Nov 19 11:46:14 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 19 Nov 2009 22:46:14 +0300 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app Message-ID: Hi there! I've got annoying FS behavior: There are 2 channels executing the same Java application (application itself is an IVR). If I try to bridge them with uuid_bridged then both channels are killed. Here is a log from FS console: uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 (sofia/internal/ 1005 at 192.168.147.130) State Change CS_EXECUTE -> CS_HIBERNATE 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1005 at 192.168.147.130 [BREAK] 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 (sofia/internal/ 1001 at master.agent.starpoundtech.net) State Change CS_EXECUTE -> CS_HIBERNATE 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at master.agent.starpoundtec 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1005 at 192.168.147.130 [BREAK] 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] (FS version is 1.0.4) Any thoughts? Artem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/cec920e7/attachment.html From jerry.richards at teotech.com Thu Nov 19 12:00:14 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 19 Nov 2009 12:00:14 -0800 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP Message-ID: <7C3068B7C76746E4A9ACC216574035B9@greyhawk.tonecommander.com> Hello, I just pasted a log in the Pastebin with Freeswitch logging enabled. Does anyone know a way to prevent FS from connecting the call prior to the callee answering? Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Thursday, November 05, 2009 3:50 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Want 183 w/SDP, but Get 200 w/SDP I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry From anthony.minessale at gmail.com Thu Nov 19 12:18:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Nov 2009 14:18:05 -0600 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP In-Reply-To: <7C3068B7C76746E4A9ACC216574035B9@greyhawk.tonecommander.com> References: <7C3068B7C76746E4A9ACC216574035B9@greyhawk.tonecommander.com> Message-ID: <191c3a030911191218m6fb6992eg5b4eaf338397ed0b@mail.gmail.com> set enable-3pcc to "proxy" instead of "true" On Thu, Nov 19, 2009 at 2:00 PM, Jerry Richards wrote: > > Hello, > > I just pasted a log in the Pastebin with Freeswitch logging enabled. Does > anyone know a way to prevent FS from connecting the call prior to the > callee > answering? > > Best Regards, > Jerry > > > -----Original Message----- > From: Jerry Richards [mailto:jerry.richards at teotech.com] > Sent: Thursday, November 05, 2009 3:50 PM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: Want 183 w/SDP, but Get 200 w/SDP > > > I am trying to make a call through a Gateway that sends the INVITE with no > SDP and ONLY wants the 200 OK w/SDP when the callee answers. > > For some reason, Freeswitch answers the call with 200 OK w/SDP even before > the callee answers the phone. Is this to provide ringback? Can I disable > that action? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/3f24680c/attachment.html From msc at freeswitch.org Thu Nov 19 12:25:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Nov 2009 12:25:36 -0800 Subject: [Freeswitch-users] Extension Configuration - XML File Entries for Group configuration In-Reply-To: <73DD76AE07884A1D9535EF27C5841DAD@bp1.ad.bp.com> References: <73DD76AE07884A1D9535EF27C5841DAD@bp1.ad.bp.com> Message-ID: <87f2f3b90911191225l243a5cdflf8c61b8ebc76dfcb@mail.gmail.com> On Thu, Nov 19, 2009 at 3:33 AM, Dave Stevenson wrote: > Hi, > > Can someone please help me understand a little more about Group > configuration ? > > I believe that Group Membership is configured in the > \conf\directory\default.xml file > > I've done this and the caller groups seem to work fine. > > However, each extension in the \conf\directory\default directory, e.g., > 111.xml also has an entry for "callgroup" > > Can someone explain what the difference in these two options is please ? > > The groups defined in conf/directory/default.xml correspond to the "group" channel or group_call API as can be found in conf/dialplan/default.xml, extensions 2000, 2001, and 2002. Go to the fs_cli and type this: group_call sales at 1.1.1.1 (where 1.1.1.1 is your FS IP addr) You'll see that it returns a nicely formatted multiple dialstring for dialing everyone in the group. These have nothing to do with the "callgroup" variable that is defined on each user in the default directory. That is just a variable - it isn't required and doesn't have to be used, but it's available if you want it for some reason. (For example, it will show up in XML CDRs for auth'd calls from the user.) Bottom line: if you're trying to dial multiple users (i.e. "group call") then just use the group definitions in the directory and use either the group_call API (like in ext 2000) or use the "group" channel (like in ext 2001 and 2002). -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/f67b0def/attachment.html From msc at freeswitch.org Thu Nov 19 12:28:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Nov 2009 12:28:39 -0800 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP In-Reply-To: <191c3a030911191218m6fb6992eg5b4eaf338397ed0b@mail.gmail.com> References: <7C3068B7C76746E4A9ACC216574035B9@greyhawk.tonecommander.com> <191c3a030911191218m6fb6992eg5b4eaf338397ed0b@mail.gmail.com> Message-ID: <87f2f3b90911191228i52c89598v78845d80abced257@mail.gmail.com> On Thu, Nov 19, 2009 at 12:18 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > set enable-3pcc to "proxy" instead of "true" > > FYI, the wiki entry is here: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#enable-3pcc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/cb8deb43/attachment-0001.html From msc at freeswitch.org Thu Nov 19 12:30:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Nov 2009 12:30:29 -0800 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: References: Message-ID: <87f2f3b90911191230u512878e3x122efa0357cf83c5@mail.gmail.com> On Thu, Nov 19, 2009 at 11:46 AM, Artem Shiyanov wrote: > Hi there! > > I've got annoying FS behavior: > There are 2 channels executing the same Java application (application > itself is an IVR). If I try to bridge them with uuid_bridged then both > channels are killed. Here is a log from FS console: > uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 > 7d6c08fc-62bf-4a6c-a9ae-763d607e43de > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 > (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> > CS_HIBERNATE > 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1005 at 192.168.147.130 [BREAK] > 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 > (sofia/internal/1001 at master.agent.starpoundtech.net) State Change > CS_EXECUTE -> CS_HIBERNATE > 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called > API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 > 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: > +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de > > freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] > switch_core_session.c:933 Send signal > sofia/internal/1001 at master.agent.starpoundtec > 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal > sofia/internal/1005 at 192.168.147.130 [BREAK] > > 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal > sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] > > (FS version is 1.0.4) > > Any thoughts? > > First, update to latest trunk - there are many behaviors that have been tweaked and repaired since early August when 1.0.4 came out. Try it on latest trunk and see if the behavior persists, is different, or is gone. Please report back and let us know how it all goes. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/f40d08e8/attachment.html From mrene_lists at avgs.ca Thu Nov 19 12:33:17 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 19 Nov 2009 12:33:17 -0800 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: References: Message-ID: I don't see any hangups here, are you talking about the BREAK signals? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 19-Nov-09, at 11:46 AM, Artem Shiyanov wrote: > Hi there! > > I've got annoying FS behavior: > There are 2 channels executing the same Java application > (application itself is an IVR). If I try to bridge them with > uuid_bridged then both channels are killed. Here is a log from FS > console: > uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c- > a9ae-763d607e43de > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 (sofia/internal/1005 at 192.168.147.130 > ) State Change CS_EXECUTE -> CS_HIBERNATE > 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook > called > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done > playing file > 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done > playing file > 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/1005 at 192.168.147.130 [BREAK] > 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 (sofia/internal/1001 at master.agent.starpoundtech.net > ) State Change CS_EXECUTE -> CS_HIBERNATE > 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook > called > API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 > 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: > +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de > > freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] > switch_core_session.c:933 Send signal sofia/internal/1001 at master.agent.starpoundtec > 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send > signal sofia/internal/1005 at 192.168.147.130 [BREAK] > > 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking > stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send > signal sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] > > (FS version is 1.0.4) > > Any thoughts? > > > Artem > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/98559953/attachment.html From stevendt at primrosebank.net Thu Nov 19 12:36:30 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 20:36:30 -0000 Subject: [Freeswitch-users] Extension Configuration - XML File Entriesfor Group configuration References: <73DD76AE07884A1D9535EF27C5841DAD@bp1.ad.bp.com> <87f2f3b90911191225l243a5cdflf8c61b8ebc76dfcb@mail.gmail.com> Message-ID: <843A417506F445D6ABCECFA05037BA2D@bp1.ad.bp.com> Thanks Michael, I think I've got it ! regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Thursday, November 19, 2009 8:25 PM Subject: Re: [Freeswitch-users] Extension Configuration - XML File Entriesfor Group configuration On Thu, Nov 19, 2009 at 3:33 AM, Dave Stevenson wrote: Hi, Can someone please help me understand a little more about Group configuration ? I believe that Group Membership is configured in the \conf\directory\default.xml file I've done this and the caller groups seem to work fine. However, each extension in the \conf\directory\default directory, e.g., 111.xml also has an entry for "callgroup" Can someone explain what the difference in these two options is please ? The groups defined in conf/directory/default.xml correspond to the "group" channel or group_call API as can be found in conf/dialplan/default.xml, extensions 2000, 2001, and 2002. Go to the fs_cli and type this: group_call sales at 1.1.1.1 (where 1.1.1.1 is your FS IP addr) You'll see that it returns a nicely formatted multiple dialstring for dialing everyone in the group. These have nothing to do with the "callgroup" variable that is defined on each user in the default directory. That is just a variable - it isn't required and doesn't have to be used, but it's available if you want it for some reason. (For example, it will show up in XML CDRs for auth'd calls from the user.) Bottom line: if you're trying to dial multiple users (i.e. "group call") then just use the group definitions in the directory and use either the group_call API (like in ext 2000) or use the "group" channel (like in ext 2001 and 2002). -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/2acca687/attachment.html From stevendt at primrosebank.net Thu Nov 19 12:43:51 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 20:43:51 -0000 Subject: [Freeswitch-users] Another Group Question - on VoiceMail Message-ID: Hi again ! I have FreeSwitch configured such that if someone dials in from the PSTN line, a group of phones ring. If nobody answers, the group extension number (100) picks up the call and voice mail kicks in. So far, so good, each of the individual phones logs a missed call and anyone in the group can call into voice mail and go to the extension 100 mailbox to check if there are any messages but the individual phones are not notified that a Voice message is waiting. Is there any way that each extension in the group can be notified that a group Voice Mail is waiting to be picked up so that each phone shows the message waiting indication ? Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/44c0b35b/attachment.html From brian at freeswitch.org Thu Nov 19 12:49:59 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 14:49:59 -0600 Subject: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv Message-ID: We have removed the two modules using the reference code from BroadVoice and added a lib with a new interface from Steve Underwood and mod_bv.c using this lib... We know their is ONE last bug to be fixed in the lib before its working so please do not open any jira's if you try to run it because it will crash right now. Thanks for your understanding and once this is fixed it'll work with aastra and x-lite on both 32bit and 64bit systems without any issues. Thanks, Brian West From msc at freeswitch.org Thu Nov 19 12:57:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Nov 2009 12:57:10 -0800 Subject: [Freeswitch-users] Need help configuring our FreeSWITCH instance In-Reply-To: References: Message-ID: <87f2f3b90911191257n3fbea0eagce482a363c3abb99@mail.gmail.com> On Wed, Nov 18, 2009 at 6:54 PM, John Platts wrote: > I have installed FreeSWITCH on our server, and need some help configuring > our FreeSWITCH instance. All of the numbers associated with our FreeSWITCH > instance are in the format: 1NPANXXXXXX (where NPA is the area code, and > NXXXXXX are the last 7 digits of the phone number). > > I need the following configuration: > > - Calls coming from our IP to IP gateway into our FreeSWITCH instance > needs to be routed to the endpoint that is registered with FreeSWITCH > - Calls coming from any of the registered SIP endpoints need to be sent > to the appropriate destination. The appropriate destination for any number > that is not registered with FreeSWITCH is our IP to IP gateway. > - Our IP to IP gateway does not require any SIP registration or > authentication. > - G.729 (but not G.729 Annex B), G.711 mu-law, and G.711 A-law need to > be enabled > - SIP registrar enabled for registering endpoints other than our IP-IP > gateway > - SIP traffic needs to be accepted to and from both the IP-IP gateway > and from the registered SIP endpoints. > > > How do I get the above configured in FreeSWITCH? > > I'd say you have two choices: roll up your sleeves and start learning or email consulting at freeswitch.org and get some paid help. All of the questions you asked are answered in the wiki (and in some cases, mailing list history) but the answers require some foundational knowledge for them to make sense. If you are not a VoIP user then I'd recommend going the paid route and getting a professional to assist you - it will be the fastest way to get up and running. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/a2a94f08/attachment.html From msc at freeswitch.org Thu Nov 19 13:01:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Nov 2009 13:01:01 -0800 Subject: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv In-Reply-To: References: Message-ID: <87f2f3b90911191301q1083e23fs63fc11722bb60aa5@mail.gmail.com> On Thu, Nov 19, 2009 at 12:49 PM, Brian West wrote: > We have removed the two modules using the reference code from > BroadVoice and added a lib with a new interface from Steve Underwood > and mod_bv.c using this lib... We know their is ONE last bug to be > fixed in the lib before its working so please do not open any jira's > if you try to run it because it will crash right now. > > Thanks for your understanding and once this is fixed it'll work with > aastra and x-lite on both 32bit and 64bit systems without any issues. > > Thanks, Brian West > > Thanks to Brian, Tony, Mike, and Steve U. for all their hard work on this. Not only did they get this implemented quickly, they found a few bugs and reported back to the Broadcom guys. :) Excellent work all the way around. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/e4c46407/attachment.html From lists at tigertech.com Thu Nov 19 13:08:08 2009 From: lists at tigertech.com (Robert L Mathews) Date: Thu, 19 Nov 2009 13:08:08 -0800 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <191c3a030911190811w267162a2p35cf85bb7e62be40@mail.gmail.com> References: <4B032142.1000308@tigertech.com> <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> <4B04682A.6000309@tigertech.com> <191c3a030911181528j7a38ce32gb2fc6fdd585932a9@mail.gmail.com> <4B04EF22.1030404@tigertech.com> <191c3a030911190811w267162a2p35cf85bb7e62be40@mail.gmail.com> Message-ID: <4B05B3B8.6000708@tigertech.com> Anthony Minessale wrote: > Like I said, > The timer by default is designed to make sure that none of the audio is > lost for situations like FAX etc. Right, that makes sense. I've updated the wiki entries I made to warn about this. > We do not use sleep for the timers we have timer objects into the code > derived from a high priority thread sending conditional broadcasts to > the timer objects. Sorry for not being clear. When I said it "sleeps", I just meant "the operating system isn't scheduling any FreeSWITCH threads to run for some period of time, for whatever reason". > What kind of CPU are you using and what kind of hardware that you > suspect you are getting delayed cpu scheduling on a small number of calls? Well, I'm using 2.4 GHz dual Xeons, but couldn't this situation happen on any hardware, if it also has non-FreeSWITCH processes consuming lots of CPU time? That's because the timer needs to make sure that rtp_common_read() is called at least once every 20 ms. If it can't be called that often, for any reason, then FreeSWITCH will fall behind the RTP stream. At that point, audio latency will certainly increase unless some of the packets are discarded. I could duplicate the latency on 1.0.4 by running many other non-FreeSWITCH processes on the same server, so that all the freeswitch threads get starved for CPU time. FreeSWITCH then can't read the RTP packets as fast as they come in, and since the 1.0.4 code didn't flush those extra packets in conferences, that caused noticeable latency. Imposing heavy server load is obviously a silly thing to do, but something similar could happen on any server that fires up lots of non-FreeSWITCH, CPU-hungry processes. (In my case it was virus scanners.) Not using a dedicated server is also silly if people care about call quality, but I was just initially using it for conferences, and I didn't care if some packets were dropped. But conference packet dropping didn't happen on 1.0.4. Instead, a noticeable lag developed, which I did care about. Since 1.0.5 *does* work the way I expect in conferences and other bridged calls (discarding packets), I'm *definitely* not complaining -- please consider this a resolved issue! I agree that it makes sense to preserve all packets for some RTP streams such as faxes and DTMF recognition, and basing that decision on whether the call is bridged makes as much sense as anything else I can think of (although perhaps that flag isn't getting set properly for the third leg of eavesdrop-converted-to-three-way calls). I've been impressed by the extremely high performance of FreeSWITCH. The conference latency I was hearing in 1.0.4 was caused by the fact that I'm stressing the server with separate, unrelated processes, which is a foolish thing to do if you care about audio quality. I was just hoping that FreeSWITCH could more gracefully deal with such foolishness in cases where people *don't* care about audio quality... and 1.0.5 does. That's perfect. Thanks again! -- Robert L Mathews, Tiger Technologies From dave at 3c.co.uk Thu Nov 19 13:15:21 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 19 Nov 2009 14:15:21 -0700 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <4B056F13.6050106@coppice.org> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> <1258646095.12820.300.camel@localhost.localdomain> <4B056F13.6050106@coppice.org> Message-ID: <1258665321.12582.6.camel@localhost.localdomain> On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote: > The audio path between kernel and user space is not stable with any > current PC based telephony system. At some point in the day the odd > chunk of data is lost here and there, whether you use asterisk, > callweaver, yate or FS, with dahdi or sangoma. This is the key problem > for user space echo cancellation. When the path hiccups, the EC goes > crazy, and howls. So far kernel space EC has been the only way to keep > the path length rock solid. Why do you think this is? Getting data from kernel space to user space isn't something which it's difficult to do reliably: the disk system manages it. Even if data is being lost, buffer overruns can be dealt with by using bigger buffers, or timestamping blocks of data on their way in so that missing blocks can be detected. --Dave From lon at kickasspixels.com Thu Nov 19 13:16:53 2009 From: lon at kickasspixels.com (Lon Baker) Date: Thu, 19 Nov 2009 13:16:53 -0800 Subject: [Freeswitch-users] Radius for registration Message-ID: <5d3e0dc60911191316g57a875fbqc15da46fc9847913@mail.gmail.com> Hi everyone, I want to verify what the wiki says, you can use a radius server as the data source for your registrations? Lon From JCasale at activenetwerx.com Thu Nov 19 14:26:02 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 19 Nov 2009 22:26:02 +0000 Subject: [Freeswitch-users] Another Group Question - on VoiceMail In-Reply-To: References: Message-ID: >Is there any way that each extension in the group can be notified that a >group Voice Mail is waiting to be picked up so that each phone shows the >message waiting indication ? Wouldn't this be simply accomplished by setting the vicemail as box 100 for each of the users (such as ext 101....1xx)? From stevendt at primrosebank.net Thu Nov 19 14:49:47 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 22:49:47 -0000 Subject: [Freeswitch-users] Another Group Question - on VoiceMail References: Message-ID: Thanks Joseph, that would be one way, but it would mean that everyone had a common mailbox for all calls, I just wanted to do it for calls coming in on the PSTN line. Maybe that's not possible though ? regards Dave ----- Original Message ----- From: "Joseph L. Casale" To: Sent: Thursday, November 19, 2009 10:26 PM Subject: Re: [Freeswitch-users] Another Group Question - on VoiceMail > >Is there any way that each extension in the group can be notified that a >>group Voice Mail is waiting to be picked up so that each phone shows the >>message waiting indication ? > > Wouldn't this be simply accomplished by setting the vicemail as box 100 > for > each of the users (such as ext 101....1xx)? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch-users-list at metik.com Thu Nov 19 15:23:40 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 19 Nov 2009 18:23:40 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> Message-ID: <4B05D37C.4000607@metik.com> He should be able to just use "Additional Option" to add option 150 (and the associated IP address to which the TFTP server is bound). Brian West wrote: > Some Cisco phones need DHCP option 150. > > /b > > On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: > > >> Metik, >> >> thanks a lot for the tip, I will certainly look at it, particularly >> if it >> does DHCP too. >> >> At the moment, I use my ADSL Router to provide DHCP to the network >> but I've >> just discovered that you can't configure options in its DHCP server >> to point >> to the TFTP server for the phone. At the moment, I have to have the >> phone >> set to a static IP address to be able to configure the TFTP server >> address >> which is not as flexible as using DHCP. I had thought about changing >> over to >> use Windows Server DHCP services but it sounds like ttpd32 would do >> the >> trick. >> >> I just need to decide whether I want all of my machines to rely on >> getting >> their IP address from another PC - it feels like having DHCP in the >> router >> is more robust. >> >> Regards >> Dave >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Mailings at kh-dev.de Thu Nov 19 15:54:04 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Fri, 20 Nov 2009 00:54:04 +0100 Subject: [Freeswitch-users] Media got stuck after attended transfer... In-Reply-To: <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> References: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com> <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com> <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> Message-ID: Hi, one of my customers is willing to contribute for t38 integration. The basic idea is to connect HylaFAX to FS: t38modem <-> FreeSWITCH <-> Media Gateway with t38 support All this without media proxy. Another idea might be to implement t38 origination/termination with a class 1 modem input/output for use with HylaFAX. Do you know how much money we need to collect for t38 support? How much time is needed for implementing this? Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, October 16, 2009 2:10 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Media got stuck after attended transfer... On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga > wrote: hi, any clue when can t38 be added? "Eventually." :) Of course, if we could get more to add to the bounty it might grease the wheels of innovation. http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/c6eb7827/attachment-0001.html From jason at jasonjgw.net Thu Nov 19 17:15:42 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Nov 2009 12:15:42 +1100 Subject: [Freeswitch-users] RTP issues (possibly nat-related) Message-ID: <20091120011542.GA20754@jdc.jasonjgw.net> I have upgraded FreeSWITCH several times recently for testing purposes. Also, my router's configuration has changed slightly as I have moved from tunneled IPv6 to a new native IPv6-over-ADSL trial. However, the problem now is related to my ISP's IPv4-only SIP service, and the symptoms are as follows. 1. If I call a test number, sometimes it all works perfectly. 2. On other occasions (with no discernible pattern) the call connects but no audio is received from the remote end. When this occurs, tshark shows that rtp packets are being sent out to the correct IPv4 address of the server. I am using Stun to handle nat, as my router does not support any of the nat configuration protocols. I want to establish whether it's a router issue or a FreeSWITCH problem. The router is going to be replaced eventually with a small form-factor Linux box and an ADSL2+ card from Traverse Technologies (http://www.traverse.com.au/), but given my priorities at the moment, it won't happen until next year. I can compare SIP traces of that would help. From brian at freeswitch.org Thu Nov 19 17:25:59 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 19:25:59 -0600 Subject: [Freeswitch-users] RTP issues (possibly nat-related) In-Reply-To: <20091120011542.GA20754@jdc.jasonjgw.net> References: <20091120011542.GA20754@jdc.jasonjgw.net> Message-ID: <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> I think the fix for this is coming to an SVN repo near you... so give it a few and update. /b On Nov 19, 2009, at 7:15 PM, Jason White wrote: > I have upgraded FreeSWITCH several times recently for testing > purposes. Also, > my router's configuration has changed slightly as I have moved from > tunneled > IPv6 to a new native IPv6-over-ADSL trial. > > However, the problem now is related to my ISP's IPv4-only SIP > service, and the > symptoms are as follows. > > 1. If I call a test number, sometimes it all works perfectly. > > 2. On other occasions (with no discernible pattern) the call > connects but no > audio is received from the remote end. > > When this occurs, tshark shows that rtp packets are being sent out > to the > correct IPv4 address of the server. > > I am using Stun to handle nat, as my router does not support any of > the nat > configuration protocols. I want to establish whether it's a router > issue or a > FreeSWITCH problem. The router is going to be replaced eventually > with a small > form-factor Linux box and an ADSL2+ card from Traverse Technologies > (http://www.traverse.com.au/), but given my priorities at the > moment, it won't > happen until next year. > > I can compare SIP traces of that would help. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From steveu at coppice.org Thu Nov 19 17:57:34 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 20 Nov 2009 09:57:34 +0800 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258665321.12582.6.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> <1258646095.12820.300.camel@localhost.localdomain> <4B056F13.6050106@coppice.org> <1258665321.12582.6.camel@localhost.localdomain> Message-ID: <4B05F78E.3080007@coppice.org> On 11/20/2009 05:15 AM, David Knell wrote: > On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote: > > >> The audio path between kernel and user space is not stable with any >> current PC based telephony system. At some point in the day the odd >> chunk of data is lost here and there, whether you use asterisk, >> callweaver, yate or FS, with dahdi or sangoma. This is the key problem >> for user space echo cancellation. When the path hiccups, the EC goes >> crazy, and howls. So far kernel space EC has been the only way to keep >> the path length rock solid. >> > Why do you think this is? Getting data from kernel space to user space > isn't something which it's difficult to do reliably: the disk system > manages it. Even if data is being lost, buffer overruns can be dealt > with by using bigger buffers, or timestamping blocks of data on their > way in so that missing blocks can be detected. > Disk isn't audio. Audio is real time, and real time constraints are a harsh mistress. Big buffers are out of the question, due to latency. Some mitigation could be provided if you can detect where missing chunks occur and their exact size. Right now, the I/O schemes do not provide for that, and incorporating support would be tough. You'd need some out of band indication, like an ioctl or something, which would lead to more user space/kernel space exchanges, further increasing the problem. Steve From brian at freeswitch.org Thu Nov 19 18:17:54 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 20:17:54 -0600 Subject: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv In-Reply-To: <87f2f3b90911191301q1083e23fs63fc11722bb60aa5@mail.gmail.com> References: <87f2f3b90911191301q1083e23fs63fc11722bb60aa5@mail.gmail.com> Message-ID: <3E187F0F-66DC-4282-8643-90662F7863BA@freeswitch.org> It now works.. update and have fun! /b On Nov 19, 2009, at 3:01 PM, Michael Collins wrote: > > > On Thu, Nov 19, 2009 at 12:49 PM, Brian West > wrote: > We have removed the two modules using the reference code from > BroadVoice and added a lib with a new interface from Steve Underwood > and mod_bv.c using this lib... We know their is ONE last bug to be > fixed in the lib before its working so please do not open any jira's > if you try to run it because it will crash right now. > > Thanks for your understanding and once this is fixed it'll work with > aastra and x-lite on both 32bit and 64bit systems without any issues. > > Thanks, > Brian West > > Thanks to Brian, Tony, Mike, and Steve U. for all their hard work on > this. Not only did they get this implemented quickly, they found a > few bugs and reported back to the Broadcom guys. :) Excellent work > all the way around. > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/c2ea4953/attachment.html From dfansler at dv-fansler.com Thu Nov 19 18:18:36 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Thu, 19 Nov 2009 21:18:36 -0500 Subject: [Freeswitch-users] APT Utility In-Reply-To: References: <005a01ca6898$f16d99d0$d448cd70$@com> Message-ID: <000301ca6987$c47edbb0$4d7c9310$@com> Thanks for your answers Rob and Shelby. I found more info on apt-get and ran it against all the missing dependences noted. I also ran through the sequence of commands Shelby suggested. In the end, running dpkg -checkbuilddeps I got the following in return dpkg-checkbuilddeps: Unnet builddependencies: debhelper (>=5) then followed the instructions for Ubuntu to enable freeswitch nano /etc/default/freeswitch FREESWITCH_ENABLE="true" And then tried invoke -rc.d freeswitch start but nothing obvious happened. I am only using Ubuntu since it came as a free DVD in the Linux Pro mag that the article about Freeswitch was in. Is there a better version of Linux to use? thanks David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Wednesday, November 18, 2009 5:53 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] APT Utility Hi David, When using Apt, you would install packages with: apt-get install Or search for packages with apt-cache search If you're not root, you'll need to stick "sudo " in front of those command. Honestly, you might want to find a better tutorial with explicit command-by-command instructions. Good luck! Rob On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: Greetings - I am trying to startup a freeSwitch on a P4 running Ubuntu 9.04 "Jaunty". I know very little about Linux. I decided to try this after reading the article in Linux Pro Magazine. I have been following the detailed instructions in the wiki for using Ubuntu Jaunty, however I have run into an unknown - "Use your favorite APT utility to get the needed packages". I am good at following direct instructions - but this statement is too vague for my minimal minimal - did I mention minimal - knowledge of Linux. Could someone please give me detailed instructions on how to use APT utility to get the needed packages - and what are the needed packages? Thanks kindly, David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/06a798be/attachment-0001.html From ujjval at simplesignal.com Thu Nov 19 18:37:06 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Thu, 19 Nov 2009 18:37:06 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C5A@EXMBXCLUS01.citservers.local> Cool, I will explore that option when I have some time. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Wednesday, November 18, 2009 11:02 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hi again UK, IVR is designed to naturally return to previous or top menus. I don't think there's a way to change this default behavior. Maybe its time to move to a script-based pin validation system so you have the full control you need. http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR Rob On Nov 18, 2009, at 11:34 PM, Ujjval Karihaloo wrote: > I have used the following setting in ivr.conf.xml to setup > conferencing with moderator. > > However, the issue I have is - the user enters 123456 and then say > if it's a moderator they enter wrong Moderator PIN 3 times then it > takes the user back to the main menu..."conference_menu" and asks > for main conf pin (123456) once again. > > I would like the caller to be disconnected if they get into the > Moderator menu and enter wrong Moderator PIN 3 times. > > greet-long="welcome_please_enter_conference_pin.wav" > greet-short="check_and_try_again.wav" > invalid-sound="passcode_invalid.wav" > exit-sound="voicemail/vm-goodbye.wav" > timeout="10000" > inter-digit-timeout="5000" > max-failures="3" > max-timeouts="3" > digit-len="7"> > param="conference_123456_moderator_menu" /> > > > greet- > long > = > "conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant > .wav" > greet-short="check_moderator_pin_or_1_to_join.wav" > invalid-sound="invalid_moderator_pin.wav" > exit-sound="voicemail/vm-goodbye.wav" > timeout="10000" > inter-digit-timeout="5000" > max-failures="3" > max-timeouts="3" > digit-len="5"> > > > > > > > > > Ujjval Karihaloo > VP Voice Engineering > IP Phone: +13032428610 > E-Fax: +17202391690 > > SimpleSignal Inc. > 88 Inverness Circle East > Suite K105 > Englewood, CO 80112 > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Thursday, November 05, 2009 7:52 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hi UK, > > From what I've done and read, the caller-controls (in > conference.conf.xml) can be modified to almost anything you can think > of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a > caller control just for those with the moderator flag. So unless you > want everyone able to mute/kick everyone then you can't do it. > > The wiki seems to indicate this as well: > > "Be aware that the caller-controls are applied across the entire > conference. You cannot enter one member of the conference using > caller- > controls ABC and then enter a second member using caller-controls > XYZ." > > http://wiki.freeswitch.org/wiki/Mod_conference > > > I think this might be a limitation of mod_conference. Perhaps one of > the pros can chime in if I'm off-base or there's some nifty way to > accomplish this. > > Cheers, > Rob > > On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > >> Any ideas on the below...has anyone implemented the below: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, November 02, 2009 12:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Rob: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 9:34 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hm, strange. I haven't seen that before. Can you pastebin your logs >> at debug level? >> >> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >> >>> It's strange... a tcpdump tells me that there is no DTMF from my >>> provider when using IVR, but when I call into a TN that goes >>> directly into the Conference App, I see DTMF from the provider. >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 7:23 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I've never had any problem with that. Is your logging at debug >>> level >>> so you can see the RECV DTFM in the log/fs_cli? Are you calling >>> from >>> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >>> isn't passing them through. >>> >>> Make sure your logging is turned up then try something simpler, like >>> calling the echo application, and see if DTFM comes through. >>> >>> Rob >>> >>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>> >>>> Rob: >>>> >>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>> using the below configuration....so it basically timesout. >>>> >>>> UK >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Ujjval Karihaloo >>>> Sent: Monday, October 26, 2009 9:21 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>> Moderator >>>> >>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>> correct.. >>>> =============== >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> ====================== >>>> >>>> I will try it out. >>>> >>>> Again thx a lot for your help. Will keep everyone posted. >>>> >>>> ________________________________________ >>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>> Sent: Friday, October 23, 2009 12:22 PM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>> Moderator >>>> >>>> I just re-tested with the pin in my dial plan: >>>> >>>> >>>> >>>> And it doesn't challenge me for the pin. I just drop right in. I >>>> figured this is how it was intended, since the wiki says the pin is >>>> set initially and only challenged in later attempts [by future >>>> callers]: >>>> >>>> "The first time a conference name (confname) is used, it will be >>>> created on demand, and the pin will be set to what ever is >>>> specified >>>> at that time: the pin in the data string if specified, or if not, >>>> the >>>> "pin" setting in the conference profile, and if that is also >>>> unspecified, then there is no pin protection. Any later attempt to >>>> join the conference must specify the same pin number, if one >>>> existed >>>> when it was created. " >>>> >>>> >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> >>>> I'm sure there's a scenario where its used and useful, the wiki >>>> just >>>> doesn't explain it. >>>> >>>> Rob >>>> >>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>> >>>>> Well first off you're not defining a pine here... >>>>> >>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>> pin >>>>> number] >>>>> >>>>> That might be why its not asking for a pin. >>>>> >>>>> /b >>>>> >>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From b0ef at esben-stien.name Thu Nov 19 19:16:55 2009 From: b0ef at esben-stien.name (Esben Stien) Date: Fri, 20 Nov 2009 04:16:55 +0100 Subject: [Freeswitch-users] Freeswitch Video Capture and Playback Message-ID: <87k4xlga1k.fsf@quasar.esben-stien.name> I'm using ekiga with mod_fsv, trying to record and play back video. When I dial the record extension, it seems to record something, as the video file gets bigger. Trying then to dial the extension for play back, just hangs up, with freeswitch saying: od_fsv.c:247 File version does not match! There seems to be no information on the FSV format or the mod_fsv module on the wiki. Is this at all supposed to work?. What clients and codecs were successful?. Any pointers as to what I can try?. -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n From anthony.minessale at gmail.com Thu Nov 19 18:49:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Nov 2009 20:49:40 -0600 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: References: Message-ID: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> Your "annoying behaviour" is the exact behavior you should be getting considering what you told FS to do. As soon as you call uuid_bridge you are transferring both legs of the call to bridge to each other. This means your java app must exit so the channels can connect to each other. remember that you hangup hook can be called when the channel is transferred not only when it hangs up. you have to test which is happening based on the input to your callback. On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: > Hi there! > > I've got annoying FS behavior: > There are 2 channels executing the same Java application (application > itself is an IVR). If I try to bridge them with uuid_bridged then both > channels are killed. Here is a log from FS console: > uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 > 7d6c08fc-62bf-4a6c-a9ae-763d607e43de > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 > (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> > CS_HIBERNATE > 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1005 at 192.168.147.130 [BREAK] > 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 > (sofia/internal/1001 at master.agent.starpoundtech.net) State Change > CS_EXECUTE -> CS_HIBERNATE > 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called > API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 > 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: > +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de > > freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] > switch_core_session.c:933 Send signal > sofia/internal/1001 at master.agent.starpoundtec > 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal > sofia/internal/1005 at 192.168.147.130 [BREAK] > > 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal > sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] > > (FS version is 1.0.4) > > Any thoughts? > > > Artem > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/8c06b56f/attachment.html From ujjval at simplesignal.com Thu Nov 19 19:17:32 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Thu, 19 Nov 2009 19:17:32 -0800 Subject: [Freeswitch-users] upgrading to latest SVN Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> Getting error below..not sure whats wrong..which line number in what file does this refer to? [root at ss_freeswitch log]# freeswitch 2009-11-19 20:15:44.725118 [INFO] switch_event.c:568 Activate Eventing Engine. 2009-11-19 20:15:44.727095 [DEBUG] switch_event.c:556 Create event dispatch thread 0 Cannot Initialize [[error near line 733]: missing >] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/351b99b6/attachment-0001.html From jason at jasonjgw.net Thu Nov 19 19:30:07 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Nov 2009 14:30:07 +1100 Subject: [Freeswitch-users] upgrading to latest SVN In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> Message-ID: <20091120033007.GA23643@jdc.jasonjgw.net> Ujjval Karihaloo wrote: > Getting error below..not sure whats wrong..which line number in what file > does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. From rob4manhere at gmail.com Thu Nov 19 19:41:30 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 19 Nov 2009 21:41:30 -0600 Subject: [Freeswitch-users] APT Utility In-Reply-To: <000301ca6987$c47edbb0$4d7c9310$@com> References: <005a01ca6898$f16d99d0$d448cd70$@com> <000301ca6987$c47edbb0$4d7c9310$@com> Message-ID: <0B55F774-9F77-4B4D-891D-7FD9595E644A@gmail.com> Ubuntu has pretty good package management with apt-get and should work well for a beginner. The recommended OS (though FreeSWITCH runs on a wide variety of platforms) is 64-bit CentOS. You can get it here: http://www.centos.org/ if you'd like, but at this point I think it's fine to just keep digging into whichever flavor of linux you have handy. If you have FreeSWITCH compiled and installed, have you tried just starting it from the command line? cd /usr/local/freeswitch ; ./bin/ freeswitch On Nov 19, 2009, at 8:18 PM, David V. Fansler wrote: > Thanks for your answers Rob and Shelby. I found more info on apt- > get and ran it against all the missing dependences noted. I also > ran through the sequence of commands Shelby suggested. In the end, > running dpkg ?checkbuilddeps I got the following in return > > dpkg-checkbuilddeps: Unnet builddependencies: debhelper (>=5) > > then followed the instructions for Ubuntu to enable freeswitch > nano /etc/default/freeswitch > FREESWITCH_ENABLE=?true? > > And then tried > invoke ?rc.d freeswitch start > but nothing obvious happened. > > I am only using Ubuntu since it came as a free DVD in the Linux Pro > mag that the article about Freeswitch was in. Is there a better > version of Linux to use? > thanks > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Wednesday, November 18, 2009 5:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] APT Utility > > Hi David, > > When using Apt, you would install packages with: > > apt-get install > > Or search for packages with > > apt-cache search > > > If you're not root, you'll need to stick "sudo " in front of those > command. Honestly, you might want to find a better tutorial with > explicit command-by-command instructions. > > Good luck! > Rob > > On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: > > > Greetings ? I am trying to startup a freeSwitch on a P4 running > Ubuntu 9.04 ?Jaunty?. I know very little about Linux. I decided to > try this after reading the article in Linux Pro Magazine. I have > been following the detailed instructions in the wiki for using > Ubuntu Jaunty, however I have run into an unknown ? ?Use your > favorite APT utility to get the needed packages?. > I am good at following direct instructions ? but this statement is > too vague for my minimal minimal ? did I mention minimal - knowledge > of Linux. > > Could someone please give me detailed instructions on how to use APT > utility to get the needed packages ? and what are the needed packages? > Thanks kindly, > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/1e8440aa/attachment.html From ujjval at simplesignal.com Thu Nov 19 19:49:06 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Thu, 19 Nov 2009 19:49:06 -0800 Subject: [Freeswitch-users] upgrading to latest SVN In-Reply-To: <20091120033007.GA23643@jdc.jasonjgw.net> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> <20091120033007.GA23643@jdc.jasonjgw.net> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C67@EXMBXCLUS01.citservers.local> I really didn't change anything. I was running 1.0.4 and now built from SVN...I see the oddly placed entry in ivr.conf.xml ...removed it now its error on somewhere here: ERROR is: [root at ss_freeswitch freeswitch]# freeswitch 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event dispatch thread 0 Cannot Initialize [[error near line 2840]: unexpected closing tag ] -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, November 19, 2009 8:30 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN Ujjval Karihaloo wrote: > Getting error below..not sure whats wrong..which line number in what file > does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ujjval at simplesignal.com Thu Nov 19 21:22:39 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Thu, 19 Nov 2009 21:22:39 -0800 Subject: [Freeswitch-users] upgrading to latest SVN In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C67@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> <20091120033007.GA23643@jdc.jasonjgw.net> <3C04B27FC880044F8FCD735D0D952FF7175DAC4C67@EXMBXCLUS01.citservers.local> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C72@EXMBXCLUS01.citservers.local> Does svn update try to merge the config files..Need some help, I think it has added some entries in my config files that is causing tag mismatches.. Please advise how to get back my orig config? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo Sent: Thursday, November 19, 2009 8:49 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN I really didn't change anything. I was running 1.0.4 and now built from SVN...I see the oddly placed entry in ivr.conf.xml ...removed it now its error on somewhere here: ERROR is: [root at ss_freeswitch freeswitch]# freeswitch 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event dispatch thread 0 Cannot Initialize [[error near line 2840]: unexpected closing tag ] -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, November 19, 2009 8:30 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN Ujjval Karihaloo wrote: > Getting error below..not sure whats wrong..which line number in what file > does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mrene_lists at avgs.ca Thu Nov 19 21:30:33 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 19 Nov 2009 21:30:33 -0800 Subject: [Freeswitch-users] upgrading to latest SVN In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C72@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> <20091120033007.GA23643@jdc.jasonjgw.net> <3C04B27FC880044F8FCD735D0D952FF7175DAC4C67@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC4C72@EXMBXCLUS01.citservers.local> Message-ID: Nothing will replace your configs automatically in the whole build system. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 19-Nov-09, at 9:22 PM, Ujjval Karihaloo wrote: > Does svn update try to merge the config files..Need some help, I > think it has added some entries in my config files that is causing > tag mismatches.. > > Please advise how to get back my orig config? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Ujjval Karihaloo > Sent: Thursday, November 19, 2009 8:49 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] upgrading to latest SVN > > I really didn't change anything. > > I was running 1.0.4 and now built from SVN...I see the oddly placed > entry in ivr.conf.xml > > ...removed it now its error on > somewhere here: > > > > > > > > > > > > > > ERROR is: > [root at ss_freeswitch freeswitch]# freeswitch > 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate > Eventing Engine. > 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event > dispatch thread 0 > Cannot Initialize [[error near line 2840]: unexpected closing tag section>] > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Jason White > Sent: Thursday, November 19, 2009 8:30 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] upgrading to latest SVN > > Ujjval Karihaloo wrote: >> Getting error below..not sure whats wrong..which line number in >> what file >> does this refer to? > > freeswitch/log/freeswitch.xml.fsxml > > This will be due to a syntax error somewhere in your configuration. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Nov 19 21:37:10 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 20 Nov 2009 00:37:10 -0500 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: <4B0387F1.7070105@savion.huji.ac.il> References: <4B0266F4.8070602@savion.huji.ac.il> <4B0387F1.7070105@savion.huji.ac.il> Message-ID: <193640CC-3E62-4248-8E80-CE7FE82653C0@jerris.com> check out sofia_contact function. If you use this in combination with binding profiles together so they are one table I think this should work right. Mike On Nov 18, 2009, at 12:36 AM, Eli Hayun wrote: > Brian West wrote: >> >> Why do you need to know the destination profile like that? You get to >> pick that on your own so you should already know that before hand. >> >> >> /b >> >> On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote: >> >> >>> Hi >>> We have more then one profile. To make a call I have to enter : bridge >>> sofia/profile/number at ip >>> The problem is when I use : "${use_profile}" I am getting the caller >>> profile, and I need the destination profile. >>> >>> How do I get this information? >>> >> > Thanks for your answer. > > The problem is when I call to that number that the phone hook to other server, I cannot make the call. > Is there is a variable that can tell me the destination profile? > Lets say the other profile called "ph1" I have to dial > sofia/ph1/xxxxx at host to make the call. Is there other way to do that? From jason at jasonjgw.net Thu Nov 19 21:42:03 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Nov 2009 16:42:03 +1100 Subject: [Freeswitch-users] upgrading to latest SVN In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C72@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> <20091120033007.GA23643@jdc.jasonjgw.net> <3C04B27FC880044F8FCD735D0D952FF7175DAC4C67@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC4C72@EXMBXCLUS01.citservers.local> Message-ID: <20091120054203.GA26970@jdc.jasonjgw.net> Ujjval Karihaloo wrote: > Does svn update try to merge the config files..Need some help, I think it > has added some entries in my config files that is causing tag mismatches.. Building and installing FreeSWITCH won't interfere with your configuration. I would suggest using Git or another version control system to keep track of configuration files. I prefer Git. From frank at carmickle.com Thu Nov 19 22:13:47 2009 From: frank at carmickle.com (Frank Carmickle) Date: Fri, 20 Nov 2009 01:13:47 -0500 Subject: [Freeswitch-users] Another Group Question - on VoiceMail In-Reply-To: References: Message-ID: <20091120061347.GB31924@base.carmickle.com> On Thu, Nov 19, Joseph L. Casale wrote: > >Is there any way that each extension in the group can be notified that a > >group Voice Mail is waiting to be picked up so that each phone shows the > >message waiting indication ? > > Wouldn't this be simply accomplished by setting the vicemail as box 100 for > each of the users (such as ext 101....1xx)? Check out the directory parameter MWI-Account. Along with setting the mailbox variable. The variable mailbox sets what voicemail box a person dialing the extension will be dropped in to and MWI_Account, the param, will tell the phone what mailbox to subscribe to. HTH --FC From jason at jasonjgw.net Thu Nov 19 23:01:37 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Nov 2009 18:01:37 +1100 Subject: [Freeswitch-users] RTP issues (possibly nat-related) In-Reply-To: <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> References: <20091120011542.GA20754@jdc.jasonjgw.net> <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> Message-ID: <20091120070137.GA28316@jdc.jasonjgw.net> Brian West wrote: > I think the fix for this is coming to an SVN repo near you... so give > it a few and update. Thanks Brian! I'll watch the svn logs and update when the fix lands. From brian at freeswitch.org Thu Nov 19 23:13:16 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 01:13:16 -0600 Subject: [Freeswitch-users] RTP issues (possibly nat-related) In-Reply-To: <20091120070137.GA28316@jdc.jasonjgw.net> References: <20091120011542.GA20754@jdc.jasonjgw.net> <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> <20091120070137.GA28316@jdc.jasonjgw.net> Message-ID: Update and see if the problem is gone. /b On Nov 20, 2009, at 1:01 AM, Jason White wrote: > > Thanks Brian! > > I'll watch the svn logs and update when the fix lands. From jason at jasonjgw.net Fri Nov 20 00:32:18 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Nov 2009 19:32:18 +1100 Subject: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv In-Reply-To: <3E187F0F-66DC-4282-8643-90662F7863BA@freeswitch.org> References: <87f2f3b90911191301q1083e23fs63fc11722bb60aa5@mail.gmail.com> <3E187F0F-66DC-4282-8643-90662F7863BA@freeswitch.org> Message-ID: <20091120083218.GA17939@jdc.jasonjgw.net> Brian West wrote: > It now works.. update and have fun! Unfortunately it fails to compile under Debian: mod_bv.c:33:24: error: broadvoice.h: No such file or directory The header file exists, so I assume the include path specified to gcc just isn't right. From dfansler at dv-fansler.com Fri Nov 20 03:02:48 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Fri, 20 Nov 2009 06:02:48 -0500 Subject: [Freeswitch-users] APT Utility In-Reply-To: <0B55F774-9F77-4B4D-891D-7FD9595E644A@gmail.com> References: <005a01ca6898$f16d99d0$d448cd70$@com> <000301ca6987$c47edbb0$4d7c9310$@com> <0B55F774-9F77-4B4D-891D-7FD9595E644A@gmail.com> Message-ID: <005301ca69d0$ff7aea80$fe70bf80$@com> Thanks Rob - I guess that it is not installed yet - I get a directory not found trying your suggestion. Taking a look at the freeswitch directory, there is no bin directory. I will keep scratching away at it. David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Thursday, November 19, 2009 10:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] APT Utility Ubuntu has pretty good package management with apt-get and should work well for a beginner. The recommended OS (though FreeSWITCH runs on a wide variety of platforms) is 64-bit CentOS. You can get it here: http://www.centos.org/ if you'd like, but at this point I think it's fine to just keep digging into whichever flavor of linux you have handy. If you have FreeSWITCH compiled and installed, have you tried just starting it from the command line? cd /usr/local/freeswitch ; ./bin/freeswitch On Nov 19, 2009, at 8:18 PM, David V. Fansler wrote: Thanks for your answers Rob and Shelby. I found more info on apt-get and ran it against all the missing dependences noted. I also ran through the sequence of commands Shelby suggested. In the end, running dpkg -checkbuilddeps I got the following in return dpkg-checkbuilddeps: Unnet builddependencies: debhelper (>=5) then followed the instructions for Ubuntu to enable freeswitch nano /etc/default/freeswitch FREESWITCH_ENABLE="true" And then tried invoke -rc.d freeswitch start but nothing obvious happened. I am only using Ubuntu since it came as a free DVD in the Linux Pro mag that the article about Freeswitch was in. Is there a better version of Linux to use? thanks David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Wednesday, November 18, 2009 5:53 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] APT Utility Hi David, When using Apt, you would install packages with: apt-get install Or search for packages with apt-cache search If you're not root, you'll need to stick "sudo " in front of those command. Honestly, you might want to find a better tutorial with explicit command-by-command instructions. Good luck! Rob On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: Greetings - I am trying to startup a freeSwitch on a P4 running Ubuntu 9.04 "Jaunty". I know very little about Linux. I decided to try this after reading the article in Linux Pro Magazine. I have been following the detailed instructions in the wiki for using Ubuntu Jaunty, however I have run into an unknown - "Use your favorite APT utility to get the needed packages". I am good at following direct instructions - but this statement is too vague for my minimal minimal - did I mention minimal - knowledge of Linux. Could someone please give me detailed instructions on how to use APT utility to get the needed packages - and what are the needed packages? Thanks kindly, David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/4c572b1f/attachment-0001.html From rob4manhere at gmail.com Fri Nov 20 03:56:09 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Fri, 20 Nov 2009 05:56:09 -0600 Subject: [Freeswitch-users] APT Utility In-Reply-To: <005301ca69d0$ff7aea80$fe70bf80$@com> References: <005a01ca6898$f16d99d0$d448cd70$@com> <000301ca6987$c47edbb0$4d7c9310$@com> <0B55F774-9F77-4B4D-891D-7FD9595E644A@gmail.com> <005301ca69d0$ff7aea80$fe70bf80$@com> Message-ID: <7296F933-0972-47A7-B988-01557C0BDCC5@gmail.com> Hi David, Apt was just for getting your dependencies in order. Now you can go about the business of compiling and installing Freeswitch. You might start with 1.0.4 so you don't have to mess with svn yet. http://wiki.freeswitch.org/wiki/Installation_Guide#FreeSWITCH_1.0.4_.22Phoenix.22_Release Just keep reading through the wiki or google for tutorials. Good luck! Rob On Nov 20, 2009, at 5:02 AM, David V. Fansler wrote: > Thanks Rob ? I guess that it is not installed yet ? I get a > directory not found trying your suggestion. Taking a look at the > freeswitch directory, there is no bin directory. I will keep > scratching away at it. > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Thursday, November 19, 2009 10:42 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] APT Utility > > Ubuntu has pretty good package management with apt-get and should > work well for a beginner. The recommended OS (though FreeSWITCH > runs on a wide variety of platforms) is 64-bit CentOS. You can get > it here: http://www.centos.org/ if you'd like, but at this point I > think it's fine to just keep digging into whichever flavor of linux > you have handy. > > If you have FreeSWITCH compiled and installed, have you tried just > starting it from the command line? cd /usr/local/freeswitch ; ./bin/ > freeswitch > > > On Nov 19, 2009, at 8:18 PM, David V. Fansler wrote: > > > Thanks for your answers Rob and Shelby. I found more info on apt- > get and ran it against all the missing dependences noted. I also > ran through the sequence of commands Shelby suggested. In the end, > running dpkg ?checkbuilddeps I got the following in return > > dpkg-checkbuilddeps: Unnet builddependencies: debhelper (>=5) > > then followed the instructions for Ubuntu to enable freeswitch > nano /etc/default/freeswitch > FREESWITCH_ENABLE=?true? > > And then tried > invoke ?rc.d freeswitch start > but nothing obvious happened. > > I am only using Ubuntu since it came as a free DVD in the Linux Pro > mag that the article about Freeswitch was in. Is there a better > version of Linux to use? > thanks > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Wednesday, November 18, 2009 5:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] APT Utility > > Hi David, > > When using Apt, you would install packages with: > > apt-get install > > Or search for packages with > > apt-cache search > > > If you're not root, you'll need to stick "sudo " in front of those > command. Honestly, you might want to find a better tutorial with > explicit command-by-command instructions. > > Good luck! > Rob > > On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: > > > > Greetings ? I am trying to startup a freeSwitch on a P4 running > Ubuntu 9.04 ?Jaunty?. I know very little about Linux. I decided to > try this after reading the article in Linux Pro Magazine. I have > been following the detailed instructions in the wiki for using > Ubuntu Jaunty, however I have run into an unknown ? ?Use your > favorite APT utility to get the needed packages?. > I am good at following direct instructions ? but this statement is > too vague for my minimal minimal ? did I mention minimal - knowledge > of Linux. > > Could someone please give me detailed instructions on how to use APT > utility to get the needed packages ? and what are the needed packages? > Thanks kindly, > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/454fac6e/attachment-0001.html From siniypin at gmail.com Fri Nov 20 04:30:00 2009 From: siniypin at gmail.com (RobertT) Date: Fri, 20 Nov 2009 15:30:00 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> Message-ID: <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> Well, I start 2 user agents. Each of them successfully registers as 1000 & 1001 extensions via tcp SIP transport. Then I issue a call, say from 1000 to 1001, and watch it being disconnected in several seconds by recieving client due to abovementioned conditions (no completing answer from FS). Why is it happening??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/a8b82cbb/attachment.html From kjv at ken-ton.com Fri Nov 20 05:44:13 2009 From: kjv at ken-ton.com (Karl J. Vesterling) Date: Fri, 20 Nov 2009 08:44:13 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> <868A4E38-D947-4291-BBD7-4F4C9E5B239E@ken-ton.com> Message-ID: <257758F8-77DF-4D05-BA80-B3E115CCD5AB@ken-ton.com> Then why is the 7540 in the Subject of this thread? I hadn't found data on the 7540 either, but hey, for a while the Cisco WAAS device was a Cisco Product and had not any support from Cisco. Matter of fact, I had to get the regional sales director on the phone to argue successfully to the cisco support drone that the WAAS Devices were indeed Cisco devices, and that we were entitled to support, and that Cisco did have a support queue for the WAAS product. (It was quite humorous to listen in as the banter went back and forth. At which point I asked, "Do you now understand the source of my frustration?") (This wasn't a problem before since we had a back door into the WAAS developers, but our back door was on vacation at the time of this support request.) So, you can see my confusion here with regard to 7540 vs 7940... Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Nov 19, 2009, at 1:17 PM, Brian West wrote: > I don't think a 7540 exists. > > /b > > On Nov 19, 2009, at 12:11 PM, Karl J. Vesterling wrote: > >> I haven't provisioned any 7540's... Good luck! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From kjv at ken-ton.com Fri Nov 20 05:45:32 2009 From: kjv at ken-ton.com (Karl J. Vesterling) Date: Fri, 20 Nov 2009 08:45:32 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <4B05D37C.4000607@metik.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> <4B05D37C.4000607@metik.com> Message-ID: <40B86F87-D47E-4164-9ABD-794CA96B7B41@ken-ton.com> You'd think so wouldn't you... Even the DHCP Server with Snow Leopard Server lacks this basic functionality. If someone knows how to enable it, please post destructions on how-to here... Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Nov 19, 2009, at 6:23 PM, Metik wrote: > He should be able to just use "Additional Option" to add option 150 (and > the associated IP address to which the TFTP server is bound). > > Brian West wrote: >> Some Cisco phones need DHCP option 150. >> >> /b >> >> On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: >> >> >>> Metik, >>> >>> thanks a lot for the tip, I will certainly look at it, particularly >>> if it >>> does DHCP too. >>> >>> At the moment, I use my ADSL Router to provide DHCP to the network >>> but I've >>> just discovered that you can't configure options in its DHCP server >>> to point >>> to the TFTP server for the phone. At the moment, I have to have the >>> phone >>> set to a static IP address to be able to configure the TFTP server >>> address >>> which is not as flexible as using DHCP. I had thought about changing >>> over to >>> use Windows Server DHCP services but it sounds like ttpd32 would do >>> the >>> trick. >>> >>> I just need to decide whether I want all of my machines to rely on >>> getting >>> their IP address from another PC - it feels like having DHCP in the >>> router >>> is more robust. >>> >>> Regards >>> Dave >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Fri Nov 20 06:25:44 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 08:25:44 -0600 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> Message-ID: <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> Well depends are you using x-lite 4 beta? you didn't include ANY logs... I know TCP to TCP works fine I use that daily. can you include some debug logs or join #freeswitch on irc.freenode.net? /b On Nov 20, 2009, at 6:30 AM, RobertT wrote: > Well, I start 2 user agents. Each of them successfully registers as > 1000 & 1001 extensions via tcp SIP transport. Then I issue a call, > say from 1000 to 1001, and watch it being disconnected in several > seconds by recieving client due to abovementioned conditions (no > completing answer from FS). Why is it happening??? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Fri Nov 20 06:33:06 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 08:33:06 -0600 Subject: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv In-Reply-To: <20091120083218.GA17939@jdc.jasonjgw.net> References: <87f2f3b90911191301q1083e23fs63fc11722bb60aa5@mail.gmail.com> <3E187F0F-66DC-4282-8643-90662F7863BA@freeswitch.org> <20091120083218.GA17939@jdc.jasonjgw.net> Message-ID: You'll have to rebootstrap and make sure libs/broadvoice builds /b On Nov 20, 2009, at 2:32 AM, Jason White wrote: > Brian West wrote: >> It now works.. update and have fun! > > Unfortunately it fails to compile under Debian: > > mod_bv.c:33:24: error: broadvoice.h: No such file or directory > > The header file exists, so I assume the include path specified to > gcc just > isn't right. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/2b08e373/attachment.html From dave at 3c.co.uk Fri Nov 20 07:51:37 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 20 Nov 2009 08:51:37 -0700 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <4B05F78E.3080007@coppice.org> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> <1258646095.12820.300.camel@localhost.localdomain> <4B056F13.6050106@coppice.org> <1258665321.12582.6.camel@localhost.localdomain> <4B05F78E.3080007@coppice.org> Message-ID: <1258732297.12582.41.camel@localhost.localdomain> On Fri, 2009-11-20 at 09:57 +0800, Steve Underwood wrote: > On 11/20/2009 05:15 AM, David Knell wrote: > > On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote: > > > > > >> The audio path between kernel and user space is not stable with any > >> current PC based telephony system. At some point in the day the odd > >> chunk of data is lost here and there, whether you use asterisk, > >> callweaver, yate or FS, with dahdi or sangoma. This is the key problem > >> for user space echo cancellation. When the path hiccups, the EC goes > >> crazy, and howls. So far kernel space EC has been the only way to keep > >> the path length rock solid. > >> > > Why do you think this is? Getting data from kernel space to user space > > isn't something which it's difficult to do reliably: the disk system > > manages it. Even if data is being lost, buffer overruns can be dealt > > with by using bigger buffers, or timestamping blocks of data on their > > way in so that missing blocks can be detected. > > > Disk isn't audio. Audio is real time, and real time constraints are a > harsh mistress. Big buffers are out of the question, due to latency. Not necessarily. A decent-sized FIFO, mostly run empty, but there to buffer data in the case of the user-side not being able to accept it for a short period wouldn't necessarily add to latency unless it were needed. The user side could then make a decision as to how to deal with the queued data - dump it or handle it - according to its requirements. > Some mitigation could be provided if you can detect where missing chunks > occur and their exact size. Right now, the I/O schemes do not provide > for that, and incorporating support would be tough. You'd need some out > of band indication, like an ioctl or something, which would lead to more > user space/kernel space exchanges, further increasing the problem. I don't think it'd be all that hard. Were I to do this, I'd probably: - define an error return (ESLIP, EDATALOST, something like that) which might be returned by read/write - add an ioctl to enable and disable it - maybe add an ioctl to indicate how much data's been lost Doesn't break existing stuff, doesn't add any overhead under normal conditions, would be handy for better reliability with EC, DTMF, fax, etc. --Dave From jlenk at frontiernet.net Fri Nov 20 07:59:28 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Fri, 20 Nov 2009 07:59:28 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <1258634740580-4031590.post@n2.nabble.com> References: <1258634740580-4031590.post@n2.nabble.com> Message-ID: <1258732768082-4038514.post@n2.nabble.com> That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch-users-list at metik.com Fri Nov 20 08:04:24 2009 From: freeswitch-users-list at metik.com (Metik) Date: Fri, 20 Nov 2009 11:04:24 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <40B86F87-D47E-4164-9ABD-794CA96B7B41@ken-ton.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> <4B05D37C.4000607@metik.com> <40B86F87-D47E-4164-9ABD-794CA96B7B41@ken-ton.com> Message-ID: <4B06BE08.20703@metik.com> Although I'm not familiar with Snow Leopard's DHCP server implementation, I would assume that to expose that functionality--it is a matter of tweaking what is under the hood. If that is not the case--you could just build and/or install ISC's DHCP server. As far as tftpd32, I have used it in the past and it does support it. It is not by any means feature rich but should suffice given his needs. The other alternative is to install Cygwin-based build of the ISC DHCP server. -metik Karl J. Vesterling wrote: > You'd think so wouldn't you... > > Even the DHCP Server with Snow Leopard Server lacks this basic functionality. > If someone knows how to enable it, please post destructions on how-to here... > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 > > On Nov 19, 2009, at 6:23 PM, Metik wrote: > > >> He should be able to just use "Additional Option" to add option 150 (and >> the associated IP address to which the TFTP server is bound). >> >> Brian West wrote: >> >>> Some Cisco phones need DHCP option 150. >>> >>> /b >>> >>> On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: >>> >>> >>> >>>> Metik, >>>> >>>> thanks a lot for the tip, I will certainly look at it, particularly >>>> if it >>>> does DHCP too. >>>> >>>> At the moment, I use my ADSL Router to provide DHCP to the network >>>> but I've >>>> just discovered that you can't configure options in its DHCP server >>>> to point >>>> to the TFTP server for the phone. At the moment, I have to have the >>>> phone >>>> set to a static IP address to be able to configure the TFTP server >>>> address >>>> which is not as flexible as using DHCP. I had thought about changing >>>> over to >>>> use Windows Server DHCP services but it sounds like ttpd32 would do >>>> the >>>> trick. >>>> >>>> I just need to decide whether I want all of my machines to rely on >>>> getting >>>> their IP address from another PC - it feels like having DHCP in the >>>> router >>>> is more robust. >>>> >>>> Regards >>>> Dave >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From kristian.kielhofner at gmail.com Fri Nov 20 08:41:22 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 20 Nov 2009 11:41:22 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> Message-ID: <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> Finally got a chance to test this, the results are the same. Why am I getting this? Is it because I'm executing ring_ready before I attempt the bridge? Is it because I'm using a socket? On Wed, Nov 11, 2009 at 10:14 PM, Anthony Minessale wrote: > dont execute bridge that way, your bridge itself is the other thing already > broadcasting. > > > api uuid_transfer bridge:sofia/myprofile/foo at bar.com inline > > if you want to do more after the bridge > set the variable park_after_bridge=true to make it go back to idle > > > On Wed, Nov 11, 2009 at 4:33 PM, Kristian Kielhofner > wrote: >> >> Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64. >> >> On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner >> wrote: >> > From the trace: >> > >> ..snip.. >> >> -- >> Kristian Kielhofner >> http://www.astlinux.org >> http://blog.krisk.org >> http://www.star2star.com >> http://www.submityoursip.com >> http://www.voalte.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From info at daccii.it Fri Nov 20 08:46:17 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Fri, 20 Nov 2009 17:46:17 +0100 Subject: [Freeswitch-users] Call doesn't work while registration work for a VOIP provider In-Reply-To: References: <4B059CA7.3040201@daccii.it> Message-ID: <4B06C7D9.9010901@daccii.it> Thank you, this works perfectly! Brian West ha scritto: > I'm going to guess gw+exsorsa is what they don't like. try extensions- > in-contact=true on the gateway config. > > /b -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/82442704/attachment.vcf From brian at freeswitch.org Fri Nov 20 08:49:50 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 10:49:50 -0600 Subject: [Freeswitch-users] Call doesn't work while registration work for a VOIP provider In-Reply-To: <4B06C7D9.9010901@daccii.it> References: <4B059CA7.3040201@daccii.it> <4B06C7D9.9010901@daccii.it> Message-ID: <6ADE7D71-9D61-485B-B829-005540D09610@freeswitch.org> Can you please document that on the wiki for the providers and in the sofia config pages? /b On Nov 20, 2009, at 10:46 AM, Albano Daniele Salvatore - Lavoro wrote: > Thank you, this works perfectly! > > Brian West ha scritto: >> I'm going to guess gw+exsorsa is what they don't like. try >> extensions- in-contact=true on the gateway config. >> /b > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From gauravs74 at yahoo.com Fri Nov 20 01:16:34 2009 From: gauravs74 at yahoo.com (Gaurav Singh) Date: Fri, 20 Nov 2009 01:16:34 -0800 (PST) Subject: [Freeswitch-users] Broadvoice 32 transcoding support? Message-ID: <144359.82983.qm@web113920.mail.gq1.yahoo.com> Hi, Does freeswitch support transcoding between broadvoice (BV32 ) and G711 ? Did anyone try using freeswitch with Xten/counterpath Sip phone configured with broadvoice32? Also, can someone recomend? another free sip phone supporting BV 32? Thanks Gaurav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/23f793ef/attachment.html From igor at 3gnt.net Fri Nov 20 09:09:26 2009 From: igor at 3gnt.net (Igor Neves) Date: Fri, 20 Nov 2009 17:09:26 +0000 Subject: [Freeswitch-users] freeswitch.spec patch Message-ID: <4B06CD46.6050408@3gnt.net> Hi, Attached it's a patch that corrects the problem when doing upgrade to other older version of freeswitch rpm the freeswitch user was being deleted. This patch was made against freeswitch.spec from freeswitch 1.0.4. It should be applied with "patch -p0 < freeswitch.spec". Cheers, -- Igor Neves 3GNTW - Tecnologias de Informa??o, Lda SIP: igor at 3gnt.net JID: igor at 3gnt.net ICQ: 249075444 MSN: igor at 3gnt.net TLM: 00351914503611 PSTN: 00351252377120 -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: freesswitch.patch Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/b9ad9cbc/attachment.pl From william.suffill at gmail.com Fri Nov 20 09:19:39 2009 From: william.suffill at gmail.com (William Suffill) Date: Fri, 20 Nov 2009 12:19:39 -0500 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <4B06CD46.6050408@3gnt.net> References: <4B06CD46.6050408@3gnt.net> Message-ID: <6b65470d0911200919y34546bvc93f3a5c976c1dc7@mail.gmail.com> 1.0.4 is quite old at this point so patches should be against trunk from SVN to make sure they apply against the latest codebase instead of the released version. Also the project prefers that patches be kept in Jira for tracking purposes. http://jira.freeswitch.org/ Thanks for the contribution though and the patch looks small enough that it should be easy to apply to trunk. From itamar at ispbrasil.com.br Fri Nov 20 09:20:47 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Fri, 20 Nov 2009 15:20:47 -0200 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <4B06CD46.6050408@3gnt.net> References: <4B06CD46.6050408@3gnt.net> Message-ID: do you like to help packaging it for fedora, centos and rhel ? On Fri, Nov 20, 2009 at 3:09 PM, Igor Neves wrote: > Hi, > > Attached it's a patch that corrects the problem when doing upgrade to other > older version of freeswitch rpm the freeswitch user was being deleted. > > This patch was made against freeswitch.spec from freeswitch 1.0.4. > It should be applied with "patch -p0 < freeswitch.spec". > > Cheers, > > -- > Igor Neves > 3GNTW - Tecnologias de Informa??o, Lda > > ?SIP: igor at 3gnt.net ? ? JID: igor at 3gnt.net > ?ICQ: 249075444 ? ? ? ? MSN: igor at 3gnt.net > ?TLM: 00351914503611 ? ?PSTN: 00351252377120 > ------------ Itamar Reis Peixoto e-mail/msn/google talk/sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From igor at 3gnt.net Fri Nov 20 09:29:06 2009 From: igor at 3gnt.net (Igor Neves) Date: Fri, 20 Nov 2009 17:29:06 +0000 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: References: <4B06CD46.6050408@3gnt.net> Message-ID: <4B06D1E2.3060008@3gnt.net> Hi, I have it working on CentOS, I can help packaging it, what does it involves more precisely? On 11/20/2009 05:20 PM, Itamar Reis Peixoto wrote: > do you like to help packaging it for fedora, centos and rhel ? > > > > On Fri, Nov 20, 2009 at 3:09 PM, Igor Neves wrote: > >> Hi, >> >> Attached it's a patch that corrects the problem when doing upgrade to other >> older version of freeswitch rpm the freeswitch user was being deleted. >> >> This patch was made against freeswitch.spec from freeswitch 1.0.4. >> It should be applied with "patch -p0< freeswitch.spec". >> >> Cheers, >> >> -- >> Igor Neves >> 3GNTW - Tecnologias de Informa??o, Lda >> >> SIP: igor at 3gnt.net JID: igor at 3gnt.net >> ICQ: 249075444 MSN: igor at 3gnt.net >> TLM: 00351914503611 PSTN: 00351252377120 >> >> > ------------ > > Itamar Reis Peixoto > > e-mail/msn/google talk/sip: itamar at ispbrasil.com.br > skype: itamarjp > icq: 81053601 > +55 11 4063 5033 > +55 34 3221 8599 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Igor Neves 3GNTW - Tecnologias de Informa??o, Lda SIP: igor at 3gnt.net JID: igor at 3gnt.net ICQ: 249075444 MSN: igor at 3gnt.net TLM: 00351914503611 PSTN: 00351252377120 From igor at 3gnt.net Fri Nov 20 09:31:18 2009 From: igor at 3gnt.net (Igor Neves) Date: Fri, 20 Nov 2009 17:31:18 +0000 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <6b65470d0911200919y34546bvc93f3a5c976c1dc7@mail.gmail.com> References: <4B06CD46.6050408@3gnt.net> <6b65470d0911200919y34546bvc93f3a5c976c1dc7@mail.gmail.com> Message-ID: <4B06D266.7020109@3gnt.net> Ok, But how should I proceed? Thanks, On 11/20/2009 05:19 PM, William Suffill wrote: > 1.0.4 is quite old at this point so patches should be against trunk > from SVN to make sure they apply against the latest codebase instead > of the released version. Also the project prefers that patches be kept > in Jira for tracking purposes. > > http://jira.freeswitch.org/ > > Thanks for the contribution though and the patch looks small enough > that it should be easy to apply to trunk. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Igor Neves 3GNTW - Tecnologias de Informa??o, Lda SIP: igor at 3gnt.net JID: igor at 3gnt.net ICQ: 249075444 MSN: igor at 3gnt.net TLM: 00351914503611 PSTN: 00351252377120 From brian at freeswitch.org Fri Nov 20 09:33:12 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 11:33:12 -0600 Subject: [Freeswitch-users] Broadvoice 32 transcoding support? In-Reply-To: <144359.82983.qm@web113920.mail.gq1.yahoo.com> References: <144359.82983.qm@web113920.mail.gq1.yahoo.com> Message-ID: <5C5BA5E8-0AB7-4251-8557-46E39503FEC9@freeswitch.org> Yes it works. You'll need SVN as of last night. Build mod_bv and load it. Works with Aastra and x-lite as far as I can tell... I do have one issue lingering with x-lite but its only when calling 9999 but i'm working on that one. I have no idea why you would want to transcode from BV32 to G711 since its a 16k to 8k resample too... you might as well use BV16 if you are doing that. /b On Nov 20, 2009, at 3:16 AM, Gaurav Singh wrote: > Hi, > > Does freeswitch support transcoding between broadvoice (BV32 ) and > G711 ? > > Did anyone try using freeswitch with Xten/counterpath Sip phone > configured with broadvoice32? Also, can someone recomend another > free sip phone supporting BV 32? > > Thanks > Gaurav > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/406f00d1/attachment.html From brian at freeswitch.org Fri Nov 20 09:34:07 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 11:34:07 -0600 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <4B06D1E2.3060008@3gnt.net> References: <4B06CD46.6050408@3gnt.net> <4B06D1E2.3060008@3gnt.net> Message-ID: Well contribute your patches against SVN Trunk to http://jira.freeswitch.org /b On Nov 20, 2009, at 11:29 AM, Igor Neves wrote: > Hi, > > I have it working on CentOS, I can help packaging it, what does it > involves more precisely? > > > On 11/20/2009 05:20 PM, Itamar Reis Peixoto wrote: >> do you like to help packaging it for fedora, centos and rhel ? >> >> From msc at freeswitch.org Fri Nov 20 09:34:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Nov 2009 09:34:38 -0800 Subject: [Freeswitch-users] Broadvoice 32 transcoding support? In-Reply-To: <144359.82983.qm@web113920.mail.gq1.yahoo.com> References: <144359.82983.qm@web113920.mail.gq1.yahoo.com> Message-ID: <87f2f3b90911200934n20373bc6tf01677ec8d2bb11d@mail.gmail.com> On Fri, Nov 20, 2009 at 1:16 AM, Gaurav Singh wrote: > Hi, > > Does freeswitch support transcoding between broadvoice (BV32 ) and G711 ? > Try latest trunk. There was a new update just added very recently... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/39e52e58/attachment.html From brian at freeswitch.org Fri Nov 20 09:34:47 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 11:34:47 -0600 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <4B06D266.7020109@3gnt.net> References: <4B06CD46.6050408@3gnt.net> <6b65470d0911200919y34546bvc93f3a5c976c1dc7@mail.gmail.com> <4B06D266.7020109@3gnt.net> Message-ID: <925C296B-2582-4A38-9E43-A4FBF9B9224E@freeswitch.org> Hope on IRC and talk to MikeJ in #freeswitch he can direct you better on what to do vs not do since he maintains the builds system in FreeSWITCH. /b On Nov 20, 2009, at 11:31 AM, Igor Neves wrote: > Ok, > > But how should I proceed? > > Thanks, From msc at freeswitch.org Fri Nov 20 09:40:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Nov 2009 09:40:46 -0800 Subject: [Freeswitch-users] APT Utility In-Reply-To: <7296F933-0972-47A7-B988-01557C0BDCC5@gmail.com> References: <005a01ca6898$f16d99d0$d448cd70$@com> <000301ca6987$c47edbb0$4d7c9310$@com> <0B55F774-9F77-4B4D-891D-7FD9595E644A@gmail.com> <005301ca69d0$ff7aea80$fe70bf80$@com> <7296F933-0972-47A7-B988-01557C0BDCC5@gmail.com> Message-ID: <87f2f3b90911200940m12f339dbkf511dc76dee35ec2@mail.gmail.com> On Fri, Nov 20, 2009 at 3:56 AM, Rob Forman wrote: > Hi David, > > Apt was just for getting your dependencies in order. Now you can go about > the business of compiling and installing Freeswitch. You might start with > 1.0.4 so you don't have to mess with svn yet. > > > http://wiki.freeswitch.org/wiki/Installation_Guide#FreeSWITCH_1.0.4_.22Phoenix.22_Release > > Just keep reading through the wiki or google for tutorials. Good luck! > Rob > > Actually, if you want to avoid SVN then get the latest 1.0.5 pre-release version. It will be more stable than 1.0.4. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/1492290a/attachment.html From costa.zikalala at gmail.com Fri Nov 20 09:51:59 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Fri, 20 Nov 2009 19:51:59 +0200 Subject: [Freeswitch-users] phpmod fails to make Message-ID: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> I've been trying to make phpmod without any success. I've even tried to ./configure --with-php and it did't help. I've just upgraded to latest svn and am running FS on FC11. I keep getting this error message: make: php-config: Command not found g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:717:18: error: zend.h: No such file or directory esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory esl_wrap.cpp:719:17: error: php.h: No such file or directory esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or directory esl_wrap.cpp:980:17: error: esl.h: No such file or directory esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? declared void esl_wrap.cpp:793: error: ?zval? was not declared in this scope esl_wrap.cpp:793: error: ?z? was not declared in this scope esl_wrap.cpp:793: error: expected primary-expression before ?void? esl_wrap.cpp:793: error: expected primary-expression before ?*? token esl_wrap.cpp:793: error: ?type? was not declared in this scope esl_wrap.cpp:793: error: expected primary-expression before ?int? make: *** [esl_wrap.o] Error 1 Please help. Thanks Costa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/e992646d/attachment.html From brian at freeswitch.org Fri Nov 20 09:58:36 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 11:58:36 -0600 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> Message-ID: I'm going to guess you did cd libs/esl/php then typed make.. move up one dir first then type make phpmod.. but you seem to be missing all the php dev headers. /b On Nov 20, 2009, at 11:51 AM, Costa Zikalala wrote: > I've been trying to make phpmod without any success. I've even tried > to ./configure --with-php and it did't help. > I've just upgraded to latest svn and am running FS on FC11. > > I keep getting this error message: > > make: php-config: Command not found > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o > esl_wrap.o > esl_wrap.cpp:717:18: error: zend.h: No such file or directory > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory > esl_wrap.cpp:719:17: error: php.h: No such file or directory > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or > directory > esl_wrap.cpp:980:17: error: esl.h: No such file or directory > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope > esl_wrap.cpp:788: error: ISO C++ forbids declaration of > ?ZEND_RSRC_DTOR_FUNC? with no type > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this > scope > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? > declared void > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > esl_wrap.cpp:793: error: ?z? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?void? > esl_wrap.cpp:793: error: expected primary-expression before ?*? token > esl_wrap.cpp:793: error: ?type? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?int? > make: *** [esl_wrap.o] Error 1 > > Please help. > Thanks > Costa From intralanman at freeswitch.org Fri Nov 20 10:05:03 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 20 Nov 2009 13:05:03 -0500 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> Message-ID: try installing php-devel -Ray On Nov 20, 2009, at 12:51 PM, Costa Zikalala wrote: > I've been trying to make phpmod without any success. I've even tried > to ./configure --with-php and it did't help. > I've just upgraded to latest svn and am running FS on FC11. > > I keep getting this error message: > > make: php-config: Command not found > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o > esl_wrap.o > esl_wrap.cpp:717:18: error: zend.h: No such file or directory > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory > esl_wrap.cpp:719:17: error: php.h: No such file or directory > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or > directory > esl_wrap.cpp:980:17: error: esl.h: No such file or directory > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope > esl_wrap.cpp:788: error: ISO C++ forbids declaration of > ?ZEND_RSRC_DTOR_FUNC? with no type > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this > scope > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? > declared void > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > esl_wrap.cpp:793: error: ?z? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?void? > esl_wrap.cpp:793: error: expected primary-expression before ?*? token > esl_wrap.cpp:793: error: ?type? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?int? > make: *** [esl_wrap.o] Error 1 > > Please help. > Thanks > Costa > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Fri Nov 20 10:06:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Nov 2009 10:06:43 -0800 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> Message-ID: <87f2f3b90911201006t15f69ef5pa4cd2a265a9f3c2b@mail.gmail.com> On Fri, Nov 20, 2009 at 9:51 AM, Costa Zikalala wrote: > I've been trying to make phpmod without any success. I've even tried to > ./configure --with-php and it did't help. > I've just upgraded to latest svn and am running FS on FC11. > > I keep getting this error message: > > make: php-config: Command not found > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o > esl_wrap.cpp:717:18: error: zend.h: No such file or directory > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory > esl_wrap.cpp:719:17: error: php.h: No such file or directory > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or directory > esl_wrap.cpp:980:17: error: esl.h: No such file or directory > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope > esl_wrap.cpp:788: error: ISO C++ forbids declaration of > ?ZEND_RSRC_DTOR_FUNC? with no type > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this scope > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? > declared void > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > esl_wrap.cpp:793: error: ?z? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?void? > esl_wrap.cpp:793: error: expected primary-expression before ?*? token > esl_wrap.cpp:793: error: ?type? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?int? > make: *** [esl_wrap.o] Error 1 > > Did you install the php-devel stuff? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/27494a57/attachment.html From costa.zikalala at gmail.com Fri Nov 20 10:50:26 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Fri, 20 Nov 2009 20:50:26 +0200 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> Message-ID: <59daa2cd0911201050x18e60941le605d0a40da9cf48@mail.gmail.com> Thanks for quick responses guys, yes I was doing it under the php directory. I've now also installed php-devel, and am now getting this error: make[1]: Entering directory `/home/Costa/freeswitch-1.0.4/libs/esl/php' g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lcrypt -lcrypt -ledit -lncurses -lresolv -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt -lm -lcrypt -o ESL.so -L. /usr/bin/ld: cannot find -ledit collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make[1]: Leaving directory `/home/Costa/freeswitch-1.0.4/libs/esl/php' make: *** [phpmod] Error 2 2009/11/20 Brian West > I'm going to guess you did cd libs/esl/php then typed make.. move up > one dir first then type make phpmod.. but you seem to be missing all > the php dev headers. > > /b > > On Nov 20, 2009, at 11:51 AM, Costa Zikalala wrote: > > > I've been trying to make phpmod without any success. I've even tried > > to ./configure --with-php and it did't help. > > I've just upgraded to latest svn and am running FS on FC11. > > > > I keep getting this error message: > > > > make: php-config: Command not found > > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o > > esl_wrap.o > > esl_wrap.cpp:717:18: error: zend.h: No such file or directory > > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory > > esl_wrap.cpp:719:17: error: php.h: No such file or directory > > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory > > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or > > directory > > esl_wrap.cpp:980:17: error: esl.h: No such file or directory > > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory > > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope > > esl_wrap.cpp:788: error: ISO C++ forbids declaration of > > ?ZEND_RSRC_DTOR_FUNC? with no type > > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this > > scope > > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token > > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? > > declared void > > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > > esl_wrap.cpp:793: error: ?z? was not declared in this scope > > esl_wrap.cpp:793: error: expected primary-expression before ?void? > > esl_wrap.cpp:793: error: expected primary-expression before ?*? token > > esl_wrap.cpp:793: error: ?type? was not declared in this scope > > esl_wrap.cpp:793: error: expected primary-expression before ?int? > > make: *** [esl_wrap.o] Error 1 > > > > Please help. > > Thanks > > Costa > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/0affb30a/attachment-0001.html From siniypin at gmail.com Fri Nov 20 11:07:22 2009 From: siniypin at gmail.com (RobertT) Date: Fri, 20 Nov 2009 22:07:22 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> Message-ID: <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> No, I don't use Xlite. I use my own .Net wrapper around pjsip ua lib. Foreseeing uncertaincies about it's quality I may say that pjsua reference implementation yields the same results in this scenario. Actually I have no doubt that FS is working nicely with tcp and tls as well because I had it working till some moment. And I don't know what the hell happened. =( In order to check if it is something related to my config I switched it back to default and conducted the same test with (urghhh) no luck as well. So now I wonder what could cause this very-very strange behavior? Some issues with network? But why the UDP works then? All traces (FS SIP, FS console, SIP caller and callee's) are here: http://pastebin.com/m2008de4e Regards, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/6a84b66a/attachment.html From intralanman at freeswitch.org Fri Nov 20 11:14:47 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 20 Nov 2009 14:14:47 -0500 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: <59daa2cd0911201050x18e60941le605d0a40da9cf48@mail.gmail.com> References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> <59daa2cd0911201050x18e60941le605d0a40da9cf48@mail.gmail.com> Message-ID: you need additional libs... editline-devel or something similar. alternatively, you can remove them in the Makefile -Ray On Nov 20, 2009, at 1:50 PM, Costa Zikalala wrote: > Thanks for quick responses guys, yes I was doing it under the php > directory. > I've now also installed php-devel, and am now getting this error: > > make[1]: Entering directory `/home/Costa/freeswitch-1.0.4/libs/esl/ > php' > g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lcrypt -lcrypt - > ledit -lncurses -lresolv -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt - > lm -lcrypt -o ESL.so -L. > /usr/bin/ld: cannot find -ledit > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > make[1]: Leaving directory `/home/Costa/freeswitch-1.0.4/libs/esl/php' > make: *** [phpmod] Error 2 > > > > > 2009/11/20 Brian West > I'm going to guess you did cd libs/esl/php then typed make.. move up > one dir first then type make phpmod.. but you seem to be missing all > the php dev headers. > > /b > > On Nov 20, 2009, at 11:51 AM, Costa Zikalala wrote: > > > I've been trying to make phpmod without any success. I've even tried > > to ./configure --with-php and it did't help. > > I've just upgraded to latest svn and am running FS on FC11. > > > > I keep getting this error message: > > > > make: php-config: Command not found > > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o > > esl_wrap.o > > esl_wrap.cpp:717:18: error: zend.h: No such file or directory > > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory > > esl_wrap.cpp:719:17: error: php.h: No such file or directory > > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory > > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or > > directory > > esl_wrap.cpp:980:17: error: esl.h: No such file or directory > > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory > > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope > > esl_wrap.cpp:788: error: ISO C++ forbids declaration of > > ?ZEND_RSRC_DTOR_FUNC? with no type > > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this > > scope > > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token > > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? > > declared void > > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > > esl_wrap.cpp:793: error: ?z? was not declared in this scope > > esl_wrap.cpp:793: error: expected primary-expression before ?void? > > esl_wrap.cpp:793: error: expected primary-expression before ?*? > token > > esl_wrap.cpp:793: error: ?type? was not declared in this scope > > esl_wrap.cpp:793: error: expected primary-expression before ?int? > > make: *** [esl_wrap.o] Error 1 > > > > Please help. > > Thanks > > Costa > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/e0683bde/attachment.html From stevendt at primrosebank.net Fri Nov 20 12:14:51 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Fri, 20 Nov 2009 20:14:51 -0000 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls Message-ID: <31F6B654EEB247F493DEA83DDF816753@bp1.ad.bp.com> Hi, I have just purchased an ATA (Pluscom SIP VoIP ATA, model VPA-11) to try to use a normal (analogue) cordless phone with FreeSwitch. I have got the ATA setup and talking to FreeSwitch, it has registered the right extension and can receive and pick-up incoming calls. However, I can't dial numbers successfully, I can get the dial tone and dial the numbers, but the target numbers (internal or external) are not recognised by FreeSwitch. I have pasted a dump in the pastebin, but I can't see anything that tells me what the problem might be. Can anyone suggest what the problem might be please ? Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/b384530c/attachment.html From costa.zikalala at gmail.com Fri Nov 20 13:43:32 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Fri, 20 Nov 2009 23:43:32 +0200 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> <59daa2cd0911201050x18e60941le605d0a40da9cf48@mail.gmail.com> Message-ID: <59daa2cd0911201343s15cf1d9dlc843c30e51b03582@mail.gmail.com> Thanks Ray, I installed libedit-devel and it worked like charm. Costa 2009/11/20 Raymond Chandler > you need additional libs... editline-devel or something similar. > alternatively, you can remove them in the Makefile > > -Ray > > > On Nov 20, 2009, at 1:50 PM, Costa Zikalala wrote: > > Thanks for quick responses guys, yes I was doing it under the php > directory. > I've now also installed php-devel, and am now getting this error: > > make[1]: Entering directory `/home/Costa/freeswitch-1.0.4/libs/esl/php' > g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lcrypt -lcrypt -ledit > -lncurses -lresolv -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt -lm -lcrypt -o > ESL.so -L. > /usr/bin/ld: cannot find -ledit > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > make[1]: Leaving directory `/home/Costa/freeswitch-1.0.4/libs/esl/php' > make: *** [phpmod] Error 2 > > > > > 2009/11/20 Brian West > >> I'm going to guess you did cd libs/esl/php then typed make.. move up >> one dir first then type make phpmod.. but you seem to be missing all >> the php dev headers. >> >> /b >> >> On Nov 20, 2009, at 11:51 AM, Costa Zikalala wrote: >> >> > I've been trying to make phpmod without any success. I've even tried >> > to ./configure --with-php and it did't help. >> > I've just upgraded to latest svn and am running FS on FC11. >> > >> > I keep getting this error message: >> > >> > make: php-config: Command not found >> > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o >> > esl_wrap.o >> > esl_wrap.cpp:717:18: error: zend.h: No such file or directory >> > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >> > esl_wrap.cpp:719:17: error: php.h: No such file or directory >> > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory >> > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or >> > directory >> > esl_wrap.cpp:980:17: error: esl.h: No such file or directory >> > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory >> > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope >> > esl_wrap.cpp:788: error: ISO C++ forbids declaration of >> > ?ZEND_RSRC_DTOR_FUNC? with no type >> > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this >> > scope >> > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token >> > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? >> > declared void >> > esl_wrap.cpp:793: error: ?zval? was not declared in this scope >> > esl_wrap.cpp:793: error: ?z? was not declared in this scope >> > esl_wrap.cpp:793: error: expected primary-expression before ?void? >> > esl_wrap.cpp:793: error: expected primary-expression before ?*? token >> > esl_wrap.cpp:793: error: ?type? was not declared in this scope >> > esl_wrap.cpp:793: error: expected primary-expression before ?int? >> > make: *** [esl_wrap.o] Error 1 >> > >> > Please help. >> > Thanks >> > Costa >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/9f5cbef2/attachment-0001.html From Russell.Mosemann at cune.org Fri Nov 20 14:38:21 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Fri, 20 Nov 2009 22:38:21 -0000 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls In-Reply-To: <31F6B654EEB247F493DEA83DDF816753@bp1.ad.bp.com> Message-ID: <20091120223822.23FEF3E941A@mail.cune.org> Dave Stevenson said: > I have pasted a dump in the pastebin, URL? -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From stevendt at primrosebank.net Fri Nov 20 14:46:03 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Fri, 20 Nov 2009 22:46:03 -0000 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls References: <20091120223822.23FEF3E941A@mail.cune.org> Message-ID: <02EE75E8713B4A1BBA3BBEB03B91AF7D@bp1.ad.bp.com> Ooops - sorry about that ! OK, here you go .... http://pastebin.freeswitch.org/11205 I'd really appreciate some help with this as I'm really struggling. I think that the right tones are being sent as I can transfer a Voice Mail call to the phone and activate the voice prompts correctly. I have entered a blank dialplan in the ATA which should let all numbers be processed. FreeSwitch just won't play ball though ! Regards Dave ----- Original Message ----- From: To: Sent: Friday, November 20, 2009 10:38 PM Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out,but can receive calls > Dave Stevenson said: >> I have pasted a dump in the pastebin, > > URL? > > -- > Russell Mosemann > > > > ________________________________________________________ > Concordia University, Nebraska > See http://www.cune.edu/ for the latest news and events! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From malay.thakershi at continuityhealth.com Fri Nov 20 15:18:44 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Fri, 20 Nov 2009 17:18:44 -0600 Subject: [Freeswitch-users] mod_flite sound profiles Message-ID: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> Hello all, I am not able to play any female sound in mod_flite. I did try setting all 4 voice types one by one but it only says male voice. And the voice quality is not good at all. Am I doing something wrong? mObjMainSession.Answer(); mObjMainSession.sleep(1000, 0); //set tts engine mObjMainSession.SetTtsParameters("flite", "awb"); //Introduction message mObjMainSession.Speak("Hello... Welcome to phone system for assessment"); mObjMainSession.sleep(1000, 0); Also, can someone tell me what is the best way to get TTS going with good quality? Thank you for help. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/c414cb22/attachment.html From james at talent.com.au Fri Nov 20 15:19:59 2009 From: james at talent.com.au (James Budge) Date: Sat, 21 Nov 2009 09:19:59 +1000 Subject: [Freeswitch-users] OS X compile error Message-ID: make[6]: *** [mod_amr.so] Error 1 make[5]: *** [all] Error 1 make[4]: *** [mod_amr-all] Error 1 make[3]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[2]: *** [all-recursive] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 OS X 10.6.2 Xcode 3.2.1 FS Updated to revision 15582 From brian at freeswitch.org Fri Nov 20 15:32:27 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 17:32:27 -0600 Subject: [Freeswitch-users] OS X compile error In-Reply-To: References: Message-ID: You have left out the whole bits that say what exactly failed... look UP farther. btw don't make -j http://wiki.freeswitch.org/wiki/Installation_Guide#64-bit_Mac_OS_X_.28Snow_Leopard.29 /b On Nov 20, 2009, at 5:19 PM, James Budge wrote: > make[6]: *** [mod_amr.so] Error 1 > make[5]: *** [all] Error 1 > make[4]: *** [mod_amr-all] Error 1 > make[3]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + make install + > +----------------------------------------------+ > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all] Error 2 > make: *** [current] Error 2 > > > OS X 10.6.2 > Xcode 3.2.1 > FS Updated to revision 15582 From brian at freeswitch.org Fri Nov 20 15:33:00 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 17:33:00 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> Message-ID: <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> You pay top dollar for it. The free stuff just isn't as good as what you PAY good money for. I don't expect that to change anytime soon. /b On Nov 20, 2009, at 5:18 PM, Malay Thakershi wrote: > Also, can someone tell me what is the best way to get TTS going with > good quality? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/5aba3035/attachment.html From jason at jasonjgw.net Fri Nov 20 16:02:02 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 21 Nov 2009 11:02:02 +1100 Subject: [Freeswitch-users] RTP issues (possibly nat-related) In-Reply-To: References: <20091120011542.GA20754@jdc.jasonjgw.net> <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> <20091120070137.GA28316@jdc.jasonjgw.net> Message-ID: <20091121000202.GA21139@jdc.jasonjgw.net> I can still reproduce this as of rev. 15584. Symptom: 1. I called a test number via my ISP (IPv4, subject to nat). This worked. 2. I placed a second call to the same number 30 seconds later - connected, but no audio received. From mike at jerris.com Fri Nov 20 16:23:02 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 20 Nov 2009 19:23:02 -0500 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <925C296B-2582-4A38-9E43-A4FBF9B9224E@freeswitch.org> References: <4B06CD46.6050408@3gnt.net> <6b65470d0911200919y34546bvc93f3a5c976c1dc7@mail.gmail.com> <4B06D266.7020109@3gnt.net> <925C296B-2582-4A38-9E43-A4FBF9B9224E@freeswitch.org> Message-ID: <5D94EAF2-B446-4E0B-99D0-C7C4FC39456C@jerris.com> This was merged into trunk. On Nov 20, 2009, at 12:34 PM, Brian West wrote: > Hope on IRC and talk to MikeJ in #freeswitch he can direct you better > on what to do vs not do since he maintains the builds system in > FreeSWITCH. > > /b > > On Nov 20, 2009, at 11:31 AM, Igor Neves wrote: > >> Ok, >> >> But how should I proceed? >> >> Thanks, > From mike at jerris.com Fri Nov 20 16:28:57 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 20 Nov 2009 19:28:57 -0500 Subject: [Freeswitch-users] change event value In-Reply-To: <4B04E766.8070706@savion.huji.ac.il> References: <4B04E766.8070706@savion.huji.ac.il> Message-ID: no. On Nov 19, 2009, at 1:36 AM, Eli Hayun wrote: > Hi > Is there is a way to intercept an event (for example : REGISTER) and > change one of its parameters (for example: the extension number) and > fire up the corrected event? > > I need it to set the speedial of the phone value to be "**xxxxx" but to > make it register as "xxxxx" From brian at freeswitch.org Fri Nov 20 16:48:53 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 18:48:53 -0600 Subject: [Freeswitch-users] RTP issues (possibly nat-related) In-Reply-To: <20091121000202.GA21139@jdc.jasonjgw.net> References: <20091120011542.GA20754@jdc.jasonjgw.net> <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> <20091120070137.GA28316@jdc.jasonjgw.net> <20091121000202.GA21139@jdc.jasonjgw.net> Message-ID: <7163EBBF-F043-4C63-88BC-0A1F47F5906E@freeswitch.org> Can you give me some console logs and sip traces... maybe an rtp pcap? Thanks, Brian On Nov 20, 2009, at 6:02 PM, Jason White wrote: > I can still reproduce this as of rev. 15584. > > Symptom: > > 1. I called a test number via my ISP (IPv4, subject to nat). This > worked. > > 2. I placed a second call to the same number 30 seconds later - > connected, but > no audio received. From Prometheus001 at gmx.net Fri Nov 20 16:56:45 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 21 Nov 2009 01:56:45 +0100 Subject: [Freeswitch-users] Problems with Voicemail Message-ID: <4B073ACD.1090708@gmx.net> Hello, i have a couple of problems with voicemail. Voicemails are recorded but not played in any way. 1) when I call my voicemail, I can hear the number of new messages, but I canot not hear the recorded files itself. I hear the following * "You have 1 urgent new message in forder inbox" * "You have 7 new messages in forder inbox" * "New message number 1 Jan 011970 at 1 am" * (message is NOT played) * "You have 1 urgent new message in forder inbox" * "You have 7 new messages in forder inbox" * "press 1 to listen, press 2...." * (I press 1) * "New message number 1 Jan 011970 at 1 am" * (message is NOT played) * "You have 1 urgent new message in forder inbox" * "You have 7 new messages in forder inbox" * ... The voicemail files are stored in the file system as wav files and I can play them manually from the file system - so there is sound inside. 2) Another strange thing is that all recorded calls are announced with a date of 01.Jan.1970 although the databse shows correct values. 3) Alternatively playing it on the web Gui on http://fs.ip:8080/api/voicemail/web doesn't work either. Date is again 01.Jan.1970 and shown length of the file is always 00:00:00, although the database shows the correct number of seconds 4) Just to note that whenever I expect a recorded file to be played I see the following on the console 2009-11-21 00:17:02.511110 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA In my installation Freeswitch is running in a cluster and voicemails are stored in a mysql database. read_epoch is always 0, so file seems that Freeswitch never reads and updates an entry. Grepping mysql however shows a number of queries against the database and also the filenames are correctly read (output of ngrep): select * from voicemail_msgs where username='200' and domain='sip11.mydomain.com' and read_epoch=0 order by read_flags, created_epoch 1258748304.0.200.sip11.mydomain.com$db2801c4-d611-11de-8c58-554df1d6d322.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_b0fbf9e6-d611-11de-8c58-554df1d6d322.wav.15..A_URGENT 1258746833.0.200.sip11.mydomain.com$6e486a2e-d60e-11de-bb97-eb22f15930a0.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_50e727c2-d60e-11de-bb97-eb22f15930a0.wav.7..B_NORMAL 1258748679.0.200.sip11.mydomain.com$bac4dd0c-d612-11de-9618-afbc82bc409a.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_9e7865c4-d612-11de-9618-afbc82bc409a.wav.13..B_NORMAL 1258749095.0.200.sip11.mydomain.com$b2376082-d613-11de-80e8-89d0ee29138d.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_a6caaefc-d613-11de-80e8-89d0ee29138d.wav.6..B_NORMAL 1258749417.0.200.sip11.mydomain.com$726b375c-d614-11de-bb4c-6d51cf20cc23.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_6777907a-d614-11de-bb4c-6d51cf20cc23.wav.5..B_NORMAL 1258750260.0.200.sip11.mydomain.com$68cecedc-d616-11de-b8c8-69b0064d633e.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_5b6afb6c-d616-11de-b8c8-69b0064d633e.wav.9..B_NORMAL 1258753767.0.200.sip11.mydomain.com$93657422-d61e-11de-b8c8-69b0064d633e.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_84c1c588-d61e-11de-b8c8-69b0064d633e.wav.10..B_NORMAL Here's the debug log: EXECUTE sofia/internal/200 at sip1.mydomain.com send_display(VM 200) 2009-11-20 23:16:36.392353 [DEBUG] mod_dptools.c:703 sofia/internal/200 at sip1.mydomain.com receive message [DISPLAY] EXECUTE sofia/internal/200 at sip1.mydomain.com voicemail(check default sip11.mydomain.com 200) 2009-11-20 23:16:36.392353 [DEBUG] mod_voicemail.c:799 [default] rwlock 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-hello.wav] (en:en) 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:36.392353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:37.612349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:37.712349 [DEBUG] switch_channel.c:182 sofia/internal/200 at sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:37.812353 [DEBUG] switch_channel.c:182 sofia/internal/200 at sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:37.942376 [DEBUG] switch_channel.c:182 sofia/internal/200 at sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:38.062368 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:38.612348 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:38.712354 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1] (en:en) 2009-11-20 23:16:38.712354 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:38.712354 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:39.172350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:39.272353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-urgent-new.wav] (en:en) 2009-11-20 23:16:39.272353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:39.272353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:40.052349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:40.152353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-message.wav] (en:en) 2009-11-20 23:16:40.152353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:40.152353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:40.732349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:40.832349 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:16:40.832349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:40.832349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:42.012350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:42.112350 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:42.112350 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:42.112350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:42.112350 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:42.672354 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:42.772362 [DEBUG] switch_ivr_play_say.c:273 Handle say:[7] (en:en) 2009-11-20 23:16:42.772362 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:42.772362 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:43.312349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:43.412353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-new.wav] (en:en) 2009-11-20 23:16:43.412353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:43.412353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:43.752353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:43.872362 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-messages.wav] (en:en) 2009-11-20 23:16:43.872362 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:43.872362 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:44.532350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:44.652368 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:16:44.652368 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:44.652368 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:45.832349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:45.932353 [DEBUG] switch_channel.c:182 sofia/internal/200 at sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:46.062365 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:46.062365 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-new.wav] (en:en) 2009-11-20 23:16:46.062365 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:46.062365 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:46.392349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:46.492349 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-message_number.wav] (en:en) 2009-11-20 23:16:46.492349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:46.492349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:47.312349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:47.412349 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1] (en:en) 2009-11-20 23:16:47.412349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:47.412349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:47.872372 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:47.992349 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:47.992349 [DEBUG] switch_ivr_play_say.c:273 Handle say:[25] (en:en) 2009-11-20 23:16:47.992349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:47.992349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:48.632353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:48.632353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:48.632353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:49.152353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:49.152353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:49.152353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:49.612349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:49.612349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:49.612349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:50.092349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:50.092349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:50.092349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:50.532349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:50.532349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:50.532349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:51.012350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:51.012350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:51.012350 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:51.572349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:51.572349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:51.572349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:51.852353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:51.852353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:51.852353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:52.312350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:52.312350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:52.312350 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:52.892356 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:52.892356 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:52.892356 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:53.472353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA 2009-11-20 23:16:53.602359 [DEBUG] switch_channel.c:182 sofia/internal/200 at sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:53.722367 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:53.722367 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:53.722367 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:53.722367 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:54.272349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:54.372353 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1] (en:en) 2009-11-20 23:16:54.372353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:54.372353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:54.832441 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:54.943764 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-urgent-new.wav] (en:en) 2009-11-20 23:16:54.943764 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:54.943764 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:55.732350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:55.847095 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-message.wav] (en:en) 2009-11-20 23:16:55.847095 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:55.847095 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:56.432351 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:56.532354 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:16:56.532354 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:56.532354 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:57.712350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:57.812353 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:57.812353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:57.812353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:57.812353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:58.372350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:58.472356 [DEBUG] switch_ivr_play_say.c:273 Handle say:[7] (en:en) 2009-11-20 23:16:58.472356 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:58.472356 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:59.012350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:59.112350 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-new.wav] (en:en) 2009-11-20 23:16:59.112350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:59.112350 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:59.452351 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:59.572350 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-messages.wav] (en:en) 2009-11-20 23:16:59.572350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:59.572350 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:17:00.232350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:17:00.332371 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:17:00.332371 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:17:00.332371 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/f850a9b4/attachment-0001.html From brian at freeswitch.org Fri Nov 20 17:08:06 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 19:08:06 -0600 Subject: [Freeswitch-users] Problems with Voicemail In-Reply-To: <4B073ACD.1090708@gmx.net> References: <4B073ACD.1090708@gmx.net> Message-ID: <976A0342-4F4B-4035-9201-D56F8625AE12@freeswitch.org> I'm going to venture to guess maybe the file was recorded in a different codec and NOT pcma? /b On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: > 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / > usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA From brian at freeswitch.org Fri Nov 20 17:06:01 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 19:06:01 -0600 Subject: [Freeswitch-users] Problems with Voicemail In-Reply-To: <4B073ACD.1090708@gmx.net> References: <4B073ACD.1090708@gmx.net> Message-ID: This should give you some sort of clue. /b On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: > 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / > usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA From mike at jerris.com Fri Nov 20 17:17:23 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 20 Nov 2009 20:17:23 -0500 Subject: [Freeswitch-users] Media got stuck after attended transfer... In-Reply-To: References: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com> <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com> <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> Message-ID: I think a better approach here is to use spandsp. We already have some groundwork done for this. If you are interested in contributing, please email consulting at freeswitch.org and we can discuss further. Mike On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: > Hi, > > one of my customers is willing to contribute for t38 integration. > > The basic idea is to connect HylaFAX to FS: > t38modem <-> FreeSWITCH <-> Media Gateway with t38 support > All this without media proxy. > > Another idea might be to implement t38 origination/termination with a class 1 modem input/output for use with HylaFAX. > > Do you know how much money we need to collect for t38 support? > How much time is needed for implementing this? > > Thanks, Klaus > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: Friday, October 16, 2009 2:10 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Media got stuck after attended transfer... > > > > On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga wrote: > hi, any clue when can t38 be added? > > > "Eventually." :) Of course, if we could get more to add to the bounty it might grease the wheels of innovation. > > http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch > > Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) > > -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/9b0c7e8c/attachment.html From stevendt at primrosebank.net Fri Nov 20 17:20:47 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Sat, 21 Nov 2009 01:20:47 -0000 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Issue Closed References: <20091120223822.23FEF3E941A@mail.cune.org> <02EE75E8713B4A1BBA3BBEB03B91AF7D@bp1.ad.bp.com> Message-ID: <7449BC3B2C05406DAE264F5A11BAF110@bp1.ad.bp.com> OK, Just to close this one out, I've just spent some time on IRC and Michael Collins very quickly helped me get this sorted. I am using a cheap ATA which has a couple of "issues". I should probably have included it in the original post, but the ATA is a :- Pluscom SIP VoIP ATA - Model VPA-11. Quote from the ATA Manual ....... "If a default dial plan string is not required, the Default Dial Plan String field on the General configuration page (Section 4.1.2) can be left empty, in which case the default dial pattern to accept all dialed digits will be incorporated. The default dial pattern, [0-9*]>#[0-9*].e[0-9*].ft4, is transparent to the user and will not be displayed on the General configuration page" I had already (or so I thought) deleted the default ATA dialplan as I suspected that it was causing problems. As it turns out though, even with an apparently blank dialplan configured, the ATA was inserting some characters after the dialed number which were confusing the number handling in FreeSwitch. Michael quickly spotted the problem from my Pastebin dump and took about 10 seconds to come up with a fix ! Adding the following section to the top of the default FreeSwitch dialplan strips these extra characters off the string that FreeSwitch sees To: Sent: Friday, November 20, 2009 10:46 PM Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out,but can receive calls > Ooops - sorry about that ! > > OK, here you go .... > > http://pastebin.freeswitch.org/11205 > > I'd really appreciate some help with this as I'm really struggling. > > I think that the right tones are being sent as I can transfer a Voice Mail > call to the phone and activate the voice prompts correctly. > > I have entered a blank dialplan in the ATA which should let all numbers be > processed. > > FreeSwitch just won't play ball though ! > > Regards > Dave > > > > ----- Original Message ----- > From: > To: > Sent: Friday, November 20, 2009 10:38 PM > Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial > Out,but can receive calls > > >> Dave Stevenson said: >>> I have pasted a dump in the pastebin, >> >> URL? >> >> -- >> Russell Mosemann >> >> >> >> ________________________________________________________ >> Concordia University, Nebraska >> See http://www.cune.edu/ for the latest news and events! >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Fri Nov 20 19:34:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Nov 2009 21:34:05 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> Message-ID: <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> results cant possibly be the same there is not even any broadcast involved in uuid_transfer ? you need to attach a console trace with debug log up On Fri, Nov 20, 2009 at 10:41 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Finally got a chance to test this, the results are the same. > > Why am I getting this? Is it because I'm executing ring_ready before > I attempt the bridge? Is it because I'm using a socket? > > On Wed, Nov 11, 2009 at 10:14 PM, Anthony Minessale > wrote: > > dont execute bridge that way, your bridge itself is the other thing > already > > broadcasting. > > > > > > api uuid_transfer bridge:sofia/myprofile/foo at bar.cominline > > > > if you want to do more after the bridge > > set the variable park_after_bridge=true to make it go back to idle > > > > > > On Wed, Nov 11, 2009 at 4:33 PM, Kristian Kielhofner > > wrote: > >> > >> Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64. > >> > >> On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner > >> wrote: > >> > From the trace: > >> > > >> ..snip.. > >> > >> -- > >> Kristian Kielhofner > >> http://www.astlinux.org > >> http://blog.krisk.org > >> http://www.star2star.com > >> http://www.submityoursip.com > >> http://www.voalte.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/3adcc0e9/attachment-0001.html From thangappan143 at gmail.com Sat Nov 21 01:22:27 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Sat, 21 Nov 2009 14:52:27 +0530 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback Message-ID: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have implemented using Perl language. I did the following steps: => Set the playback_delimiter variable => Set the playback_sleep_val variable => Set the event lock as true => Set the freeswitch ( my own) variable as zero => Wait in the loop until the variable is been set as zero => Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) => Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played successfully). => Wait in the loop until the variable is been set as one. => Set the Event lock as false => Trying to get the DTMF digits ( Have a assurance that all the voice files are played). The problem is, The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* *NOTE*: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. Please help me................? Thanks in advance. -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/4c9e3b93/attachment.html From shiyanov at gmail.com Sat Nov 21 01:58:29 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Sat, 21 Nov 2009 12:58:29 +0300 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> References: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> Message-ID: Anthony, >>As soon as you call uuid_bridge you are transferring both legs of the call to bridge to each other. >>This means your java app must exit so the channels can connect to each other. I didn't know that. Now my java app is exiting upon the onHangup() call so everything has become "ok". Thank you much. I'll add note to the wiki about this issue. Artem On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Your "annoying behaviour" is the exact behavior you should be getting > considering what you told FS to do. > > As soon as you call uuid_bridge you are transferring both legs of the call > to bridge to each other. > This means your java app must exit so the channels can connect to each > other. > > remember that you hangup hook can be called when the channel is transferred > not only when it hangs up. > you have to test which is happening based on the input to your callback. > > > On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: > >> Hi there! >> >> I've got annoying FS behavior: >> There are 2 channels executing the same Java application (application >> itself is an IVR). If I try to bridge them with uuid_bridged then both >> channels are killed. Here is a log from FS console: >> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 >> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> >> CS_HIBERNATE >> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing >> file >> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing >> file >> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal >> sofia/internal/1005 at 192.168.147.130 [BREAK] >> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 >> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change >> CS_EXECUTE -> CS_HIBERNATE >> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: >> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >> >> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] >> switch_core_session.c:933 Send signal >> sofia/internal/1001 at master.agent.starpoundtec >> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal >> sofia/internal/1005 at 192.168.147.130 [BREAK] >> >> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream >> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal >> sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] >> >> (FS version is 1.0.4) >> >> Any thoughts? >> >> >> Artem >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/78ead067/attachment.html From mike at yes.net.ua Sat Nov 21 03:41:06 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Sat, 21 Nov 2009 13:41:06 +0200 Subject: [Freeswitch-users] Using odbc in FS core Message-ID: <1382216794.20091121134106@yes.net.ua> Hello Folks, I'm interesting in completely moving away from sqlite and use postgresql everywhere including core ( switch_core.c ) All other applications can use odbc without issues (sofia, limit, fifo etc), but as I see in core only sqlite3 supported. I correctly set 'core-db-dsn' parameter, but looks like the problem that latest psqlodbc_08_04_0100 don't support multiple statements in one request that is often used in switch_core_sqldb.c: > sql = switch_mprintf( > "update channels set uuid='%q' where uuid='%q' and hostname='%q';" > "update calls set caller_uuid='%q' where caller_uuid='%q' and hostname='%q';" > "update calls set callee_uuid='%q' where callee_uuid='%q' and hostname='%q'", > switch_event_get_header_nil(event, "unique-id"), > ... SKIP ... So, does anyone have any clue how to us postgresql in the FS core? Thanks. -- Mike Tkachuk From mike at yes.net.ua Sat Nov 21 04:02:07 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Sat, 21 Nov 2009 14:02:07 +0200 Subject: [Freeswitch-users] Using odbc in FS core In-Reply-To: <1382216794.20091121134106@yes.net.ua> References: <1382216794.20091121134106@yes.net.ua> Message-ID: <1013085378.20091121140207@yes.net.ua> Hello, Looks like the issue is not in multi statements in one request. Manually creating DB schema helped and everything started up. I will continue testing Also in code I see such construction: > switch_cache_db_execute_sql(dbh, "begin;delete from channels where hostname='';delete from channels where hostname='';commit;", &err); Anyone can explain why to do such delete twice and in transaction? Thanks. Saturday, November 21, 2009 1:41:06 PM, you wrote: MT> Hello Folks, MT> I'm interesting in completely moving away from sqlite and use MT> postgresql everywhere including core ( switch_core.c ) MT> All other applications can use odbc without issues (sofia, limit, MT> fifo etc), but as I see in core only sqlite3 supported. MT> I correctly set 'core-db-dsn' parameter, but looks like the problem MT> that latest psqlodbc_08_04_0100 don't support multiple statements in MT> one request that is often used in switch_core_sqldb.c: >> sql = switch_mprintf( >> "update channels set uuid='%q' where uuid='%q' and hostname='%q';" >> "update calls set caller_uuid='%q' where caller_uuid='%q' and hostname='%q';" >> "update calls set callee_uuid='%q' where callee_uuid='%q' and hostname='%q'", >> switch_event_get_header_nil(event, "unique-id"), >> ... SKIP ... MT> So, does anyone have any clue how to us postgresql in the FS core? MT> Thanks. MT> -- MT> Mike Tkachuk -- Mike Tkachuk From stevendt at primrosebank.net Sat Nov 21 04:10:23 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Sat, 21 Nov 2009 12:10:23 -0000 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Updated References: <20091120223822.23FEF3E941A@mail.cune.org><02EE75E8713B4A1BBA3BBEB03B91AF7D@bp1.ad.bp.com> <7449BC3B2C05406DAE264F5A11BAF110@bp1.ad.bp.com> Message-ID: <0E0B2CB55DD7426C91286784F4BF7353@bp1.ad.bp.com> Sorry for the extended forum thread on this subject - This really IS the last post ! I have now got the ATA to work without the dialplan fix provided by Michael. After I'd implemented the "fix", I had more of an idea of what the problem was and was better able to go through the Polycom VPA-11 setup screens through its web interface to see if there were any options that might have had a bearing on the problem. Under Configuration VoIP Non-Line Config General Parameters VoIP General There is an option to "Append UserId" - the Default Value is Yes. That was where the "extra characters" were coming from. Setting this option to No, makes the ATA behave more as expected, Regards Dave ----- Original Message ----- From: "Dave Stevenson" To: Sent: Saturday, November 21, 2009 1:20 AM Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out,but can receive calls - Issue Closed > OK, > > Just to close this one out, I've just spent some time on IRC and Michael > Collins very quickly helped me get this sorted. > > I am using a cheap ATA which has a couple of "issues". > > I should probably have included it in the original post, but the ATA is a > :- > > Pluscom SIP VoIP ATA - Model VPA-11. > > Quote from the ATA Manual ....... > > "If a default dial plan string is not required, the Default Dial Plan > String > field on the General configuration page (Section 4.1.2) > can be left empty, in which case the default dial pattern to accept all > dialed digits will be incorporated. > The default dial pattern, [0-9*]>#[0-9*].e[0-9*].ft4, is transparent to > the > user and will not be displayed on the General > configuration page" > > I had already (or so I thought) deleted the default ATA dialplan as I > suspected that it was causing problems. > > As it turns out though, even with an apparently blank dialplan configured, > the ATA was inserting some characters after the dialed number which were > confusing the number handling in FreeSwitch. > > Michael quickly spotted the problem from my Pastebin dump and took about > 10 > seconds to come up with a fix ! > > Adding the following section to the top of the default FreeSwitch dialplan > strips these extra characters off the string that FreeSwitch sees > > > > > > > > > Thanks a lot Michael ! > > regards > Dave > From Prometheus001 at gmx.net Sat Nov 21 04:14:17 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 21 Nov 2009 13:14:17 +0100 Subject: [Freeswitch-users] Problems with Voicemail In-Reply-To: <976A0342-4F4B-4035-9201-D56F8625AE12@freeswitch.org> References: <4B073ACD.1090708@gmx.net> <976A0342-4F4B-4035-9201-D56F8625AE12@freeswitch.org> Message-ID: <4B07D999.4040004@gmx.net> I installed all sounds from SVN, but usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA isn't there. I checked another, older installation and couldn't this file either. I think that freeswitch tries to build a sound path for the file to be played, and some parts of the path are missing. I expect it would play a recorded message at that time in /usr/local/freeswitch/storage/voicemail/default/${domain} and the defined format is "wav" not pcma. I also set "storage_dir" explicitely in the voicemail configs,but this also didn't help. Best regards Peter Brian West schrieb: > I'm going to venture to guess maybe the file was recorded in a > different codec and NOT pcma? > > /b > > On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: > > >> 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / >> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From siniypin at gmail.com Sat Nov 21 05:28:17 2009 From: siniypin at gmail.com (RobertT) Date: Sat, 21 Nov 2009 16:28:17 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> Message-ID: <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> Attached is graphical representation of SIP message flow. You can see that for some reason FS doesn't resend to callee an ACK message recieved from caller. Regards, RobertT -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/ab8948d5/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: TCP FS SIP msgs.PNG Type: image/png Size: 14515 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/ab8948d5/attachment.png From brian at freeswitch.org Sat Nov 21 07:46:08 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 21 Nov 2009 09:46:08 -0600 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> Message-ID: <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> Well since we aren't a proxy we wouldn't resend the one we receive... what svn rev and are you using proxy media? /b On Nov 21, 2009, at 7:28 AM, RobertT wrote: > Attached is graphical representation of SIP message flow. You can > see that for some reason FS doesn't resend to callee an ACK message > recieved from caller. > > Regards, RobertT From anthony.minessale at gmail.com Sat Nov 21 08:14:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 21 Nov 2009 10:14:59 -0600 Subject: [Freeswitch-users] Using odbc in FS core In-Reply-To: <1013085378.20091121140207@yes.net.ua> References: <1382216794.20091121134106@yes.net.ua> <1013085378.20091121140207@yes.net.ua> Message-ID: <191c3a030911210814l6e50b883uba61815fcd36afe1@mail.gmail.com> we had the code slightly out of order, you should update to latest trunk for the right version. The test of 2 deletes is to see if your odbc driver will fail when trying to execute 2 statements at once so I can properly fail over to sqlite because transactions are mandatory for a database running core in odbc. On Sat, Nov 21, 2009 at 6:02 AM, Mike Tkachuk wrote: > Hello, > > Looks like the issue is not in multi statements in one request. > Manually creating DB schema helped and everything started up. > I will continue testing > > Also in code I see such construction: > > switch_cache_db_execute_sql(dbh, "begin;delete from channels where > hostname='';delete from channels where hostname='';commit;", &err); > Anyone can explain why to do such delete twice and in transaction? > > Thanks. > > > > Saturday, November 21, 2009 1:41:06 PM, you wrote: > > MT> Hello Folks, > > MT> I'm interesting in completely moving away from sqlite and use > MT> postgresql everywhere including core ( switch_core.c ) > > MT> All other applications can use odbc without issues (sofia, limit, > MT> fifo etc), but as I see in core only sqlite3 supported. > > MT> I correctly set 'core-db-dsn' parameter, but looks like the problem > MT> that latest psqlodbc_08_04_0100 don't support multiple statements in > MT> one request that is often used in switch_core_sqldb.c: > > >> sql = switch_mprintf( > >> "update channels set uuid='%q' where uuid='%q' and hostname='%q';" > >> "update calls set caller_uuid='%q' where caller_uuid='%q' and > hostname='%q';" > >> "update calls set callee_uuid='%q' where callee_uuid='%q' and > hostname='%q'", > >> switch_event_get_header_nil(event, "unique-id"), > >> ... SKIP ... > > MT> So, does anyone have any clue how to us postgresql in the FS core? > > MT> Thanks. > > MT> -- > MT> Mike Tkachuk > > > > -- > Mike Tkachuk > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/f31802d3/attachment-0001.html From anthony.minessale at gmail.com Sat Nov 21 08:34:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 21 Nov 2009 10:34:21 -0600 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> Message-ID: <191c3a030911210834o6c134ec0v8a57df04b946f8cf@mail.gmail.com> cant you use the execute_complete events to tell when your playback is done or var is set? On Sat, Nov 21, 2009 at 3:22 AM, Thangappan.M wrote: > Dear all, > > I am in the process of implementing IVR using event outbound > socket (async mode). > I have implemented using Perl language. > > I did the following steps: > => Set the playback_delimiter variable > => Set the playback_sleep_val variable > => Set the event lock as true > => Set the freeswitch ( my own) variable as zero > => Wait in the loop until the variable is been set as > zero > => Playback the voice files ( Here I combined the voice > files with the delimiter value if more than one voice files are there) > => Set the freeswitch(my own) variable as true ( This is > used to identify whether the voice files are played > successfully). > => Wait in the loop until the variable is been set as > one. > => Set the Event lock as false > > => Trying to get the DTMF digits ( Have a assurance > that all the voice files are played). > > The problem is, > > The above steps are working fine when the voice file count is > lesser than or equal to 10. After the voice files are played only the > variable(my own freeswitch) is set. Based on the variable I am doing further > things. > > But when I tried to give the voice files count of more than 10 > the variable has been set while starting to play back the first voice file > itself . Because of this I am not able to proceed further. > > *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* > > *NOTE*: I also referred mod_file_string documentation. In that they > specified 128 files can be used to play back the voice files using > playback_delimiter option. > > Please help me................? > Thanks in advance. > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/3d7dff3c/attachment.html From abeka at greatiam.com Sat Nov 21 14:15:37 2009 From: abeka at greatiam.com (Sam Abekah-Mensah) Date: Sat, 21 Nov 2009 22:15:37 +0000 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway Message-ID: <4B086689.6080804@greatiam.com> I need help as I cannot receive calls through VOIPUSER. This is a learning setup Attached are my conf files. What is wrong with them ? When I dial from a landline I get a continuous beep. Attached are my gateway and the conf file to transfer. Sopfia Status is my screen message. I can see a FAIL and cannot make head or tail of all that message. Hopefully anyone using voipuser or in fact any of you clever folks can make sense of this. Thanks for your time. -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: sofia status.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/06405ed7/attachment.txt -------------- next part -------------- A non-text attachment was scrubbed... Name: voipuser.xml Type: text/xml Size: 300 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/06405ed7/attachment.xml -------------- next part -------------- A non-text attachment was scrubbed... Name: voipuser_org.xml Type: text/xml Size: 271 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/06405ed7/attachment-0001.xml From siniypin at gmail.com Sat Nov 21 15:23:05 2009 From: siniypin at gmail.com (RobertT) Date: Sun, 22 Nov 2009 02:23:05 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> Message-ID: <2160023e0911211523k7998d048nced3af8fb805e770@mail.gmail.com> Yep, I use proxy media. First it started with 1.0.4 release, then I've updated a week or two ago with the latest svn trunk, not sure what was the rev number. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/124d5003/attachment.html From msc at freeswitch.org Sat Nov 21 16:51:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Sat, 21 Nov 2009 16:51:54 -0800 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Updated In-Reply-To: <0E0B2CB55DD7426C91286784F4BF7353@bp1.ad.bp.com> References: <20091120223822.23FEF3E941A@mail.cune.org> <02EE75E8713B4A1BBA3BBEB03B91AF7D@bp1.ad.bp.com> <7449BC3B2C05406DAE264F5A11BAF110@bp1.ad.bp.com> <0E0B2CB55DD7426C91286784F4BF7353@bp1.ad.bp.com> Message-ID: <87f2f3b90911211651j6f0d5fd1raab7a77805bcfb56@mail.gmail.com> On Sat, Nov 21, 2009 at 4:10 AM, Dave Stevenson wrote: > Sorry for the extended forum thread on this subject - This really IS the > last post ! > > I have now got the ATA to work without the dialplan fix provided by > Michael. > > After I'd implemented the "fix", I had more of an idea of what the problem > was and was better able to go through the Polycom VPA-11 setup screens > through its web interface to see if there were any options that might have > had a bearing on the problem. > > Under Configuration > VoIP > Non-Line Config > General Parameters > VoIP General > > There is an option to "Append UserId" - the Default Value is Yes. > > That was where the "extra characters" were coming from. > > Setting this option to No, makes the ATA behave more as expected, > > Regards > Dave > > Nice work! I can understand why it was easier to look for this AFTER we did the band-aid solution and not before. :) You next task is to visit http://wiki.freeswitch.org and sign up. Please add the setup details for this device. I already created stub to get you going: http://wiki.freeswitch.org/wiki/Interop_List#Pluscom Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/00cd2e85/attachment.html From dfansler at dv-fansler.com Sat Nov 21 18:57:18 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Sat, 21 Nov 2009 18:57:18 -0800 (PST) Subject: [Freeswitch-users] IP1001 Setup Message-ID: <24395241.1258858638513.JavaMail.root@whwamui-deputy.pas.sa.earthlink.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/877e3d61/attachment-0001.html From mcampbellsmith at gmail.com Sat Nov 21 23:35:24 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 22 Nov 2009 18:35:24 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS Message-ID: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> HI All, Has anyone got some recommendations on which ATA to buy that supports TLS and SRTP? Thanks! From yehavi.bourvine at gmail.com Sat Nov 21 23:39:19 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 22 Nov 2009 09:39:19 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: <193640CC-3E62-4248-8E80-CE7FE82653C0@jerris.com> References: <4B0266F4.8070602@savion.huji.ac.il> <4B0387F1.7070105@savion.huji.ac.il> <193640CC-3E62-4248-8E80-CE7FE82653C0@jerris.com> Message-ID: Thanks Mike! However, this doesn't fully solve my problem. When using sofia_contact() indeed it works ok with finding the destination's profile. However, it breaks the BLFs... When calling *sofia/sip_profile/local-user%local-do**main* the BLF works ok. When calling sofia_contact(*sofia/sip_profile/local-user at local-domain*) BLF doesn't work (nothing is sent to the watching phone). Any more clues??? Thanks! __Yehavi: 2009/11/20 Michael Jerris > check out sofia_contact function. If you use this in combination with > binding profiles together so they are one table I think this should work > right. > > Mike > > On Nov 18, 2009, at 12:36 AM, Eli Hayun wrote: > > > Brian West wrote: > >> > >> Why do you need to know the destination profile like that? You get to > >> pick that on your own so you should already know that before hand. > >> > >> > >> /b > >> > >> On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote: > >> > >> > >>> Hi > >>> We have more then one profile. To make a call I have to enter : bridge > >>> sofia/profile/number at ip > >>> The problem is when I use : "${use_profile}" I am getting the caller > >>> profile, and I need the destination profile. > >>> > >>> How do I get this information? > >>> > >> > > Thanks for your answer. > > > > The problem is when I call to that number that the phone hook to other > server, I cannot make the call. > > Is there is a variable that can tell me the destination profile? > > Lets say the other profile called "ph1" I have to dial > > sofia/ph1/xxxxx at host to make the call. Is there other way to do that? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/596028b4/attachment.html From itamar at ispbrasil.com.br Sun Nov 22 00:20:04 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Sun, 22 Nov 2009 06:20:04 -0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> Message-ID: sipura/linksys look in ebay. On Sun, Nov 22, 2009 at 5:35 AM, Mark Campbell-Smith wrote: > HI All, > > Has anyone got some recommendations on which ATA to buy that supports > TLS and SRTP? > > Thanks! -- ------------ Itamar Reis Peixoto e-mail/msn/google talk/sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From mcampbellsmith at gmail.com Sun Nov 22 01:21:34 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 22 Nov 2009 20:21:34 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> Message-ID: <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> Do LInksys devices support TLS and SRTP that FS supports? 3102 at least doesn't according to this post http://osdir.com/ml/telephony.freeswitch.user/2008-08/msg00904.html On Sun, Nov 22, 2009 at 7:20 PM, Itamar Reis Peixoto wrote: > sipura/linksys > > look in ebay. > > > On Sun, Nov 22, 2009 at 5:35 AM, Mark Campbell-Smith > wrote: >> HI All, >> >> Has anyone got some recommendations on which ATA to buy that supports >> TLS and SRTP? >> >> Thanks! > > > > > -- > ------------ > > Itamar Reis Peixoto > > e-mail/msn/google talk/sip: itamar at ispbrasil.com.br > skype: itamarjp > icq: 81053601 > +55 11 4063 5033 > +55 34 3221 8599 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From itamar at ispbrasil.com.br Sun Nov 22 01:41:01 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Sun, 22 Nov 2009 07:41:01 -0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> Message-ID: it's support SRTP On Sun, Nov 22, 2009 at 7:21 AM, Mark Campbell-Smith wrote: > Do LInksys devices support TLS and SRTP that FS supports? ?3102 at > least doesn't according to this post -- ------------ Itamar Reis Peixoto e-mail/msn/google talk/sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From lon at kickasspixels.com Sun Nov 22 03:25:40 2009 From: lon at kickasspixels.com (Lon Baker) Date: Sun, 22 Nov 2009 03:25:40 -0800 Subject: [Freeswitch-users] Clarification about channel variables please. Message-ID: <5d3e0dc60911220325i69f663b0meeff47c551be6999@mail.gmail.com> Are either global or regular channel variable mutable during a call? Or can they only be set before and after? Any clarification would help, since the existing wiki doesn't make it clear. Lon From michal.bielicki at halo2.pl Sun Nov 22 04:06:09 2009 From: michal.bielicki at halo2.pl (Michal Bielicki) Date: Sun, 22 Nov 2009 13:06:09 +0100 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: <4B086689.6080804@greatiam.com> References: <4B086689.6080804@greatiam.com> Message-ID: Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah: > > I need help as I cannot receive calls through VOIPUSER. This is a learning setup Attached are my conf files. What is wrong with them ? When I dial from a landline I get a continuous beep. > > Attached are my gateway and the conf file to transfer. Sopfia Status is my screen message. I can see a FAIL and cannot make head or tail of all that message. Hopefully anyone using voipuser or in fact any of you clever folks can make sense of this. > > Thanks for your time. > > 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy RFC2833 Mode! > 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] > 2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] > 2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec sofia/external/nobody at 213.166.5.133 PCMA/8000 20 ms 160 samples > 2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101 > 2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 (sofia/external/nobody at 213.166.5.133) State Change CS_NEW -> CS_INIT > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send signal sofia/external/nobody at 213.166.5.133 [BREAK] > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 (sofia/external/nobody at 213.166.5.133) Running State Change CS_INIT > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 (sofia/external/nobody at 213.166.5.133) State INIT > 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 sofia/external/nobody at 213.166.5.133 SOFIA INIT > 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 (sofia/external/nobody at 213.166.5.133) State Change CS_INIT -> CS_ROUTING > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send signal sofia/external/nobody at 213.166.5.133 [BREAK] > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 (sofia/external/nobody at 213.166.5.133) State INIT going to sleep > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 (sofia/external/nobody at 213.166.5.133) Running State Change CS_ROUTING > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 (sofia/external/nobody at 213.166.5.133) State ROUTING > 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 sofia/external/nobody at 213.166.5.133 SOFIA ROUTING > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 sofia/external/nobody at 213.166.5.133 Standard ROUTING > 2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing anonymous->abeka in context public > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->unloop] continue=false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->outside_call] continue=true > Dialplan: sofia/external/nobody at 213.166.5.133 Absolute Condition [outside_call] > Dialplan: sofia/external/nobody at 213.166.5.133 Action set(outside_call=true) > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->call_debug] continue=true > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->public_extensions] continue=false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/ break=on-false > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->public_did] continue=false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->sip at sip.voipuser.org] continue=false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [sip at sip.voipuser.org] destination_number(abeka) =~ /08715042951/ break=on-false > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->Inbound-abeka at sip.voipuser.org]] continue=false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [Inbound-abeka at sip.voipuser.org]] destination_number(abeka) =~ /[08444846450]/ break=on-false > 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 (sofia/external/nobody at 213.166.5.133) State Change CS_ROUTING -> CS_EXECUTE > 2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send signal sofia/external/nobody at 213.166.5.133 [BREAK] > 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 (sofia/external/nobody at 213.166.5.133) State ROUTING going to sleep > 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 (sofia/external/nobody at 213.166.5.133) Running State Change CS_EXECUTE > 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 (sofia/external/nobody at 213.166.5.133) State EXECUTE > 2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 sofia/external/nobody at 213.166.5.133 SOFIA EXECUTE > 2009-11-21 22:07:15.706658 [DEBUG] switch_core_state_machine.c:151 sofia/external/nobody at 213.166.5.133 Standard EXECUTE > EXECUTE sofia/external/nobody at 213.166.5.133 set(outside_call=true) > 2009-11-21 22:07:15.728613 [DEBUG] mod_dptools.c:748 sofia/external/nobody at 213.166.5.133 SET [outside_call]=[true] > 2009-11-21 22:07:15.728613 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/external/nobody at 213.166.5.133 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-11-21 22:07:15.728613 [DEBUG] switch_channel.c:1683 Send signal sofia/external/nobody at 213.166.5.133 [KILL] > 2009-11-21 22:07:15.728613 [DEBUG] switch_core_session.c:932 Send signal sofia/external/nobody at 213.166.5.133 [BREAK] > 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:491 (sofia/external/nobody at 213.166.5.133) State EXECUTE going to sleep > 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:398 (sofia/external/nobody at 213.166.5.133) Running State Change CS_HANGUP > 2009-11-21 22:07:15.735830 [DEBUG] switch_core_state_machine.c:434 (sofia/external/nobody at 213.166.5.133) State HANGUP > 2009-11-21 22:07:15.735830 [DEBUG] mod_sofia.c:338 Channel sofia/external/nobody at 213.166.5.133 hanging up, cause: NORMAL_CLEARING > 2009-11-21 22:07:15.737680 [DEBUG] mod_sofia.c:417 Responding to INVITE with: 480 > 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:46 sofia/external/nobody at 213.166.5.133 Standard HANGUP, cause: NORMAL_CLEARING > 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:434 (sofia/external/nobody at 213.166.5.133) State HANGUP going to sleep > 2009-11-21 22:07:15.742930 [DEBUG] switch_core_state_machine.c:476 (sofia/external/nobody at 213.166.5.133) State Change CS_HANGUP -> CS_REPORTING > 2009-11-21 22:07:15.742930 [DEBUG] switch_core_session.c:932 Send signal sofia/external/nobody at 213.166.5.133 [BREAK] > 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:398 (sofia/external/nobody at 213.166.5.133) Running State Change CS_REPORTING > 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:612 (sofia/external/nobody at 213.166.5.133) State REPORTING > 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:53 sofia/external/nobody at 213.166.5.133 Standard REPORTING, cause: NORMAL_CLEARING > 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:612 (sofia/external/nobody at 213.166.5.133) State REPORTING going to sleep > 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:411 (sofia/external/nobody at 213.166.5.133) State Change CS_REPORTING -> CS_DESTROY > 2009-11-21 22:07:15.800497 [DEBUG] switch_core_session.c:1068 Session 2 (sofia/external/nobody at 213.166.5.133) Locked, Waiting on external entities > 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/external/nobody at 213.166.5.133) Ended > 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/nobody at 213.166.5.133 [CS_DESTROY] > 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 (sofia/external/nobody at 213.166.5.133) State DESTROY > 2009-11-21 22:07:15.802636 [DEBUG] mod_sofia.c:255 sofia/external/nobody at 213.166.5.133 SOFIA DESTROY > 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:60 sofia/external/nobody at 213.166.5.133 Standard DESTROY > 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 (sofia/external/nobody at 213.166.5.133) State DESTROY going to sleep > > > > > > > > > > > > > > > > > > > > : you seem to have not specified an extension where the call should go to my voipuser.org setup looks like: I am also surprised that your setup works with a from-domain of sip.voipuser.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Michal Bielicki HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa t. +48228753290 | f. +48228753291 michal.bielicki at halokwadrat.pl | w. www.halokwadrat.pl Knowledge & Low Prices. Guaranteed! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/01cc6d3a/attachment-0001.html From tculjaga at gmail.com Sun Nov 22 04:15:24 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sun, 22 Nov 2009 13:15:24 +0100 Subject: [Freeswitch-users] Media got stuck after attended transfer... In-Reply-To: References: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com> <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com> <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> Message-ID: <65d96fc80911220415v70d0bafbvad56c4fcb4576d8b@mail.gmail.com> it is better to enhance mod_fax with t.38 support... we have done sometihng and it is close to be work... T. On Sat, Nov 21, 2009 at 2:17 AM, Michael Jerris wrote: > I think a better approach here is to use spandsp. We already have some > groundwork done for this. If you are interested in contributing, please > email consulting at freeswitch.org and we can discuss further. > > Mike > > On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: > > Hi, > > one of my customers is willing to contribute for t38 integration. > > The basic idea is to connect HylaFAX to FS: > t38modem <-> FreeSWITCH <-> Media Gateway with t38 support > All this without media proxy. > > Another idea might be to implement t38 origination/termination with a class > 1 modem input/output for use with HylaFAX. > > Do you know how much money we need to collect for t38 support? > How much time is needed for implementing this? > > Thanks, Klaus > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, October 16, 2009 2:10 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Media got stuck after attended > transfer... > > > > On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga > wrote: > > hi, any clue when can t38 be added? > > "Eventually." :) Of course, if we could get more to add to the bounty it > might grease the wheels of innovation. > > > http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch > > Of course, I was listening to my A.M radio the other day and they said that > there was this new invention called the Internet that would let people send > documents to each other electronically. Maybe you should look into that. > Next thing you know they'll come up with telephones that people don't have > to plug into the wall and can take with them in the car. ;) > > -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/1b995dfa/attachment.html From Mailings at kh-dev.de Sun Nov 22 05:00:06 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 22 Nov 2009 14:00:06 +0100 Subject: [Freeswitch-users] Media got stuck after attended transfer... In-Reply-To: <65d96fc80911220415v70d0bafbvad56c4fcb4576d8b@mail.gmail.com> References: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com> <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com> <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> <65d96fc80911220415v70d0bafbvad56c4fcb4576d8b@mail.gmail.com> Message-ID: For "only" sending and receiving that's true. But my customer wants 2 things: - Using HylaFAX as fax server, as there are a lot of client apps and other tools - Connecting "real" fax machines using a Linksys/Cisco SPA2102 (as this is certified by their SIP/ISDN gateway vendor) So I could really need t38 handling in FS to don't make things more complicated as they already are... J Proxy mode doesn't work for me because it gives an error when resume-media-on-hold is set. Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Sunday, November 22, 2009 1:15 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Media got stuck after attended transfer... it is better to enhance mod_fax with t.38 support... we have done sometihng and it is close to be work... T. On Sat, Nov 21, 2009 at 2:17 AM, Michael Jerris > wrote: I think a better approach here is to use spandsp. We already have some groundwork done for this. If you are interested in contributing, please email consulting at freeswitch.org and we can discuss further. Mike On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: Hi, one of my customers is willing to contribute for t38 integration. The basic idea is to connect HylaFAX to FS: t38modem <-> FreeSWITCH <-> Media Gateway with t38 support All this without media proxy. Another idea might be to implement t38 origination/termination with a class 1 modem input/output for use with HylaFAX. Do you know how much money we need to collect for t38 support? How much time is needed for implementing this? Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, October 16, 2009 2:10 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Media got stuck after attended transfer... On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga > wrote: hi, any clue when can t38 be added? "Eventually." :) Of course, if we could get more to add to the bounty it might grease the wheels of innovation. http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/70a2ab76/attachment-0001.html From dfansler at dv-fansler.com Sun Nov 22 06:09:26 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Sun, 22 Nov 2009 09:09:26 -0500 (GMT-05:00) Subject: [Freeswitch-users] IP0010 SIP Phone Message-ID: <10222602.1258898966890.JavaMail.root@whwamui-deputy.pas.sa.earthlink.net> After the help of a couple of people from this list, I now have FreeSWITCH running - yeah! I have installed X-Lite on a couple of computers and they dial each other, play music on hold, etc. I have not yet connected to the outside world. I purchased an IP-0010 phone off eBay ($20 including shipping - docs at http://www.vanaccess.com/news/news_images/2007131_73_User%20Manual%20-%20IP0010.pdf) I cannot get this phone to work with the system. It gets an IP address, time/date, and a dial tone. After many tries with the http congifuration tool, I got the phone "configured" with the address of the SIP server, and a SIP User ID. When you dial an extension the FreeSWITCH window shows the following: sofica.c3844 Hanugup sofia/internal/101 at 192.168.1.165 [CS_NEW] [INCOMPATIBLE_DESTINATION] switch_core_session.c1139 Session 20 (sofia/internal/101 at 192.165.1.65) Ended switch_core_session.c1141 Close Channel sofia/internal/1001 at 192.168.1.165 [CS_DESTROY] Has anyone else tried this phone, or does anyone have suggestions I could try. I have looked through the website but have not found anything to help. Thanks, David David V. Fansler S/V Annabelle www.dv-fansler.com dfansler at dv-fansler.com From abeka at greatiam.com Sun Nov 22 09:53:13 2009 From: abeka at greatiam.com (Sam Abekah-Mensah) Date: Sun, 22 Nov 2009 17:53:13 +0000 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: References: <4B086689.6080804@greatiam.com> Message-ID: <4B097A89.2050400@greatiam.com> Hi Michael Thanks I had set it to send incoming calls to extension 1001. This is in the file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory. The contents are : Is there anything wrong with this please ? Thanks Michal Bielicki wrote: > > Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah: > >> >> I need help as I cannot receive calls through VOIPUSER. This is a >> learning setup Attached are my conf files. What is wrong with them ? >> When I dial from a landline I get a continuous beep. >> >> Attached are my gateway and the conf file to transfer. Sopfia Status >> is my screen message. I can see a FAIL and cannot make head or tail >> of all that message. Hopefully anyone using voipuser or in fact any >> of you clever folks can make sense of this. >> >> Thanks for your time. >> >> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy >> RFC2833 Mode! >> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec >> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >> 2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec >> Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] >> 2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec >> sofia/external/nobody at 213.166.5.133 PCMA/8000 20 ms 160 samples >> 2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf >> payload to 101 >> 2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 >> (sofia/external/nobody at 213.166.5.133) State Change CS_NEW -> CS_INIT >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/nobody at 213.166.5.133 [BREAK] >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/nobody at 213.166.5.133) Running State Change CS_INIT >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >> (sofia/external/nobody at 213.166.5.133) State INIT >> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 >> sofia/external/nobody at 213.166.5.133 SOFIA INIT >> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 >> (sofia/external/nobody at 213.166.5.133) State Change CS_INIT -> CS_ROUTING >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/nobody at 213.166.5.133 [BREAK] >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >> (sofia/external/nobody at 213.166.5.133) State INIT going to sleep >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/nobody at 213.166.5.133) Running State Change CS_ROUTING >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 >> (sofia/external/nobody at 213.166.5.133) State ROUTING >> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 >> sofia/external/nobody at 213.166.5.133 SOFIA ROUTING >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 >> sofia/external/nobody at 213.166.5.133 Standard ROUTING >> 2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing >> anonymous->abeka in context public >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->unloop] continue=false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->outside_call] continue=true >> Dialplan: sofia/external/nobody at 213.166.5.133 Absolute Condition >> [outside_call] >> Dialplan: sofia/external/nobody at 213.166.5.133 Action >> set(outside_call=true) >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->call_debug] continue=true >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->public_extensions] continue=false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >> [public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/ >> break=on-false >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->public_did] continue=false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >> [public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->sip at sip.voipuser.org] continue=false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >> [sip at sip.voipuser.org] destination_number(abeka) =~ /08715042951/ >> break=on-false >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->Inbound-abeka at sip.voipuser.org]] continue=false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >> [Inbound-abeka at sip.voipuser.org]] destination_number(abeka) =~ >> /[08444846450]/ break=on-false >> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 >> (sofia/external/nobody at 213.166.5.133) State Change CS_ROUTING -> >> CS_EXECUTE >> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/nobody at 213.166.5.133 [BREAK] >> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 >> (sofia/external/nobody at 213.166.5.133) State ROUTING going to sleep >> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/nobody at 213.166.5.133) Running State Change CS_EXECUTE >> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 >> (sofia/external/nobody at 213.166.5.133) State EXECUTE >> 2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 >> sofia/external/nobody at 213.166.5.133 SOFIA EXECUTE >> 2009-11-21 22:07:15.706658 [DEBUG] switch_core_state_machine.c:151 >> sofia/external/nobody at 213.166.5.133 Standard EXECUTE >> EXECUTE sofia/external/nobody at 213.166.5.133 set(outside_call=true) >> 2009-11-21 22:07:15.728613 [DEBUG] mod_dptools.c:748 >> sofia/external/nobody at 213.166.5.133 SET [outside_call]=[true] >> 2009-11-21 22:07:15.728613 [NOTICE] switch_core_state_machine.c:179 >> Hangup sofia/external/nobody at 213.166.5.133 [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-11-21 22:07:15.728613 [DEBUG] switch_channel.c:1683 Send signal >> sofia/external/nobody at 213.166.5.133 [KILL] >> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/nobody at 213.166.5.133 [BREAK] >> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:491 >> (sofia/external/nobody at 213.166.5.133) State EXECUTE going to sleep >> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/nobody at 213.166.5.133) Running State Change CS_HANGUP >> 2009-11-21 22:07:15.735830 [DEBUG] switch_core_state_machine.c:434 >> (sofia/external/nobody at 213.166.5.133) State HANGUP >> 2009-11-21 22:07:15.735830 [DEBUG] mod_sofia.c:338 Channel >> sofia/external/nobody at 213.166.5.133 hanging up, cause: NORMAL_CLEARING >> 2009-11-21 22:07:15.737680 [DEBUG] mod_sofia.c:417 Responding to >> INVITE with: 480 >> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:46 >> sofia/external/nobody at 213.166.5.133 Standard HANGUP, cause: >> NORMAL_CLEARING >> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:434 >> (sofia/external/nobody at 213.166.5.133) State HANGUP going to sleep >> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_state_machine.c:476 >> (sofia/external/nobody at 213.166.5.133) State Change CS_HANGUP -> >> CS_REPORTING >> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/nobody at 213.166.5.133 [BREAK] >> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/nobody at 213.166.5.133) Running State Change CS_REPORTING >> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:612 >> (sofia/external/nobody at 213.166.5.133) State REPORTING >> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:53 >> sofia/external/nobody at 213.166.5.133 Standard REPORTING, cause: >> NORMAL_CLEARING >> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:612 >> (sofia/external/nobody at 213.166.5.133) State REPORTING going to sleep >> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:411 >> (sofia/external/nobody at 213.166.5.133) State Change CS_REPORTING -> >> CS_DESTROY >> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_session.c:1068 Session >> 2 (sofia/external/nobody at 213.166.5.133) Locked, Waiting on external >> entities >> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1086 >> Session 2 (sofia/external/nobody at 213.166.5.133) Ended >> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1088 Close >> Channel sofia/external/nobody at 213.166.5.133 [CS_DESTROY] >> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >> (sofia/external/nobody at 213.166.5.133) State DESTROY >> 2009-11-21 22:07:15.802636 [DEBUG] mod_sofia.c:255 >> sofia/external/nobody at 213.166.5.133 SOFIA DESTROY >> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:60 >> sofia/external/nobody at 213.166.5.133 Standard DESTROY >> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >> (sofia/external/nobody at 213.166.5.133) State DESTROY going to sleep >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> : > > > you seem to have not specified an extension where the call should go to > > my voipuser.org setup looks like: > > > > > > > > > > > > > > > > I am also surprised that your setup works with a from-domain of > sip.voipuser.org > >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > *Michal Bielicki* > HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa > t. +48228753290 | f. +48228753291 > michal.bielicki at halokwadrat.pl > | > w. www.halokwadrat.pl > > > > *Knowledge & Low Prices. Guaranteed!* > From JCasale at activenetwerx.com Sun Nov 22 10:02:18 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Sun, 22 Nov 2009 18:02:18 +0000 Subject: [Freeswitch-users] Call quality problem w/ Snom M3's Message-ID: Not sure where to start with this one, the outgoing leg to our sip provider sounds perfectly fine but with the our M3's the incoming leg is super choppy. Using twinkle on my laptop yields good results in both directions so there must be an issue with just the snoms, their firmware was very old but updating it didn't help. Our sip provider uses ulaw and the snoms and vars.xml both have ulaw set as top pref. Any ideas what to look at next? Thanks, jlc From achaloyan at yahoo.com Sun Nov 22 10:02:48 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Sun, 22 Nov 2009 10:02:48 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <1258732768082-4038514.post@n2.nabble.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> Message-ID: <552708.67071.qm@web111314.mail.gq1.yahoo.com> We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org ________________________________ From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/a8f51f90/attachment.html From mike at jerris.com Sun Nov 22 10:20:17 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 22 Nov 2009 13:20:17 -0500 Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <552708.67071.qm@web111314.mail.gq1.yahoo.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> Message-ID: Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan wrote: > We discussed build integration related issues a few months ago with > Mike and seemed to find a solution which would work for both UniMRCP > and FreeSWITCH source trees. > > Now I've just got a chance to look into this a bit closer trying to > further complete VS2008 build integration in FreeSWITCH. So I've got > it working, the module is not only being built, but also is getting > loaded. Current build integration is not as seamless as I want it to > be, but probably we can start with what we have now and then discuss > and identify what can be done in the future. This concerns not only > build integration but overall integrity. > > So would you be interested in the patch? Where should I upload it? > I thought I had a Jira account, but not sure it exists any more. > > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org > > > From: Jeff Lenk > To: freeswitch-users at lists.freeswitch.org > Sent: Fri, November 20, 2009 7:59:28 PM > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch > & uniMRCP > > > That module is not currently being built for Windows. Also the library > unimrcp needs build integration work with FS to make that happen under > windows. > > > ss1 wrote: > > > > Hi Everyone, > > > > Please help freeswitch experts... !!! > > > > i have been working on freeswitch from last 2 days. i have > downloaded > > freeswitch and unimrcp (server + client) for windows. > > I tested the unimrcp client and server, which is running fine with > the > > command: run synth and run recog. I got both synth.pcm & recog.pcm > files. > > > > But my objective is to call Freeswitch through x-lite, where > freeswitch > > should call unimrcp client and return the PCM files. > > > > I tried it alot, but unable to do it. after lots of reading i > found that i > > do not have mod_unimrcp. i do not know from where to download it > and how > > to merge it into freeswitch. > > > > I would be very thankful if you may help. > > > > Thanks, > > ss > > > > > > -- > View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/f9dfb536/attachment-0001.html From mattdfong at gmail.com Sun Nov 22 10:48:50 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Mon, 23 Nov 2009 01:48:50 +0700 Subject: [Freeswitch-users] Recording with Native File PCMU Message-ID: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> I'm trying to conserve processor power by recording in native file format, PCMU in my case. It works great with the following line session:execute("record", "/tmp/my_recording."..session:getVariable("read_codec")); however it fails to work with session:execute("record_session", "/tmp/my_recording."..session:getVariable("read_codec")); or record = api:execute("sched_api", '+1 none uuid_record '..session:getVariable("uuid")..' start /tmp/my_recording.'..session:getVariable("read_codec")); Why is it that it works with record, but not with record_session or uuid_record? Is there something I'm over looking? In the latter two the consul reports 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File [/tmp/my_recording.PCMU] 8000hz as if it's recording, but /tmp/my_recording.PCMU never shows up. However if I change it to .wav instead of .PCMU it works. Any ideas? --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/6eac0a38/attachment.html From achaloyan at yahoo.com Sun Nov 22 11:03:36 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Sun, 22 Nov 2009 11:03:36 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> Message-ID: <634117.24120.qm@web111311.mail.gq1.yahoo.com> Mike, >Jira is the best, otherwise just mail me the patch and I'll take a look. I've uploaded the patch against svn trunk to http://jira.freeswitch.org/browse/MODUNIMRCP-6 it's made for win32 debug only yet. >Can you take a look at my last patch to the module to make it build please. I see. I've not noticed this change introduces API change, makes no sense to me now. I'll provide more convenient solution soon. Arsen. ________________________________ From: Michael Jerris To: "freeswitch-users at lists.freeswitch.org" Sent: Sun, November 22, 2009 10:20:17 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan wrote: We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. > >Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. > >So would you be interested in the patch? Where should I upload it? >I thought I had a Jira account, but not sure it exists any more. > >-- >Arsen Chaloyan >The author of > UniMRCP >http://www.unimrcp.org > > > > > ________________________________ From: Jeff Lenk >To: freeswitch-users at lists.freeswitch.org >Sent: Fri, November 20, 2009 7:59:28 PM >Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP > > >That module is not currently being built for Windows. Also the library >unimrcp needs build integration work with FS to make that happen under >windows. > > >ss1 wrote: >> >> Hi Everyone, >> >> Please help freeswitch experts... !!! >> >> i have been working on freeswitch from last 2 days. i have downloaded >> freeswitch and unimrcp (server + client) for windows. >> I tested the unimrcp client and server, which is running fine with the >> command: run synth and run recog. I got both synth.pcm & recog.pcm files. >> >> But my objective is to call Freeswitch through x-lite, where freeswitch >> should call unimrcp client and return the PCM files. >> >> I tried it alot, but unable to do it. after lots of reading i found that i >> do not have mod_unimrcp. i do not know from where to download it and how >> to merge it into freeswitch. >> >> I would > be very thankful if you may help. >> >> Thanks, >> ss >> >> > >-- >View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html >Sent from the freeswitch-users mailing list archive at Nabble.com. > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > _______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/1e431b0e/attachment.html From Prometheus001 at gmx.net Sun Nov 22 11:27:20 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 22 Nov 2009 20:27:20 +0100 Subject: [Freeswitch-users] Problems with Voicemail In-Reply-To: <4B07D999.4040004@gmx.net> References: <4B073ACD.1090708@gmx.net> <976A0342-4F4B-4035-9201-D56F8625AE12@freeswitch.org> <4B07D999.4040004@gmx.net> Message-ID: <4B099098.2040408@gmx.net> I now created a file inbox.PCMA and get the following: * inbox.PCMA is played * the recorded voive mail file is not played (FS does not even try to do that) * then I hear o "to listen to the recording press 1" o "to save the recording press 2" o ... Here's the debug output 2009-11-22 20:17:43.701098 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-22 20:17:44.278600 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-22 20:17:44.386776 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA] 8000hz 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-listen_to_recording.wav] (en:en) 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-22 20:17:45.201099 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-22 20:17:46.419933 [DEBUG] switch_ivr_play_say.c:1428 done playing file nGrepping port 3306 I can see that the correct filenames are retrieved from the mysql/odbc database: 1258894746.0.200.sip1.mydomain.com$d11c2a74-d766-11de-997b-bd7aecdc2a16.Gor Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_c57a5e84-d766-11de-997b-bd7aecdc2a16.wav.4..B_NORMAL.....47 1258897120.0.200.sip1.mydomain.com$580dafee-d76c-11de-84d4-a1cd7fa320b3.Gor Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_4d484a7e-d76c-11de-84d4-a1cd7fa320b3.wav.5..B_NORMAL......... Both filenames can be read. Best regards Peter Peter P GMX schrieb: > I installed all sounds from SVN, but > > usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA > > isn't there. I checked another, older installation and couldn't this > file either. > > I think that freeswitch tries to build a sound path for the file to be > played, and some parts of the path are missing. > I expect it would play a recorded message at that time in > /usr/local/freeswitch/storage/voicemail/default/${domain} and the > defined format is "wav" not pcma. > > I also set "storage_dir" explicitely in the voicemail configs,but this > also didn't help. > > Best regards > Peter > > > Brian West schrieb: > >> I'm going to venture to guess maybe the file was recorded in a >> different codec and NOT pcma? >> >> /b >> >> On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: >> >> >> >>> 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / >>> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From achaloyan at yahoo.com Sun Nov 22 11:36:59 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Sun, 22 Nov 2009 11:36:59 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <634117.24120.qm@web111311.mail.gq1.yahoo.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> <634117.24120.qm@web111311.mail.gq1.yahoo.com> Message-ID: <192864.88786.qm@web111315.mail.gq1.yahoo.com> Mike, upgrade UniMRCP to http://code.google.com/p/unimrcp/source/detail?r=1297 and remove that #if from mod_unimrcp. API is backward compatible now src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c =================================================================== --- src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c (revision 15605) +++ src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c (working copy) @@ -3510,11 +3510,7 @@ } /* Set up the media engine that will be shared with all profiles */ -#if UNI_VERSION_AT_LEAST(0,8,0) - media_engine = mpf_engine_create(1, pool); -#else media_engine = mpf_engine_create(pool); -#endif if (media_engine) { mrcp_client_media_engine_register(client, media_engine, "MediaEngine"); } Arsen ________________________________ From: Arsen Chaloyan To: freeswitch-users at lists.freeswitch.org Sent: Sun, November 22, 2009 11:03:36 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Mike, >Jira is the best, otherwise just mail me the patch and I'll take a look. I've uploaded the patch against svn trunk to http://jira.freeswitch.org/browse/MODUNIMRCP-6 it's made for win32 debug only yet. >Can you take a look at my last patch to the module to make it build please. I see. I've not noticed this change introduces API change, makes no sense to me now. I'll provide more convenient solution soon. Arsen. ________________________________ From: Michael Jerris To: "freeswitch-users at lists.freeswitch.org" Sent: Sun, November 22, 2009 10:20:17 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan wrote: We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. > >Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build > integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. > >So would you be interested in the patch? Where should I upload it? >I thought I had a Jira account, but not sure it exists any more. > >-- >Arsen Chaloyan >The author of > UniMRCP >http://www.unimrcp.org > > > > > ________________________________ From: Jeff Lenk >To: freeswitch-users at lists.freeswitch.org >Sent: Fri, > November 20, 2009 7:59:28 PM >Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP > > >That module is not currently being built for Windows. Also the library >unimrcp needs build integration work with FS to make that happen under >windows. > > >ss1 wrote: >> >> Hi Everyone, >> >> Please help freeswitch experts... !!! >> >> i have been working on freeswitch from last 2 days. i have downloaded >> freeswitch and unimrcp (server + client) for windows. >> I tested the unimrcp client and server, which is running fine with the >> command: run synth and run recog. I got both synth.pcm & recog.pcm files. >> >> But my objective is to call Freeswitch through x-lite, where freeswitch >> should call unimrcp client and return the PCM files. >> >> I tried it alot, but unable to do it. after lots of reading i found that i >> do not have mod_unimrcp. i do not know from where to download it and how >> to merge it into freeswitch. >> >> I would > be very thankful if you may help. >> >> Thanks, >> ss >> >> > >-- >View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html >Sent from the freeswitch-users mailing list archive at Nabble.com. > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > _______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/ea36e5f4/attachment-0001.html From timuckun at gmail.com Sun Nov 22 15:43:38 2009 From: timuckun at gmail.com (Tim Uckun) Date: Mon, 23 Nov 2009 12:43:38 +1300 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> References: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> Message-ID: <855e4dcf0911221543o222bef63t1c3340b0a41d57c1@mail.gmail.com> On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman wrote: > Hi Sam, > Take a look at mod_xml_curl. ?Pretty sure it'll do everything you're looking > for. Looking at that diagram it seems like mod_xml_curl makes a call for every SIP connection. That seems like overkill. Is there a way to set it up so that it caches the XML it got for a period of time? From dujinfang at gmail.com Sun Nov 22 15:54:11 2009 From: dujinfang at gmail.com (Seven Du) Date: Mon, 23 Nov 2009 07:54:11 +0800 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> Message-ID: <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> did you try without any .wav or .PCMU? 2009/11/23 Matthew Fong > I'm trying to conserve processor power by recording in native file format, > PCMU in my case. It works great with the following line > > session:execute("record", > "/tmp/my_recording."..session:getVariable("read_codec")); > > however it fails to work with > > session:execute("record_session", > "/tmp/my_recording."..session:getVariable("read_codec")); > or > record = api:execute("sched_api", '+1 none uuid_record > '..session:getVariable("uuid")..' start > /tmp/my_recording.'..session:getVariable("read_codec")); > > Why is it that it works with record, but not with record_session or > uuid_record? Is there something I'm over looking? In the latter two the > consul reports > > 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File > [/tmp/my_recording.PCMU] 8000hz > > as if it's recording, but /tmp/my_recording.PCMU never shows up. However if > I change it to .wav instead of .PCMU it works. Any ideas? > > --matt > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/019664f3/attachment.html From gurveshb at yahoo.com Sun Nov 22 01:15:22 2009 From: gurveshb at yahoo.com (Gurvesh Bhutiani) Date: Sun, 22 Nov 2009 01:15:22 -0800 (PST) Subject: [Freeswitch-users] Broadvoice 32 transcoding support? In-Reply-To: <87f2f3b90911200934n20373bc6tf01677ec8d2bb11d@mail.gmail.com> Message-ID: <58250.88935.qm@web35705.mail.mud.yahoo.com> Yes, the latest trunk works. Thank you! Gaurav --- On Fri, 11/20/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Broadvoice 32 transcoding support? > To: freeswitch-users at lists.freeswitch.org > Date: Friday, November 20, 2009, 9:34 AM > > > On Fri, Nov 20, 2009 at 1:16 AM, > Gaurav Singh > wrote: > > > Hi, > > Does freeswitch support transcoding between broadvoice > (BV32 ) and G711 > ? > Try latest trunk. There was a new update just added very > recently... > -MC > > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kdjakovic at hotmail.com Sun Nov 22 10:46:13 2009 From: kdjakovic at hotmail.com (katarina djakovic) Date: Sun, 22 Nov 2009 19:46:13 +0100 Subject: [Freeswitch-users] Problems with sighup and rotating csv files Message-ID: Hi, I am using the Freeswitch 1.0.4pre7. Great application, but I encountered a problem wich I can not solve since I am very new to it. Two things are happening. 1) The mod_cdr_csv.c (line 122 do_rotate()) does not always respond to sighup signal to rotate the cdr-csv files. Some times it happens and some times it does not. I can not see any pattern in the behaviour. Seems that sometimes functions in the mod_cdr_csv.c catch the signal and some times they do not. 2) Playing with the "kill -HUP fspid" all of a sudden I started getting two freeswitch processes in the process list. One being parent of another. Then, when I send the sighup signal to the parent - the console dies off and the other freeswitch process stays (leaving the comment "Hangup" in the fsconsole). Freeswitch conitnues to work with the remaining process. In case when I send the sighup to the child, it will rotate the log files. However, it always rotates the freeswitch.log, but randomly rotates the cdr-csv files. 3) I have a feeling that above behaviours are somehow connected, but do not understand how. Anyone can help? Any comment or idea will be very very much apreciated. Cheers, Katarina Windows Live: Make it easier for your friends to see what you?re up to on Facebook. _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/c112aef2/attachment.html From jlenk at frontiernet.net Sun Nov 22 20:16:28 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Sun, 22 Nov 2009 20:16:28 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <552708.67071.qm@web111314.mail.gq1.yahoo.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> Message-ID: <1258949788572-4048969.post@n2.nabble.com> Hi Arsen, I would be happy to help with the FS integration if you want - please do put your patch in a Jira. Jeff Date: Sun, 22 Nov 2009 10:09:41 -0800 From: ml-node+4047148-1118239605 at n2.nabble.com To: jlenk at frontiernet.net Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org From: Jeff Lenk <[hidden email]> To: [hidden email] Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org View message @ http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html To unsubscribe from Re: need help !! Problem with freeswitch & uniMRCP, click here. _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/ -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4048969.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/40949795/attachment.html From thangappan143 at gmail.com Sun Nov 22 20:34:01 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Mon, 23 Nov 2009 10:04:01 +0530 Subject: [Freeswitch-users] Fwd: Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> Message-ID: <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> I am waiting only for DTMF events. That's why I am setting freeswitch variable for knowing whether the playback has done. My question is "why this freeswitch variable is not setting properly when I play back more than 10 files using playback_delimiter option?". When I play back lesser than ten voice files the variable has been set properly. What could be the reason? ---------- Forwarded message ---------- From: Thangappan.M Date: Sat, Nov 21, 2009 at 2:52 PM Subject: Problem while playing more than 10 voice files using playback To: freeswitch-users Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have implemented using Perl language. I did the following steps: => Set the playback_delimiter variable => Set the playback_sleep_val variable => Set the event lock as true => Set the freeswitch ( my own) variable as zero => Wait in the loop until the variable is been set as zero => Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) => Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played successfully). => Wait in the loop until the variable is been set as one. => Set the Event lock as false => Trying to get the DTMF digits ( Have a assurance that all the voice files are played). The problem is, The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* *NOTE*: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. Please help me................? Thanks in advance. -- Regards, Thangappan.M -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/720ecbd7/attachment-0001.html From qinglan_zeng at hotmail.com Sun Nov 22 20:51:55 2009 From: qinglan_zeng at hotmail.com (=?gb2312?B?tPPE4MjL?=) Date: Mon, 23 Nov 2009 04:51:55 +0000 Subject: [Freeswitch-users] FS compile error under Windows: error LNK2019 In-Reply-To: References: Message-ID: All, I tried to compile FS source code under Windows while there are lots of errors: Error LNK2019, external _imp_sleep at 4 can not be resolved, this function was referred by _tMCRTStartup. Some other more similiar errors detail information attached. Any ideas? Thanks Daniel Zeng From: freeswitch-users-request at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 41, Issue 146 To: freeswitch-users at lists.freeswitch.org Date: Sun, 22 Nov 2009 11:37:32 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --??????-- From: mattdfong at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Mon, 23 Nov 2009 01:48:50 +0700 Subject: [Freeswitch-users] Recording with Native File PCMU I'm trying to conserve processor power by recording in native file format, PCMU in my case. It works great with the following line session:execute("record", "/tmp/my_recording."..session:getVariable("read_codec")); however it fails to work with session:execute("record_session", "/tmp/my_recording."..session:getVariable("read_codec")); or record = api:execute("sched_api", '+1 none uuid_record '..session:getVariable("uuid")..' start /tmp/my_recording.'..session:getVariable("read_codec")); Why is it that it works with record, but not with record_session or uuid_record? Is there something I'm over looking? In the latter two the consul reports 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File [/tmp/my_recording.PCMU] 8000hz as if it's recording, but /tmp/my_recording.PCMU never shows up. However if I change it to .wav instead of .PCMU it works. Any ideas? --matt --??????-- From: achaloyan at yahoo.com To: freeswitch-users at lists.freeswitch.org Date: Sun, 22 Nov 2009 11:03:36 -0800 Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Mike, >Jira is the best, otherwise just mail me the patch and I'll take a look. I've uploaded the patch against svn trunk to http://jira.freeswitch.org/browse/MODUNIMRCP-6 it's made for win32 debug only yet. >Can you take a look at my last patch to the module to make it build please. I see. I've not noticed this change introduces API change, makes no sense to me now. I'll provide more convenient solution soon. Arsen. From: Michael Jerris To: "freeswitch-users at lists.freeswitch.org" Sent: Sun, November 22, 2009 10:20:17 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan wrote: We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --??????-- From: Prometheus001 at gmx.net To: freeswitch-users at lists.freeswitch.org Date: Sun, 22 Nov 2009 20:27:20 +0100 Subject: Re: [Freeswitch-users] Problems with Voicemail I now created a file inbox.PCMA and get the following: * inbox.PCMA is played * the recorded voive mail file is not played (FS does not even try to do that) * then I hear o "to listen to the recording press 1" o "to save the recording press 2" o ... Here's the debug output 2009-11-22 20:17:43.701098 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-22 20:17:44.278600 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-22 20:17:44.386776 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA] 8000hz 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-listen_to_recording.wav] (en:en) 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-22 20:17:45.201099 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-22 20:17:46.419933 [DEBUG] switch_ivr_play_say.c:1428 done playing file nGrepping port 3306 I can see that the correct filenames are retrieved from the mysql/odbc database: 1258894746.0.200.sip1.mydomain.com$d11c2a74-d766-11de-997b-bd7aecdc2a16.Gor Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_c57a5e84-d766-11de-997b-bd7aecdc2a16.wav.4..B_NORMAL.....47 1258897120.0.200.sip1.mydomain.com$580dafee-d76c-11de-84d4-a1cd7fa320b3.Gor Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_4d484a7e-d76c-11de-84d4-a1cd7fa320b3.wav.5..B_NORMAL......... Both filenames can be read. Best regards Peter Peter P GMX schrieb: > I installed all sounds from SVN, but > > usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA > > isn't there. I checked another, older installation and couldn't this > file either. > > I think that freeswitch tries to build a sound path for the file to be > played, and some parts of the path are missing. > I expect it would play a recorded message at that time in > /usr/local/freeswitch/storage/voicemail/default/${domain} and the > defined format is "wav" not pcma. > > I also set "storage_dir" explicitely in the voicemail configs,but this > also didn't help. > > Best regards > Peter > > > Brian West schrieb: > >> I'm going to venture to guess maybe the file was recorded in a >> different codec and NOT pcma? >> >> /b >> >> On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: >> >> >> >>> 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / >>> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > --??????-- From: achaloyan at yahoo.com To: freeswitch-users at lists.freeswitch.org Date: Sun, 22 Nov 2009 11:36:59 -0800 Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Mike, upgrade UniMRCP to http://code.google.com/p/unimrcp/source/detail?r=1297 and remove that #if from mod_unimrcp. API is backward compatible now src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c =================================================================== --- src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c (revision 15605) +++ src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c (working copy) @@ -3510,11 +3510,7 @@ } /* Set up the media engine that will be shared with all profiles */ -#if UNI_VERSION_AT_LEAST(0,8,0) - media_engine = mpf_engine_create(1, pool); -#else media_engine = mpf_engine_create(pool); -#endif if (media_engine) { mrcp_client_media_engine_register(client, media_engine, "MediaEngine"); } Arsen From: Arsen Chaloyan To: freeswitch-users at lists.freeswitch.org Sent: Sun, November 22, 2009 11:03:36 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Mike, >Jira is the best, otherwise just mail me the patch and I'll take a look. I've uploaded the patch against svn trunk to http://jira.freeswitch.org/browse/MODUNIMRCP-6 it's made for win32 debug only yet. >Can you take a look at my last patch to the module to make it build please. I see. I've not noticed this change introduces API change, makes no sense to me now. I'll provide more convenient solution soon. Arsen. From: Michael Jerris To: "freeswitch-users at lists.freeswitch.org" Sent: Sun, November 22, 2009 10:20:17 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan wrote: We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ????????????? ????????????????! ????? _________________________________________________________________ MSN????????????????25???????????2010????????? http://kaba.msn.com.cn/?k=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/4fb7f97a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: compile_error-2.JPG Type: image/pjpeg Size: 115300 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/4fb7f97a/attachment-0001.bin From mike at jerris.com Sun Nov 22 21:12:12 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 00:12:12 -0500 Subject: [Freeswitch-users] FS compile error under Windows: error LNK2019 In-Reply-To: References: Message-ID: On Nov 22, 2009, at 11:51 PM, ??? wrote: > From keith at laaks.com Mon Nov 23 00:00:04 2009 From: keith at laaks.com (Keith Laaks) Date: Mon, 23 Nov 2009 10:00:04 +0200 Subject: [Freeswitch-users] Adding headers to INFO messages for Advice of Charge on SNOM Message-ID: <1258963204.4961.8.camel@keithl-lt> Hi, I have tried maintaining charging information on a SNOM 300's display using 'display' - but found that the phone has some timer, whereby every 60 seconds it wipes out whatever happens to be on the display at that time and replaces is with the dialled number. So not a viable option as it impacts usability. Really annoying when the display was just updated with valuable information for the user and a split second later it gets replaced. [If somebody knows how to disable this behaviour - please do tell...] I see that SNOM supports a number of features for Advice of Charge. >From their Wiki: http://wiki.snom.com/Advice_of_charge_%28AOC%29_in_SIP Example of an SIP-Info Message: ----------------------------------------------------- INFO sip:bla at snom.com SIP/2.0 From: ;tag=5354n3 To: ;tag=33rfh3 CSeq: 23423 INFO Call-ID: 3452tw43dt354dm03 AOC: charging;state=active; charging-info=currency; currency=EUR; amount=2000; multiplier=0.001 Content-Length: 0 ----------------------------------------------------- So the question - Is there some method available today to add these additional 'new' headers to an INFO message I can send out to these phones? If not, I guess it's a matter of looking at enhancing the "case SWITCH_MESSAGE_INDICATE_DISPLAY" section in mod_sofia.c ? Best Regards Keith From shiyanov at gmail.com Mon Nov 23 02:05:12 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Mon, 23 Nov 2009 13:05:12 +0300 Subject: [Freeswitch-users] Clarification about channel variables please. In-Reply-To: <5d3e0dc60911220325i69f663b0meeff47c551be6999@mail.gmail.com> References: <5d3e0dc60911220325i69f663b0meeff47c551be6999@mail.gmail.com> Message-ID: both types of variables are mutable On Sun, Nov 22, 2009 at 2:25 PM, Lon Baker wrote: > Are either global or regular channel variable mutable during a call? > Or can they only be set before and after? > > Any clarification would help, since the existing wiki doesn't make it > clear. > > Lon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/ca5a8b80/attachment.html From info at daccii.it Mon Nov 23 02:36:57 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Mon, 23 Nov 2009 11:36:57 +0100 Subject: [Freeswitch-users] User who answer the bridge in a execute_answer Message-ID: <4B0A65C9.10509@daccii.it> Hi, i'm writing some dialplan parts that get executed on execute_on_answer. In this dialplan that get executed i need to make a directory to handle recordings for record_session and my folder structure is: USER/YEAR/MONTH/HOUR-MINUTE-SECOND-CALLER_NUMBER.wav ------ ------ The call flow is: Call from external -> IVR -> Transfer to Group -> Execute on Answer -> system/bind_meta_app Pratically, i need the number (or better the user) that answered the call: what variable should i check? I tried with sip_from_user, callee_id_number and some other. Thank for your help, Best Regards, Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/eba245aa/attachment.vcf From lakindia89 at gmail.com Mon Nov 23 03:25:52 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 23 Nov 2009 16:55:52 +0530 Subject: [Freeswitch-users] Callback to the user in ESL Message-ID: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> Hi, I'm using perl ESL to control the call in freeswitch. I'm having the following scenario, but not able to get it right. Dialplan: 1. User A calls to an extention (1000). 2. My ESL program will be running, and it answers the call. 3. Then the program will get a number from the user. 4. It will hangup the call. 5. The program has to call to the number that was given by the user. In the above scenario, I was able to do until the 4th step. After hangup the call, if I say originate it is not working. Any ideas on how to do this in ESL. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/233cc5ba/attachment.html From oscav at hotmail.fr Mon Nov 23 03:45:17 2009 From: oscav at hotmail.fr (Oscav) Date: Mon, 23 Nov 2009 03:45:17 -0800 (PST) Subject: [Freeswitch-users] Execute on Answer with JavaScript Message-ID: <26476532.post@talk.nabble.com> Hi, How can we send the answer to the caller only when the callee answers, in JavaScript?? Many thanks. -- View this message in context: http://old.nabble.com/Execute-on-Answer-with-JavaScript-tp26476532p26476532.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at yes.net.ua Mon Nov 23 03:45:28 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Mon, 23 Nov 2009 13:45:28 +0200 Subject: [Freeswitch-users] Using odbc in FS core In-Reply-To: <191c3a030911210814l6e50b883uba61815fcd36afe1@mail.gmail.com> References: <1382216794.20091121134106@yes.net.ua> <1013085378.20091121140207@yes.net.ua> <191c3a030911210814l6e50b883uba61815fcd36afe1@mail.gmail.com> Message-ID: <1202092411.20091123134528@yes.net.ua> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/8c436c02/attachment.html From piotr_zurek at biprotech.com Mon Nov 23 03:52:57 2009 From: piotr_zurek at biprotech.com (=?UTF-8?B?UGlvdHIgxbt1cmVr?=) Date: Mon, 23 Nov 2009 12:52:57 +0100 Subject: [Freeswitch-users] [This is a repost. I'm not sure if my message was delivered.] How to pick up someone's phone remotely. In-Reply-To: <4AFC005A.4090200@biprotech.com> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> <4AFC005A.4090200@biprotech.com> Message-ID: <4B0A7799.6050500@biprotech.com> Hello again. This is a repost. I'm having difficulties communicating with this list (I'm getting reports from the list saying something about "excessive bounces"...), so I'm not sure anybody got this message. I'm trying to mimic behavior of my analogue PBX with FS. I want to be able to answer any incomming/transfered (from IVR or a person) call remotely, and to cancel the possibility of intercepting this call afterwards. Greetings Peter -- Original message -- My problems evolve, because I didn't know all these functions in FS are so much dependent on each other. But I'm learning fast... The scenario I written about before appears to be too much simplified version of what I need to achieve. In fact, below scenario and solution works OK only one time - when someone calls and there's no person on the called extension, and someone manually answers that phone on other extension. Then any other person can't intercept this call. Thats is correct and needed behavior. But if the same person who answered the phone transfers this call - everything goes back to normal and below solution does not work because the call has been answered already and execute_on_answer does not execute ever again during this call/channel. The same happens if there's IVR on the external extension answering calls and then forwarding to extensions - everyone can intercept last call even if it's already answered because IVR answers all call on start (and execute_on_answer doesn't get executed). So I think I need similar solution but working everywhere: on calls and transfers. Is there some variable or some other thing that I could set to block and unblock intercept when needed to get wanted behavior. Any hints? Greetings Peter Piotr Zurek pisze: > Thank You for such an elegant and simple solution that I have not > thought about. > With an exception that I'm using FS 1.0.4 right now and it appears > that something changed in time and following line should use hash > instead of db (when using default 1.0.4 FS config): > . > After a few hours of experimenting everything works as planned. > > Thank You very much. > Peter > > Ognjen Seslija pisze: >> Add the following: >> >> . >> >> after >> >> > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> >> in local extensions default example, or change it globally previously >> than this extension. You can join us on IRC if you can any more >> questions (sekil). >> >> Regards, >> Ognjen >> >> >> >> On Tue, Nov 10, 2009 at 4:01 PM, Piotr ?urek >> > wrote: >> >> Hello. >> >> Thank You developers for Freeswitch. >> I have installed it lately and it's working quite nicely, but I >> have one problem: >> >> I need to mimic behavior of my current analogue PBX installation >> using Freeswitch. >> >> This is the scenario: >> In the office with a few desks (extensions 1000-1010) and only >> one person behind one of desks (whatever extension - in example >> 1000). >> 1. There's incoming call on _one_ of extensions 1001-1010 >> 2. The person on extension 1000 wants to answer this call on his >> phone so dials #37 and this call is redirected to his phone. >> >> That's how it works on my office on analogue PBX system. Anyone >> can answer a call from any other phone as long as it hasn't been >> answered already. >> >> I tried to use the intercept action (with global example in >> default config) but it's not what I need because it intercepts >> the call even if it's already answered. I need to intercept all >> but only unanswered calls. I tried to use Redirect but it does >> not work on other's extensions call's (or does it?). >> >> Please help. >> Peter ?urek >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> -------------- next part -------------- A non-text attachment was scrubbed... Name: piotr_zurek.vcf Type: text/x-vcard Size: 414 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/63c139c3/attachment-0001.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3678 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/63c139c3/attachment-0001.bin From maciej.aniserowicz at gmail.com Mon Nov 23 04:00:39 2009 From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz) Date: Mon, 23 Nov 2009 04:00:39 -0800 (PST) Subject: [Freeswitch-users] Question about rtp-timeout-sec variable Message-ID: <1258977639954-4050650.post@n2.nabble.com> Hello, I have 2 instances of FS: one controlled by my application (making calls with TCP commands, recording sessions, listening to events etc) and one acting as a remote gateway to which all users register. When I leave the default values of rtp-timeout-sec and brutally kill x-lite during conversation, the 'hangup' event with 'media_timeout' cause is obviously sent after the default 5 minutes (and until then, the other leg is still connected to a 'dead' channel). The question is: which FS instance is responsible for terminating the connection after timeout? Only the 'remote' FS instance config seems to work. I thought that the shortest configured value should cause the timeout, but it's not the case. Am I missing something, or is this the correct behavior? Regards, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Question-about-rtp-timeout-sec-variable-tp4050650p4050650.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at yes.net.ua Mon Nov 23 04:11:31 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Mon, 23 Nov 2009 14:11:31 +0200 Subject: [Freeswitch-users] Using odbc in FS core In-Reply-To: <1202092411.20091123134528@yes.net.ua> References: <1382216794.20091121134106@yes.net.ua> <1013085378.20091121140207@yes.net.ua> <191c3a030911210814l6e50b883uba61815fcd36afe1@mail.gmail.com> <1202092411.20091123134528@yes.net.ua> Message-ID: <16025244.20091123141131@yes.net.ua> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/51d67b45/attachment.html From nameer.kazzaz at gmail.com Mon Nov 23 04:23:23 2009 From: nameer.kazzaz at gmail.com (Nameer Kazzaz) Date: Mon, 23 Nov 2009 12:23:23 +0000 Subject: [Freeswitch-users] SIP Digest nonce (stale="true") Message-ID: <4B0A7EBB.8040702@gmail.com> Hi Anthony, I'm having an issue with a gateway after the nonce-ttl expires we are sending stale="true", the cpe some how only likes stale=true without the "". I see on rev 15441 you made a change and marked it out. So my question is who is correct on this is it the CPE or are we sticking with the quoted ("true"). Thanks Nameer Kazzaz From Russell.Mosemann at cune.org Mon Nov 23 05:50:17 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Mon, 23 Nov 2009 13:50:17 -0000 Subject: [Freeswitch-users] [This is a repost. I'm not sure if my message was delivered.] How to pick up someone's phone remotely. In-Reply-To: <4B0A7799.6050500@biprotech.com> Message-ID: <20091123135017.D61CC43C11B@mail.cune.org> Piotr ??urek said: > This is a repost. I'm having difficulties communicating with this list=20 > (I'm getting reports from the list saying something about "excessive=20 > bounces"...), so I'm not sure anybody got this message. 1. http://lists.freeswitch.org/pipermail/freeswitch-users/ 2. Click the link for "Thread" for November 2009 3. Search for your topic -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From achaloyan at yahoo.com Mon Nov 23 06:33:12 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Mon, 23 Nov 2009 06:33:12 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <1258949788572-4048969.post@n2.nabble.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> <1258949788572-4048969.post@n2.nabble.com> Message-ID: <858430.90192.qm@web111301.mail.gq1.yahoo.com> Hi Jeff, Your input would be very helpful, I just wanted to understand where the problem is and contribute the way I can. I see you're the assignee, so please go ahead and let me know if there is anything left I can help with. Arsen. ________________________________ From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Mon, November 23, 2009 8:16:28 AM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Hi Arsen, I would be happy to help with the FS integration if you want - please do put your patch in a Jira. Jeff ________________________________ Date: Sun, 22 Nov 2009 10:09:41 -0800 From: [hidden email] To: [hidden email] Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org ________________________________ From: Jeff Lenk <[hidden email]> To: [hidden email] Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ View message @ http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html To unsubscribe from Re: need help !! Problem with freeswitch & uniMRCP, click here. ________________________________ Hotmail: Trusted email with powerful SPAM protection. Sign up now. ________________________________ View this message in context: RE: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/75fb2718/attachment-0001.html From brian at freeswitch.org Mon Nov 23 06:42:33 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 08:42:33 -0600 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> Message-ID: <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> If you're doing native file you DO NOT put an extension on the file name. /b On Nov 22, 2009, at 5:54 PM, Seven Du wrote: > did you try without any .wav or .PCMU? From abeka at greatiam.com Mon Nov 23 07:22:30 2009 From: abeka at greatiam.com (Otis) Date: Mon, 23 Nov 2009 15:22:30 +0000 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX Message-ID: <4B0AA8B6.2080305@greatiam.com> Hi Folks Is anyone using this on Fedora and is there a binary or installation script anywhere Thanks From brian at freeswitch.org Mon Nov 23 07:30:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 09:30:08 -0600 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <4B0AA8B6.2080305@greatiam.com> References: <4B0AA8B6.2080305@greatiam.com> Message-ID: cd /usr/src wget http://www.freeswitch.org/eg/Makefile make /b On Nov 23, 2009, at 9:22 AM, Otis wrote: > Hi Folks > > Is anyone using this on Fedora and is there a binary or installation > script anywhere > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Nov 23 07:52:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 09:52:03 -0600 Subject: [Freeswitch-users] SIP Digest nonce (stale="true") In-Reply-To: <4B0A7EBB.8040702@gmail.com> References: <4B0A7EBB.8040702@gmail.com> Message-ID: <191c3a030911230752r70b702b1g61694350f56b01e0@mail.gmail.com> The quoted true is the correct way from my research. The commented line was to test a device, a grandstream, they apparently do not accept it with quotes and I was using the unquoted version it to gather evidence to issue a bug report to them. They told me it will be fixed in the next firmware, was this the brand of device you have as well? On Mon, Nov 23, 2009 at 6:23 AM, Nameer Kazzaz wrote: > Hi Anthony, > I'm having an issue with a gateway after the nonce-ttl expires we > are sending stale="true", the cpe some how only likes stale=true without > the "". I see on rev 15441 > < > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_reg.c?r=15441#l687 > > > you made a change and marked it out. So my question is who is correct on > this is it the CPE or are we sticking with the quoted ("true"). > > Thanks > Nameer Kazzaz > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/912e243c/attachment.html From anthony.minessale at gmail.com Mon Nov 23 07:54:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 09:54:50 -0600 Subject: [Freeswitch-users] Fwd: Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> Message-ID: <191c3a030911230754i1d863180q694292e08beb7e44@mail.gmail.com> Maybe it is a race condition, I can't tell you from just such a basic description the code is complicated and I would have to reproduce it myself, but I can tell you one more time for good measure that you should use execute_complete events to tell when a command you tried to execute has finished and not poll the channel for a variable to be set because FreeSWITCH is an asynchronous application in the mode you are describing and you can never be sure of the timing. On Sun, Nov 22, 2009 at 10:34 PM, Thangappan.M wrote: > I am waiting only for DTMF events. That's why I am setting freeswitch > variable for knowing whether the playback has done. > > My question is "why this freeswitch variable is not setting properly when I > play back more than 10 files using playback_delimiter option?". > > When I play back lesser than ten voice files the variable has been set > properly. What could be the reason? > > > > ---------- Forwarded message ---------- > From: Thangappan.M > Date: Sat, Nov 21, 2009 at 2:52 PM > Subject: Problem while playing more than 10 voice files using playback > To: freeswitch-users > > > Dear all, > > I am in the process of implementing IVR using event outbound > socket (async mode). > I have implemented using Perl language. > > I did the following steps: > => Set the playback_delimiter variable > => Set the playback_sleep_val variable > => Set the event lock as true > => Set the freeswitch ( my own) variable as zero > => Wait in the loop until the variable is been set as > zero > => Playback the voice files ( Here I combined the voice > files with the delimiter value if more than one voice files are there) > => Set the freeswitch(my own) variable as true ( This is > used to identify whether the voice files are played > successfully). > => Wait in the loop until the variable is been set as > one. > => Set the Event lock as false > > => Trying to get the DTMF digits ( Have a assurance > that all the voice files are played). > > The problem is, > > The above steps are working fine when the voice file count is > lesser than or equal to 10. After the voice files are played only the > variable(my own freeswitch) is set. Based on the variable I am doing further > things. > > But when I tried to give the voice files count of more than 10 > the variable has been set while starting to play back the first voice file > itself . Because of this I am not able to proceed further. > > *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* > > *NOTE*: I also referred mod_file_string documentation. In that they > specified 128 files can be used to play back the voice files using > playback_delimiter option. > > Please help me................? > Thanks in advance. > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/7139944b/attachment.html From anthony.minessale at gmail.com Mon Nov 23 07:57:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 09:57:47 -0600 Subject: [Freeswitch-users] Problems with sighup and rotating csv files In-Reply-To: References: Message-ID: <191c3a030911230757j4bfd84e4mbf80534cecce5107@mail.gmail.com> did you really mean 1.0.4pre7 ? We are now on pre release of 1.0.5 so I cannot really debug such an old version so you may want to install one of the newer version first before you report an issue and when you do use our issue tracker not this mailing list http://jira.freeswitch.org be sure to answer all the questions carefully when filing the report. On Sun, Nov 22, 2009 at 12:46 PM, katarina djakovic wrote: > Hi, > I am using the Freeswitch 1.0.4pre7. Great application, but I encountered a > problem wich I can not solve since I am very new to it. > Two things are happening. > 1) The mod_cdr_csv.c (line 122 do_rotate()) does not always respond to > sighup signal to rotate the cdr-csv files. Some times it happens and some > times it does not. I can not see any pattern in the behaviour. Seems that > sometimes functions in the mod_cdr_csv.c catch the signal and some times > they do not. > > 2) Playing with the "kill -HUP fspid" all of a sudden I started getting two > freeswitch processes in the process list. One being parent of another. > Then, when I send the sighup signal to the parent - the console dies off > and the other freeswitch process stays (leaving the comment "Hangup" in the > fsconsole). Freeswitch conitnues to work with the remaining process. > In case when I send the sighup to the child, it will rotate the log files. > However, it always rotates the freeswitch.log, but randomly rotates the > cdr-csv files. > > 3) I have a feeling that above behaviours are somehow connected, but do not > understand how. > > Anyone can help? > Any comment or idea will be very very much apreciated. > Cheers, > Katarina > > ------------------------------ > Windows Live: Make it easier for your friends to see what you?re up to on > Facebook. > > ------------------------------ > Windows Live: Friends get your Flickr, Yelp, and Digg updates when they > e-mail you. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/d75a0624/attachment-0001.html From anthony.minessale at gmail.com Mon Nov 23 08:00:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 10:00:11 -0600 Subject: [Freeswitch-users] Media got stuck after attended transfer... In-Reply-To: References: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com> <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com> <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> <65d96fc80911220415v70d0bafbvad56c4fcb4576d8b@mail.gmail.com> Message-ID: <191c3a030911230800t6926fcb8w37d55d0bd794f185@mail.gmail.com> I think that issue has been fixed in trunk re: proxy-mode and resume-media-on-hold On Sun, Nov 22, 2009 at 7:00 AM, Klaus Hochlehnert wrote: > For ?only? sending and receiving that?s true. > > > > But my customer wants 2 things: > > - Using HylaFAX as fax server, as there are a lot of client apps and other > tools > > - Connecting ?real? fax machines using a Linksys/Cisco SPA2102 (as this is > certified by their SIP/ISDN gateway vendor) > > > > So I could really need t38 handling in FS to don?t make things more > complicated as they already are? J > > Proxy mode doesn?t work for me because it gives an error when > resume-media-on-hold is set. > > > > Klaus > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir > Culjaga > *Sent:* Sunday, November 22, 2009 1:15 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Media got stuck after attended > transfer... > > > > it is better to enhance mod_fax with t.38 support... we have done sometihng > and it is close to be work... > > T. > > On Sat, Nov 21, 2009 at 2:17 AM, Michael Jerris wrote: > > I think a better approach here is to use spandsp. We already have some > groundwork done for this. If you are interested in contributing, please > email consulting at freeswitch.org and we can discuss further. > > > > Mike > > > > On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: > > > > Hi, > > > > one of my customers is willing to contribute for t38 integration. > > > > The basic idea is to connect HylaFAX to FS: > > t38modem <-> FreeSWITCH <-> Media Gateway with t38 support > > All this without media proxy. > > > > Another idea might be to implement t38 origination/termination with a class > 1 modem input/output for use with HylaFAX. > > > > Do you know how much money we need to collect for t38 support? > > How much time is needed for implementing this? > > > > Thanks, Klaus > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, October 16, 2009 2:10 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Media got stuck after attended > transfer... > > > > > > On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga > wrote: > > hi, any clue when can t38 be added? > > > "Eventually." :) Of course, if we could get more to add to the bounty it > might grease the wheels of innovation. > > > http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch > > Of course, I was listening to my A.M radio the other day and they said that > there was this new invention called the Internet that would let people send > documents to each other electronically. Maybe you should look into that. > Next thing you know they'll come up with telephones that people don't have > to plug into the wall and can take with them in the car. ;) > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/420082bc/attachment.html From grimsqueaker13 at gmail.com Mon Nov 23 05:53:03 2009 From: grimsqueaker13 at gmail.com (Daniel Browne) Date: Mon, 23 Nov 2009 15:53:03 +0200 Subject: [Freeswitch-users] mod_ldap Message-ID: I'm new to Freeswitch and I'm looking for some help using mod_ldap to authenticate SIP endpoints to an LDAP database on registration. I have successfully installed the Freeswitch trunk and OpenLDAP 2.4.18 and I have compiled mod_ldap. I have setup a config file for mod_ldap. It is active and I can see its config in the compiled Freeswitch config file. I have turned on SIP traces, sofia logging and console logging in the Freeswitch command line. However, when I register a SIP phone to my Freeswitch server, I see no reference to mod_ldap in any logs. I have not set up the correct schema in ldap yet, so I would expect to see some indication that the module has searched my ldap server and found no useful information. My phone registers correctly and the normal tests (eg. music on hold) work. I am not sure if mod_ldap falls back on the usual xml config files if it fails to find information on the specified ldap server (as mod_xml_curl does), so I can't be sure if it is working or if my phone is simply registering in the normal way. Can anyone give me pointers on where to look next? If I can just get some feedback from the module I should be able to work out what to do. Thanks -- Grimsqueaker "Even a fool, when he holdeth his peace, is counted wise." Proverbs 17:28a -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/ae5e7b70/attachment.html From malay.thakershi at continuityhealth.com Mon Nov 23 08:07:52 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Mon, 23 Nov 2009 10:07:52 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> Message-ID: <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> Ok. I understand that. It would be great if someone can help me figure out: 1. Why mod_flite is not changing to the female voice even though I tried switching all 4 profiles it provides? 2. I would be alright for purchasing Cepstral for its quality. But FS doesn't come with it compiled I guess (it says swift.dll required when I enabled it in the config file). I asked Cepstral support but they say I have to purchase their SDK (no trial available) even though I just need it to compile it with FS. I understand I will be purchase the voices but how can I get Cepstral DLLs without purchasing the SDK. Thank you for help. Malay Thakershi From: Brian West [mailto:brian at freeswitch.org] Sent: Friday, November 20, 2009 5:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles You pay top dollar for it. The free stuff just isn't as good as what you PAY good money for. I don't expect that to change anytime soon. /b On Nov 20, 2009, at 5:18 PM, Malay Thakershi wrote: Also, can someone tell me what is the best way to get TTS going with good quality? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/7b9b73c8/attachment-0001.html From brian at freeswitch.org Mon Nov 23 08:23:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 10:23:27 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> Message-ID: If you're on linux the SDK comes with the voices. /b On Nov 23, 2009, at 10:07 AM, Malay Thakershi wrote: > Ok. I understand that. > > It would be great if someone can help me figure out: > 1. Why mod_flite is not changing to the female voice even > though I tried switching all 4 profiles it provides? > 2. I would be alright for purchasing Cepstral for its quality. > But FS doesn?t come with it compiled I guess (it says swift.dll > required when I enabled it in the config file). I asked Cepstral > support but they say I have to purchase their SDK (no trial > available) even though I just need it to compile it with FS. I > understand I will be purchase the voices but how can I get Cepstral > DLLs without purchasing the SDK. > > Thank you for help. > > > Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/869f3ee4/attachment.html From mike at jerris.com Mon Nov 23 08:28:19 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 11:28:19 -0500 Subject: [Freeswitch-users] FS compile error under Windows: error LNK2019 In-Reply-To: References: Message-ID: It sounds like the platform sdk is set up wrong. This used to be a problem with older versions of express edition. Double check that your compiler works at all with anything else. Mike On Nov 22, 2009, at 11:51 PM, ??? wrote: > All, > > I tried to compile FS source code under Windows while there are lots of errors: > > Error LNK2019, external _imp_sleep at 4 can not be resolved, this function was referred by _tMCRTStartup. > > Some other more similiar errors detail information attached. > > Any ideas? > > Thanks > Daniel Zeng -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/5a410502/attachment.html From nameer.kazzaz at gmail.com Mon Nov 23 08:35:26 2009 From: nameer.kazzaz at gmail.com (Nameer Kazzaz) Date: Mon, 23 Nov 2009 16:35:26 +0000 Subject: [Freeswitch-users] SIP Digest nonce (stale="true") In-Reply-To: <191c3a030911230752r70b702b1g61694350f56b01e0@mail.gmail.com> References: <4B0A7EBB.8040702@gmail.com> <191c3a030911230752r70b702b1g61694350f56b01e0@mail.gmail.com> Message-ID: <4B0AB9CE.5040300@gmail.com> Hey Anthony, Thanks for the quick response. No the device is a OneAccess so they are saying 'no quotes is the standard'. Thanks Nameer Anthony Minessale wrote: > The quoted true is the correct way from my research. The commented > line was to test a device, a grandstream, they apparently do not > accept it with quotes and I was using the unquoted version it to > gather evidence to issue a bug report to them. They told me it will > be fixed in the next firmware, was this the brand of device you have > as well? > > > > On Mon, Nov 23, 2009 at 6:23 AM, Nameer Kazzaz > > wrote: > > Hi Anthony, > I'm having an issue with a gateway after the nonce-ttl expires we > are sending stale="true", the cpe some how only likes stale=true > without > the "". I see on rev 15441 > > you made a change and marked it out. So my question is who is > correct on > this is it the CPE or are we sticking with the quoted ("true"). > > Thanks > Nameer Kazzaz > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Mon Nov 23 08:38:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 10:38:21 -0600 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <855e4dcf0911221543o222bef63t1c3340b0a41d57c1@mail.gmail.com> References: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> <855e4dcf0911221543o222bef63t1c3340b0a41d57c1@mail.gmail.com> Message-ID: <191c3a030911230838l103bc466p7582c1d05730f61a@mail.gmail.com> There is a formula to implement caching but it's very complicated and nobody has had time to work on it. You have to take every single input variable into account when caching because who is calling the extension, why they are calling it when they are calling it all make a difference. Web servers are designed to get thousands of hits per second so typically they can handle delivering custom xml instruction quite well. If you do not require such a dynamic setup, you could generate static files instead. On Sun, Nov 22, 2009 at 5:43 PM, Tim Uckun wrote: > On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman wrote: > > Hi Sam, > > Take a look at mod_xml_curl. Pretty sure it'll do everything you're > looking > > for. > > > Looking at that diagram it seems like mod_xml_curl makes a call for > every SIP connection. That seems like overkill. Is there a way to set > it up so that it caches the XML it got for a period of time? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/1b3337c8/attachment.html From abeka at greatiam.com Mon Nov 23 08:46:07 2009 From: abeka at greatiam.com (Otis) Date: Mon, 23 Nov 2009 16:46:07 +0000 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: <4B097A89.2050400@greatiam.com> References: <4B086689.6080804@greatiam.com> <4B097A89.2050400@greatiam.com> Message-ID: <4B0ABC4F.1010103@greatiam.com> Hello Could anyone point out what I have missed please ? At the moment I configured a gateway voipuser as described here : Any suggestion as to what path I can take will be highly welcome Thanks . Sam Abekah-Mensah wrote: >
Hi Michael > > Thanks > > I had set it to send incoming calls to extension 1001. This is in the > file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory. > The contents are : > > > > > > > > > Is there > anything wrong with this please ? > > Thanks > > > > Michal Bielicki wrote: >> >> Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah: >> >>> >>> I need help as I cannot receive calls through VOIPUSER. This is a >>> learning setup Attached are my conf files. What is wrong with them ? >>> When I dial from a landline I get a continuous beep. >>> >>> Attached are my gateway and the conf file to transfer. Sopfia Status >>> is my screen message. I can see a FAIL and cannot make head or tail >>> of all that message. Hopefully anyone using voipuser or in fact any >>> of you clever folks can make sense of this. >>> >>> Thanks for your time. >>> >>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy >>> RFC2833 Mode! >>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec >>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>> 2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec >>> Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] >>> 2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec >>> sofia/external/nobody at 213.166.5.133 PCMA/8000 20 ms 160 samples >>> 2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf >>> payload to 101 >>> 2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 >>> (sofia/external/nobody at 213.166.5.133) State Change CS_NEW -> CS_INIT >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_INIT >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>> (sofia/external/nobody at 213.166.5.133) State INIT >>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 >>> sofia/external/nobody at 213.166.5.133 SOFIA INIT >>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 >>> (sofia/external/nobody at 213.166.5.133) State Change CS_INIT -> >>> CS_ROUTING >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>> (sofia/external/nobody at 213.166.5.133) State INIT going to sleep >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_ROUTING >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 >>> (sofia/external/nobody at 213.166.5.133) State ROUTING >>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 >>> sofia/external/nobody at 213.166.5.133 SOFIA ROUTING >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 >>> sofia/external/nobody at 213.166.5.133 Standard ROUTING >>> 2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing >>> anonymous->abeka in context public >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->unloop] continue=false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (PASS) [unloop] >>> ${unroll_loops}(true) =~ /^true$/ break=on-false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [unloop] >>> ${sip_looped_call}() =~ /^true$/ break=on-false >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->outside_call] continue=true >>> Dialplan: sofia/external/nobody at 213.166.5.133 Absolute Condition >>> [outside_call] >>> Dialplan: sofia/external/nobody at 213.166.5.133 Action >>> set(outside_call=true) >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->call_debug] continue=true >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->public_extensions] continue=false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>> [public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/ >>> break=on-false >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->public_did] continue=false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>> [public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->sip at sip.voipuser.org] continue=false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>> [sip at sip.voipuser.org] destination_number(abeka) =~ /08715042951/ >>> break=on-false >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->Inbound-abeka at sip.voipuser.org]] continue=false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>> [Inbound-abeka at sip.voipuser.org]] destination_number(abeka) =~ >>> /[08444846450]/ break=on-false >>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 >>> (sofia/external/nobody at 213.166.5.133) State Change CS_ROUTING -> >>> CS_EXECUTE >>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 >>> (sofia/external/nobody at 213.166.5.133) State ROUTING going to sleep >>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_EXECUTE >>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 >>> (sofia/external/nobody at 213.166.5.133) State EXECUTE >>> 2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 >>> sofia/external/nobody at 213.166.5.133 SOFIA EXECUTE >>> 2009-11-21 22:07:15.706658 [DEBUG] switch_core_state_machine.c:151 >>> sofia/external/nobody at 213.166.5.133 Standard EXECUTE >>> EXECUTE sofia/external/nobody at 213.166.5.133 set(outside_call=true) >>> 2009-11-21 22:07:15.728613 [DEBUG] mod_dptools.c:748 >>> sofia/external/nobody at 213.166.5.133 SET [outside_call]=[true] >>> 2009-11-21 22:07:15.728613 [NOTICE] switch_core_state_machine.c:179 >>> Hangup sofia/external/nobody at 213.166.5.133 [CS_EXECUTE] >>> [NORMAL_CLEARING] >>> 2009-11-21 22:07:15.728613 [DEBUG] switch_channel.c:1683 Send signal >>> sofia/external/nobody at 213.166.5.133 [KILL] >>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:491 >>> (sofia/external/nobody at 213.166.5.133) State EXECUTE going to sleep >>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_HANGUP >>> 2009-11-21 22:07:15.735830 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/external/nobody at 213.166.5.133) State HANGUP >>> 2009-11-21 22:07:15.735830 [DEBUG] mod_sofia.c:338 Channel >>> sofia/external/nobody at 213.166.5.133 hanging up, cause: NORMAL_CLEARING >>> 2009-11-21 22:07:15.737680 [DEBUG] mod_sofia.c:417 Responding to >>> INVITE with: 480 >>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:46 >>> sofia/external/nobody at 213.166.5.133 Standard HANGUP, cause: >>> NORMAL_CLEARING >>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/external/nobody at 213.166.5.133) State HANGUP going to sleep >>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_state_machine.c:476 >>> (sofia/external/nobody at 213.166.5.133) State Change CS_HANGUP -> >>> CS_REPORTING >>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_REPORTING >>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:612 >>> (sofia/external/nobody at 213.166.5.133) State REPORTING >>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:53 >>> sofia/external/nobody at 213.166.5.133 Standard REPORTING, cause: >>> NORMAL_CLEARING >>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:612 >>> (sofia/external/nobody at 213.166.5.133) State REPORTING going to sleep >>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:411 >>> (sofia/external/nobody at 213.166.5.133) State Change CS_REPORTING -> >>> CS_DESTROY >>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_session.c:1068 >>> Session 2 (sofia/external/nobody at 213.166.5.133) Locked, Waiting on >>> external entities >>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1086 >>> Session 2 (sofia/external/nobody at 213.166.5.133) Ended >>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1088 Close >>> Channel sofia/external/nobody at 213.166.5.133 [CS_DESTROY] >>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>> (sofia/external/nobody at 213.166.5.133) State DESTROY >>> 2009-11-21 22:07:15.802636 [DEBUG] mod_sofia.c:255 >>> sofia/external/nobody at 213.166.5.133 SOFIA DESTROY >>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:60 >>> sofia/external/nobody at 213.166.5.133 Standard DESTROY >>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>> (sofia/external/nobody at 213.166.5.133) State DESTROY going to sleep >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> : >> >> >> you seem to have not specified an extension where the call should go to >> my voipuser.org setup looks like: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> I am also surprised that your setup works with a from-domain of >> sip.voipuser.org >> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> *Michal Bielicki* >> HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa >> t. +48228753290 | f. +48228753291 michal.bielicki at halokwadrat.pl >> | w. >> www.halokwadrat.pl >> >> >> >> *Knowledge & Low Prices. Guaranteed!* >> > > > >
From egable+freeswitch at gmail.com Mon Nov 23 08:48:37 2009 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Mon, 23 Nov 2009 11:48:37 -0500 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <191c3a030911230838l103bc466p7582c1d05730f61a@mail.gmail.com> References: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> <855e4dcf0911221543o222bef63t1c3340b0a41d57c1@mail.gmail.com> <191c3a030911230838l103bc466p7582c1d05730f61a@mail.gmail.com> Message-ID: Or, you can use something like Smarty to cache your generated XML on your web server and only invalidate those cached results when you change something that will impact them. On Mon, Nov 23, 2009 at 11:38 AM, Anthony Minessale wrote: > There is a formula to implement caching but it's very complicated and nobody > has had time to work on it. > You have to take every single input variable into account when caching > because who is calling the extension, why they are calling it when they are > calling it all make a difference. > > Web servers are designed to get thousands of hits per second so typically > they can handle delivering custom xml instruction quite well. > > If you do not require such a dynamic setup, you could generate static files > instead. > > > On Sun, Nov 22, 2009 at 5:43 PM, Tim Uckun wrote: >> >> On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman wrote: >> > Hi Sam, >> > Take a look at mod_xml_curl. ?Pretty sure it'll do everything you're >> > looking >> > for. >> >> >> Looking at that diagram it seems like mod_xml_curl makes a call for >> every SIP connection. That seems like overkill. ?Is there a way to set >> it up so that it caches the XML it got for a period of time? >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From anthony.minessale at gmail.com Mon Nov 23 08:50:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 10:50:11 -0600 Subject: [Freeswitch-users] SIP Digest nonce (stale="true") In-Reply-To: <4B0AB9CE.5040300@gmail.com> References: <4B0A7EBB.8040702@gmail.com> <191c3a030911230752r70b702b1g61694350f56b01e0@mail.gmail.com> <4B0AB9CE.5040300@gmail.com> Message-ID: <191c3a030911230850g5fdf5558wfb53237f3179b52b@mail.gmail.com> Tell you what, I don't have the patience for it, i'm sure most stuff does it either way and I'm sure nobody insists you have them so I will take them out so I can have some peace. On Mon, Nov 23, 2009 at 10:35 AM, Nameer Kazzaz wrote: > Hey Anthony, > Thanks for the quick response. No the device is a OneAccess so they > are saying 'no quotes is the standard'. > > Thanks > Nameer > > Anthony Minessale wrote: > > The quoted true is the correct way from my research. The commented > > line was to test a device, a grandstream, they apparently do not > > accept it with quotes and I was using the unquoted version it to > > gather evidence to issue a bug report to them. They told me it will > > be fixed in the next firmware, was this the brand of device you have > > as well? > > > > > > > > On Mon, Nov 23, 2009 at 6:23 AM, Nameer Kazzaz > > > wrote: > > > > Hi Anthony, > > I'm having an issue with a gateway after the nonce-ttl expires we > > are sending stale="true", the cpe some how only likes stale=true > > without > > the "". I see on rev 15441 > > < > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_reg.c?r=15441#l687 > > > > you made a change and marked it out. So my question is who is > > correct on > > this is it the CPE or are we sticking with the quoted ("true"). > > > > Thanks > > Nameer Kazzaz > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/6dcbf497/attachment.html From jlenk at frontiernet.net Mon Nov 23 09:01:25 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Mon, 23 Nov 2009 09:01:25 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <858430.90192.qm@web111301.mail.gq1.yahoo.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> <1258949788572-4048969.post@n2.nabble.com> <858430.90192.qm@web111301.mail.gq1.yahoo.com> Message-ID: <1258995685201-4052409.post@n2.nabble.com> Hi Arsen, I have merged your changes in now - thank you. Would you perhaps be able to look at the x64 changes I made to the projects and merge them back into your code to ease the future updating. Thanks Jeff Arsen Chaloyan wrote: > > Hi Jeff, > > > Your input would be very helpful, I just wanted to understand where the > problem is and contribute the way I can. > I see you're the assignee, so please go ahead and let me know if there is > anything left I can help with. > > Arsen. > > > > ________________________________ > From: Jeff Lenk > To: freeswitch-users at lists.freeswitch.org > Sent: Mon, November 23, 2009 8:16:28 AM > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > Hi Arsen, > > I would be happy to help with the FS integration if you want - please do > put your patch in a Jira. > > Jeff > > ________________________________ > Date: Sun, 22 Nov 2009 10:09:41 -0800 > From: [hidden email] > To: [hidden email] > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > > We discussed build integration related issues a few months ago with Mike > and seemed to find a solution which would work for both UniMRCP and > FreeSWITCH source trees. > > Now I've just got a chance to look into this a bit closer trying to > further complete VS2008 build integration in FreeSWITCH. So I've got it > working, the module is not only being built, but also is getting loaded. > Current build integration is not as seamless as I want it to be, but > probably we can start with what we have now and then discuss and identify > what can be done in the future. This concerns not only build integration > but overall integrity. > > So would you be interested in the patch? Where should I upload it? > I thought I had a Jira account, but not sure it exists any more. > > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org > > > > > > ________________________________ > From: Jeff Lenk <[hidden email]> > To: [hidden email] > Sent: Fri, November 20, 2009 7:59:28 PM > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > > That module is not currently being built for Windows. Also the library > unimrcp needs build integration work with FS to make that happen under > windows. > > > ss1 wrote: > >> >> Hi Everyone, >> >> Please help freeswitch experts... !!! >> >> i have been working on freeswitch from last 2 days. i have downloaded >> freeswitch and unimrcp (server + client) for windows. >> I tested the unimrcp client and server, which is running fine with the >> command: run synth and run recog. I got both synth.pcm & recog.pcm files. >> >> But my objective is to call Freeswitch through x-lite, where freeswitch >> should call unimrcp client and return the PCM files. >> >> I tried it alot, but unable to do it. after lots of reading i found that >> i >> do not have mod_unimrcp. i do not know from where to download it and how >> to merge it into freeswitch. >> >> I would be very thankful if you may help. >> >> Thanks, >> ss >> >> -- > View this message in context: > http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > > View message @ > http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html > To unsubscribe from Re: need help !! Problem with freeswitch & uniMRCP, > click here. > > ________________________________ > Hotmail: Trusted email with powerful SPAM protection. Sign up now. > ________________________________ > View this message in context: RE: [Freeswitch-users] need help !! Problem > with freeswitch & uniMRCP > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4052409.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jonas.gauffin at gmail.com Mon Nov 23 09:08:44 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Mon, 23 Nov 2009 18:08:44 +0100 Subject: [Freeswitch-users] NAT problem Message-ID: Hello I got the following setup: Phones -> FreeSwitch -> NAT -> Internet -> Gateway And I'm struggling to get NAT working properly. I'm running freeswitch with the "-nonat" option and have tried different ext-rtp-ip/ext-sip-ip combinations in external/internal profiles. The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly. Currently I have this setup: API CALL [sofia(status profile external)] output: ======================================================== Name external Domain Name N/A Context public Challenge Realm auto_to RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 Ext-SIP-IP 85.89.XX.XX OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false API CALL [sofia(status profile default)] output: ======================================================== Name default Domain Name N/A Alias Of internal Context public Challenge Realm auto_from RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED false STUN-AUTO-DISABLE false Sample phone registration: Call-ID: Xmbw9PyQ5Q6L2MnQ at 192.168.1.121 User: u1000009 at default Contact: "u1000009" Agent: IP PHONE 3 V1.58.004 CFG0 Status: Registered(UDP)(unknown) EXP(2009-11-23 19:26:40) Host: jonas-PC IP: 192.168.1.121 Port: 6094 Auth-User: u1000009 Auth-Realm: default MWI-Account: u1000009 at default Outbound INVITE: send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000: ------------------------------------------------------------------------ INVITE sip:0706930XXX at sipgw2.XXXXX.se SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp Max-Forwards: 69 From: "Kundtj??nst Arne" ;tag=B7pve7F6eeH7c To: > Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 CSeq: 123379614 INVITE Contact: Call-Info: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 293 X-FS-Support: update_display Remote-Party-ID: "Kundtj??nst Arne" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110 s=FreeSWITCH c=IN IP4 192.168.1.110 t=0 0 m=audio 24986 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Many thanks, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/3ca2dcf5/attachment-0001.html From brian at freeswitch.org Mon Nov 23 09:21:47 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 11:21:47 -0600 Subject: [Freeswitch-users] NAT problem In-Reply-To: References: Message-ID: You set the ext-rtp-ip on the profile the phones talk too... but you shouldn't be doing that. /b On Nov 23, 2009, at 11:08 AM, Jonas Gauffin wrote: > Hello > > I got the following setup: Phones -> FreeSwitch -> NAT -> Internet - > > Gateway > > And I'm struggling to get NAT working properly. I'm running > freeswitch with the "-nonat" option and have tried different ext-rtp- > ip/ext-sip-ip combinations in external/internal profiles. > The From header seems to be correct while contact header and SDP > uses local ip? Please help me configure everything correctly. > > Currently I have this setup: From jonas.gauffin at gmail.com Mon Nov 23 09:24:18 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Mon, 23 Nov 2009 18:24:18 +0100 Subject: [Freeswitch-users] NAT problem In-Reply-To: References: Message-ID: Ok. Found the problem. I had started using "sofia/outbound/ XXXXXX at sipgw2.XXXX.se" as bridge destination to try to get outbound_caller_id_name/outbound_caller_id_number working. It works if I use the correct profile name, "sofia/internal/ XXXXXX at sipgw2.XXXX.se" When do FS use outbound_caller_id instead of effective_caller_id? On Mon, Nov 23, 2009 at 6:08 PM, Jonas Gauffin wrote: > Hello > > I got the following setup: Phones -> FreeSwitch -> NAT -> Internet -> > Gateway > > And I'm struggling to get NAT working properly. I'm running freeswitch with > the "-nonat" option and have tried different ext-rtp-ip/ext-sip-ip > combinations in external/internal profiles. > The From header seems to be correct while contact header and SDP uses local > ip? Please help me configure everything correctly. > > Currently I have this setup: > > API CALL [sofia(status profile external)] output: > ======================================================== > Name external > Domain Name N/A > Context public > Challenge Realm auto_to > RTP-IP 192.168.1.110 > Ext-RTP-IP 85.89.XX.XX > SIP-IP 192.168.1.110 > Ext-SIP-IP 85.89.XX.XX > OUTBOUND-PROXY N/A > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > > API CALL [sofia(status profile default)] output: > ======================================================== > Name default > Domain Name N/A > Alias Of internal > Context public > Challenge Realm auto_from > RTP-IP 192.168.1.110 > Ext-RTP-IP 85.89.XX.XX > SIP-IP 192.168.1.110 > OUTBOUND-PROXY N/A > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED false > STUN-AUTO-DISABLE false > > Sample phone registration: > Call-ID: Xmbw9PyQ5Q6L2MnQ at 192.168.1.121 > User: u1000009 at default > Contact: "u1000009" > Agent: IP PHONE 3 V1.58.004 CFG0 > Status: Registered(UDP)(unknown) EXP(2009-11-23 19:26:40) > Host: jonas-PC > IP: 192.168.1.121 > Port: 6094 > Auth-User: u1000009 > Auth-Realm: default > MWI-Account: u1000009 at default > > Outbound INVITE: > send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000: > ------------------------------------------------------------------------ > INVITE sip:0706930XXX at sipgw2.XXXXX.seSIP/2.0 > Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp > Max-Forwards: 69 > From: "Kundtj??nst Arne" ;tag=B7pve7F6eeH7c > To: > > Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 > CSeq: 123379614 INVITE > Contact: > Call-Info: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 293 > X-FS-Support: update_display > Remote-Party-ID: "Kundtj??nst Arne" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110 > s=FreeSWITCH > c=IN IP4 192.168.1.110 > t=0 0 > m=audio 24986 RTP/AVP 0 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > Many thanks, > Jonas > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/686b9f8a/attachment.html From msc at freeswitch.org Mon Nov 23 09:27:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 09:27:01 -0800 Subject: [Freeswitch-users] IP0010 SIP Phone In-Reply-To: <10222602.1258898966890.JavaMail.root@whwamui-deputy.pas.sa.earthlink.net> References: <10222602.1258898966890.JavaMail.root@whwamui-deputy.pas.sa.earthlink.net> Message-ID: <87f2f3b90911230927hc286e28gac6578ce5c8432c2@mail.gmail.com> On Sun, Nov 22, 2009 at 6:09 AM, David V. Fansler wrote: > After the help of a couple of people from this list, I now have FreeSWITCH > running - yeah! I have installed X-Lite on a couple of computers and they > dial each other, play music on hold, etc. I have not yet connected to the > outside world. > > I purchased an IP-0010 phone off eBay ($20 including shipping - docs at > http://www.vanaccess.com/news/news_images/2007131_73_User%20Manual%20-%20IP0010.pdf) > I cannot get this phone to work with the system. It gets an IP address, > time/date, and a dial tone. After many tries with the http congifuration > tool, I got the phone "configured" with the address of the SIP server, and a > SIP User ID. When you dial an extension the FreeSWITCH window shows the > following: > > sofica.c3844 Hanugup sofia/internal/101 at 192.168.1.165 [CS_NEW] > [INCOMPATIBLE_DESTINATION] > switch_core_session.c1139 Session 20 (sofia/internal/101 at 192.165.1.65) > Ended > switch_core_session.c1141 Close Channel sofia/internal/1001 at 192.168.1.165[CS_DESTROY] > > Has anyone else tried this phone, or does anyone have suggestions I could > try. I have looked through the website but have not found anything to help. > > Thanks, > > David > > David, Time to do a little digging. First off, review this wiki page on reporting bugs - it has lots of useful information on how to gather information from your system and report it to the community: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd recommend that you get a debug log and a sip trace and post it to pastebin. Report back the pastebin URL here in this thread and we'll have a look. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/e8e61035/attachment.html From brian at freeswitch.org Mon Nov 23 09:31:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 11:31:02 -0600 Subject: [Freeswitch-users] NAT problem In-Reply-To: References: Message-ID: <56195859-FC9B-46C1-9ABE-88CCC26B881B@freeswitch.org> outbound_caller_id is a made up variable that is used in the defaults that are used in the examples only. /b On Nov 23, 2009, at 11:24 AM, Jonas Gauffin wrote: > Ok. Found the problem. I had started using "sofia/outbound/XXXXXX at sipgw2.XXXX.se > " as bridge destination to try to get outbound_caller_id_name/ > outbound_caller_id_number working. > It works if I use the correct profile name, "sofia/internal/XXXXXX at sipgw2.XXXX.se > " > > When do FS use outbound_caller_id instead of effective_caller_id? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/afed17c5/attachment.html From tleyden at branchcut.com Mon Nov 23 09:47:33 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Mon, 23 Nov 2009 22:17:33 +0430 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <4B0AA8B6.2080305@greatiam.com> References: <4B0AA8B6.2080305@greatiam.com> Message-ID: Yeah a kind user (Innotel) took the time to write up Cent OS installation instructions for wikipbx and posted it to the wiki: http://wikipbx.subwiki.com/forum/t-115012/freeswitch-svn-1-0-2-wikipbx-svn-61-centos-5-1-installation-instructions If you have any problems please post in the forum: http://wikipbx.subwiki.com/forum:start On Mon, Nov 23, 2009 at 7:52 PM, Otis wrote: > Hi Folks > > Is anyone using this on Fedora and is there a binary or installation > script anywhere > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/6756706d/attachment.html From jonas.gauffin at gmail.com Mon Nov 23 09:50:07 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Mon, 23 Nov 2009 18:50:07 +0100 Subject: [Freeswitch-users] NAT problem In-Reply-To: <56195859-FC9B-46C1-9ABE-88CCC26B881B@freeswitch.org> References: <56195859-FC9B-46C1-9ABE-88CCC26B881B@freeswitch.org> Message-ID: Ok. It would be a nice feature if outbound_caller_id was used by freeswitch. I do quite often bridge to both internal and external destinations in the same bridge command (as in "sofia/internal/5530,sofia/internal/ 070123456 at sipgw2.XXXX.se). This forces me to always use complete phone numbers in the caller id since my gateway would reject the call otherwise. It would be really neat if FS could use effective_caller_id (5531) for the internal bridge and outbound_caller_id (+4681235531) for the external bridge. On Mon, Nov 23, 2009 at 6:31 PM, Brian West wrote: > outbound_caller_id is a made up variable that is used in the defaults that > are used in the examples only. > > /b > > On Nov 23, 2009, at 11:24 AM, Jonas Gauffin wrote: > > Ok. Found the problem. I had started using "sofia/outbound/ > XXXXXX at sipgw2.XXXX.se" as bridge destination to try to get > outbound_caller_id_name/outbound_caller_id_number working. > It works if I use the correct profile name, "sofia/internal/ > XXXXXX at sipgw2.XXXX.se" > > When do FS use outbound_caller_id instead of effective_caller_id? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/32c8b040/attachment-0001.html From msc at freeswitch.org Mon Nov 23 09:51:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 09:51:50 -0800 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> Message-ID: <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy wrote: > Hi, > I'm using perl ESL to control the call in freeswitch. > I'm having the following scenario, but not able to get it right. > > Dialplan: > > > > > > > > > 1. User A calls to an extention (1000). > 2. My ESL program will be running, and it answers the call. > 3. Then the program will get a number from the user. > 4. It will hangup the call. > 5. The program has to call to the number that was given by the user. > > In the above scenario, I was able to do until the 4th step. After hangup > the call, if I say originate it is not working. > Any ideas on how to do this in ESL. > > I want to make sure I understand what the script is supposed to be doing. The caller will key in a phone number to your script and your script will collect those digits. The script will then hangup on the caller and originate a completely new call? Perhaps you could use sched_api to schedule a new originate command for a few seconds into the future and then hangup? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/8cc94bfe/attachment.html From brian at freeswitch.org Mon Nov 23 09:52:31 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 11:52:31 -0600 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: References: <4B0AA8B6.2080305@greatiam.com> Message-ID: s/i386/x86_64/ if you are 64bit /b On Nov 23, 2009, at 11:47 AM, Traun Leyden wrote: > > Yeah a kind user (Innotel) took the time to write up Cent OS > installation instructions for wikipbx and posted it to the wiki: > > http://wikipbx.subwiki.com/forum/t-115012/freeswitch-svn-1-0-2-wikipbx-svn-61-centos-5-1-installation-instructions > > If you have any problems please post in the forum: http://wikipbx.subwiki.com/forum:start > > On Mon, Nov 23, 2009 at 7:52 PM, Otis wrote: > Hi Folks > > Is anyone using this on Fedora and is there a binary or installation > script anywhere > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/6384852d/attachment.html From robert.hadley at teotech.com Mon Nov 23 09:53:16 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Mon, 23 Nov 2009 09:53:16 -0800 Subject: [Freeswitch-users] Building in a builddir using --srcdir option but modules still build in srcdir Message-ID: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? Thanks, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/c23bed4e/attachment.html From brian at freeswitch.org Mon Nov 23 09:56:17 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 11:56:17 -0600 Subject: [Freeswitch-users] NAT problem In-Reply-To: References: <56195859-FC9B-46C1-9ABE-88CCC26B881B@freeswitch.org> Message-ID: See default config it lsets you do that. Use the variables to store two versions of the callerid then set it depending on if its outside or inside... its rather easy to do. /b On Nov 23, 2009, at 11:50 AM, Jonas Gauffin wrote: > Ok. It would be a nice feature if outbound_caller_id was used by > freeswitch. > I do quite often bridge to both internal and external destinations > in the same bridge command (as in "sofia/internal/5530,sofia/internal/070123456 at sipgw2.XXXX.se > ). This forces me to always use complete phone numbers in the caller > id since my gateway would reject the call otherwise. > > It would be really neat if FS could use effective_caller_id (5531) > for the internal bridge and outbound_caller_id (+4681235531) for the > external bridge. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/ba55b893/attachment.html From malay.thakershi at continuityhealth.com Mon Nov 23 10:09:02 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Mon, 23 Nov 2009 12:09:02 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> Message-ID: <010701ca6c68$0950f6a0$1bf2e3e0$@thakershi@continuityhealth.com> I am not using Linux. I am using Windows 2008 server. Malay Thakershi From: Brian West [mailto:brian at freeswitch.org] Sent: Monday, November 23, 2009 10:23 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles If you're on linux the SDK comes with the voices. /b On Nov 23, 2009, at 10:07 AM, Malay Thakershi wrote: Ok. I understand that. It would be great if someone can help me figure out: 1. Why mod_flite is not changing to the female voice even though I tried switching all 4 profiles it provides? 2. I would be alright for purchasing Cepstral for its quality. But FS doesn't come with it compiled I guess (it says swift.dll required when I enabled it in the config file). I asked Cepstral support but they say I have to purchase their SDK (no trial available) even though I just need it to compile it with FS. I understand I will be purchase the voices but how can I get Cepstral DLLs without purchasing the SDK. Thank you for help. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/3a588fb3/attachment.html From brian at freeswitch.org Mon Nov 23 10:14:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 12:14:02 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <010701ca6c68$0950f6a0$1bf2e3e0$@thakershi@continuityhealth.com> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> <010701ca6c68$0950f6a0$1bf2e3e0$@thakershi@continuityhealth.com> Message-ID: <57CD54FA-C16E-4F36-9678-6CF1FB2D1A17@freeswitch.org> You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. /b On Nov 23, 2009, at 12:09 PM, Malay Thakershi wrote: > I am not using Linux. I am using Windows 2008 server. > > Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/41ffe00d/attachment-0001.html From william.suffill at gmail.com Mon Nov 23 10:22:16 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 23 Nov 2009 13:22:16 -0500 Subject: [Freeswitch-users] Git Message-ID: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> Just wondering if anyone is keeping an update to date git repo of FreeSwitch? I been using git-svn to keep a copy on my machines but it can be quite time consuming due to the per revision fetching. If there was a repo to clone that would speed up the process considerably. -- W From lon at kickasspixels.com Mon Nov 23 10:36:48 2009 From: lon at kickasspixels.com (Lon Baker) Date: Mon, 23 Nov 2009 10:36:48 -0800 Subject: [Freeswitch-users] Git In-Reply-To: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> References: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> Message-ID: William, Perhaps someone could setup one on github? It's free for open source project. Lon On Nov 23, 2009, at 10:22 AM, William Suffill wrote: > Just wondering if anyone is keeping an update to date git repo of > FreeSwitch? I been using git-svn to keep a copy on my machines but it > can be quite time consuming due to the per revision fetching. If there > was a repo to clone that would speed up the process considerably. > > -- W > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From abeka at greatiam.com Mon Nov 23 10:37:09 2009 From: abeka at greatiam.com (Otis) Date: Mon, 23 Nov 2009 18:37:09 +0000 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: References: <4B0AA8B6.2080305@greatiam.com> Message-ID: <4B0AD655.9070507@greatiam.com> Thanks. I have to get a centos box I guess. Much appreciated Samuel 'Otis' Traun Leyden wrote: > > Yeah a kind user (Innotel) took the time to write up Cent OS > installation instructions for wikipbx and posted it to the wiki: > > http://wikipbx.subwiki.com/forum/t-115012/freeswitch-svn-1-0-2-wikipbx-svn-61-centos-5-1-installation-instructions > > If you have any problems please post in the forum: > http://wikipbx.subwiki.com/forum:start > > On Mon, Nov 23, 2009 at 7:52 PM, Otis > wrote: > > Hi Folks > > Is anyone using this on Fedora and is there a binary or installation > script anywhere > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From malay.thakershi at continuityhealth.com Mon Nov 23 10:41:55 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Mon, 23 Nov 2009 12:41:55 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <57CD54FA-C16E-4F36-9678-6CF1FB2D1A17@freeswitch.org> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> <010701ca6c68$0950f6a0$1bf2e3e0$@thakershi@continuityhealth.com> <57CD54FA-C16E-4F36-9678-6CF1FB2D1A17@freeswitch.org> Message-ID: <012401ca6c6c$a19673a0$e4c35ae0$@thakershi@continuityhealth.com> Thank you for your responses. I did follow that web link to ask them as instructed but they declined. They asked me where I want to use it. I told them I wanted it to build FreeSwitch so that I can use Cepstral voices (to be purchased from them with it). Their response was they do not provide trial of the SDK. They do not support FreeSwitch. Malay Thakershi From: Brian West [mailto:brian at freeswitch.org] Sent: Monday, November 23, 2009 12:14 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. /b On Nov 23, 2009, at 12:09 PM, Malay Thakershi wrote: I am not using Linux. I am using Windows 2008 server. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/9f6321ab/attachment.html From mike at jerris.com Mon Nov 23 10:43:27 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 13:43:27 -0500 Subject: [Freeswitch-users] Git In-Reply-To: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> References: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> Message-ID: <1336A1DE-ECBA-43EA-968E-7C6BE89A1251@jerris.com> I think this one is kept up to date, but we may re-do this at some point soon, so it may get re-built. http://svn.freeswitch.org/freeswitch.git/ Mike On Nov 23, 2009, at 1:22 PM, William Suffill wrote: > Just wondering if anyone is keeping an update to date git repo of > FreeSwitch? I been using git-svn to keep a copy on my machines but it > can be quite time consuming due to the per revision fetching. If there > was a repo to clone that would speed up the process considerably. From mike at jerris.com Mon Nov 23 10:46:48 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 13:46:48 -0500 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911211523k7998d048nced3af8fb805e770@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> <2160023e0911211523k7998d048nced3af8fb805e770@mail.gmail.com> Message-ID: This looks like a nat issue to me, please re-test this against latest svn trunk and if its still not working pastebin a full sip trace and report the link back here. Mike On Nov 21, 2009, at 6:23 PM, RobertT wrote: > Yep, I use proxy media. First it started with 1.0.4 release, then I've updated a week or two ago with the latest svn trunk, not sure what was the rev number. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From william.suffill at gmail.com Mon Nov 23 10:52:06 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 23 Nov 2009 13:52:06 -0500 Subject: [Freeswitch-users] Git In-Reply-To: <1336A1DE-ECBA-43EA-968E-7C6BE89A1251@jerris.com> References: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> <1336A1DE-ECBA-43EA-968E-7C6BE89A1251@jerris.com> Message-ID: <6b65470d0911231052q57129ab5n2359c06d327d93d4@mail.gmail.com> I'd rather it be a decision by the community as a whole and authorized. Sure there are ways to have anyone who wants to on their own. Thanks for the insight. -- W From brian at freeswitch.org Mon Nov 23 10:53:30 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 12:53:30 -0600 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: References: <4B0266F4.8070602@savion.huji.ac.il> <4B0387F1.7070105@savion.huji.ac.il> <193640CC-3E62-4248-8E80-CE7FE82653C0@jerris.com> Message-ID: <6293479E-C530-4510-BD4D-592FE3E79D35@freeswitch.org> Because if you dial local-user at local-domain thats not the correct way this will usually trigger a call out and back in on the profile thus moving you one leg away from the actual user. If you're going to do that use sofia_contact and review how the defaults abstract this so you can just call user/xxxx at domain, You need to make sure the presence_id is set like the defaults have it. /b On Nov 22, 2009, at 1:39 AM, Yehavi Bourvine wrote: > Thanks Mike! However, this doesn't fully solve my problem. When > using sofia_contact() indeed it works ok with finding the > destination's profile. However, it breaks the BLFs... > > When calling sofia/sip_profile/local-user%local-domain the BLF works > ok. When calling sofia_contact(sofia/sip_profile/local-user at local- > domain) BLF doesn't work (nothing is sent to the watching phone). > > Any more clues??? > > Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/170c2443/attachment.html From mike at jerris.com Mon Nov 23 10:58:57 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 13:58:57 -0500 Subject: [Freeswitch-users] Adding headers to INFO messages for Advice of Charge on SNOM In-Reply-To: <1258963204.4961.8.camel@keithl-lt> References: <1258963204.4961.8.camel@keithl-lt> Message-ID: <914F7A85-3229-469E-92A2-FD9664FCC03D@jerris.com> Is there any rfc on this or is it something that snom just made up? On Nov 23, 2009, at 3:00 AM, Keith Laaks wrote: > > > > Hi, > > I have tried maintaining charging information on a SNOM 300's display > using 'display' - but found that the phone has some timer, whereby every > 60 seconds it wipes out whatever happens to be on the display at that > time and replaces is with the dialled number. So not a viable option as > it impacts usability. Really annoying when the display was just updated > with valuable information for the user and a split second later it gets > replaced. > [If somebody knows how to disable this behaviour - please do tell...] > > I see that SNOM supports a number of features for Advice of Charge. > >> From their Wiki: > > http://wiki.snom.com/Advice_of_charge_%28AOC%29_in_SIP > Example of an SIP-Info Message: > > ----------------------------------------------------- > INFO sip:bla at snom.com SIP/2.0 > From: ;tag=5354n3 > To: ;tag=33rfh3 > CSeq: 23423 INFO > Call-ID: 3452tw43dt354dm03 > AOC: charging;state=active; > charging-info=currency; > currency=EUR; > amount=2000; > multiplier=0.001 > Content-Length: 0 > ----------------------------------------------------- > > So the question - Is there some method available today to add these additional > 'new' headers to an INFO message I can send out to these phones? > > If not, I guess it's a matter of looking at enhancing the "case SWITCH_MESSAGE_INDICATE_DISPLAY" section in mod_sofia.c ? > From mike at jerris.com Mon Nov 23 11:01:24 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:01:24 -0500 Subject: [Freeswitch-users] User who answer the bridge in a execute_answer In-Reply-To: <4B0A65C9.10509@daccii.it> References: <4B0A65C9.10509@daccii.it> Message-ID: <9133578A-F706-46C2-9653-6C22D6E056CB@jerris.com> Try running the info app there and see if the info is anywhere in that output . Mike On Nov 23, 2009, at 5:36 AM, Albano Daniele Salvatore - Lavoro wrote: > Hi, > > i'm writing some dialplan parts that get executed on execute_on_answer. In this dialplan that get executed i need to make a directory to handle recordings for record_session and my folder structure is: > USER/YEAR/MONTH/HOUR-MINUTE-SECOND-CALLER_NUMBER.wav > > ------ > > > ------ > > The call flow is: > Call from external -> IVR -> Transfer to Group -> Execute on Answer -> system/bind_meta_app > > > Pratically, i need the number (or better the user) that answered the call: what variable should i check? > > I tried with sip_from_user, callee_id_number and some other. > > > Thank for your help, > > Best Regards, > Daniele > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Mon Nov 23 11:02:11 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:02:11 -0500 Subject: [Freeswitch-users] Execute on Answer with JavaScript In-Reply-To: <26476532.post@talk.nabble.com> References: <26476532.post@talk.nabble.com> Message-ID: This is done automatically when you bridge 2 sessions together. Mike On Nov 23, 2009, at 6:45 AM, Oscav wrote: > How can we send the answer to the caller only when the callee answers, in > JavaScript?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/28a00622/attachment.html From mike at jerris.com Mon Nov 23 11:03:34 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:03:34 -0500 Subject: [Freeswitch-users] Question about rtp-timeout-sec variable In-Reply-To: <1258977639954-4050650.post@n2.nabble.com> References: <1258977639954-4050650.post@n2.nabble.com> Message-ID: Take a look at a pcap of the traffic, I suspect the other side still has media flowing. On Nov 23, 2009, at 7:00 AM, Maciej Aniserowicz wrote: > > Hello, > I have 2 instances of FS: one controlled by my application (making calls > with TCP commands, recording sessions, listening to events etc) and one > acting as a remote gateway to which all users register. When I leave the > default values of rtp-timeout-sec and brutally kill x-lite during > conversation, the 'hangup' event with 'media_timeout' cause is obviously > sent after the default 5 minutes (and until then, the other leg is still > connected to a 'dead' channel). > The question is: which FS instance is responsible for terminating the > connection after timeout? Only the 'remote' FS instance config seems to > work. I thought that the shortest configured value should cause the timeout, > but it's not the case. Am I missing something, or is this the correct > behavior? From mike at jerris.com Mon Nov 23 11:05:06 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:05:06 -0500 Subject: [Freeswitch-users] Using odbc in FS core In-Reply-To: <1202092411.20091123134528@yes.net.ua> References: <1382216794.20091121134106@yes.net.ua> <1013085378.20091121140207@yes.net.ua> <191c3a030911210814l6e50b883uba61815fcd36afe1@mail.gmail.com> <1202092411.20091123134528@yes.net.ua> Message-ID: Yes please On Nov 23, 2009, at 6:45 AM, Mike Tkachuk wrote: > Hello Anthony, > > Is clear, thanks, I'll test and will let you know. > Should I add 'core-db-dsn' parameter description to Wiki? Maybe we need to add this parameter also to sample conf files? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/0ff063e4/attachment.html From mike at jerris.com Mon Nov 23 11:16:18 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:16:18 -0500 Subject: [Freeswitch-users] Building in a builddir using --srcdir option but modules still build in srcdir In-Reply-To: References: Message-ID: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com> The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: > I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/b1038920/attachment.html From christian.loeschenkohl at xpirio.com Mon Nov 23 11:17:53 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 23 Nov 2009 20:17:53 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing Message-ID: <4B0ADFE1.4070506@xpirio.com> hi our freeswitch server has to talk to a sonus ip-switch when we want to setup a call we do get a "100 Trying" and then a "180 Ringing" within the "180 Ringing" we get a sdp with "a=sendonly" then our freeswitch quits with a CANCEL message. i simply don't get why our freeswitch aborts the session - i think it would work if no "a=sendonly" would be present in the sdp. my technical contact doesn't want to switch 180 to 183 on the sonus side - this would also work (i think). in fact he says that 180 ringing is vaild, he isn't that wrong in this case. our freeswitch works in proxy mode, we do use trunk 15396 see a ngrep trace under http://pastebin.freeswitch.org/11235 92.63.208.36 - freeswitch 38.105.229.100 - sonus br -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From mike at jerris.com Mon Nov 23 11:22:17 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:22:17 -0500 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <012401ca6c6c$a19673a0$e4c35ae0$@thakershi@continuityhealth.com> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> <010701ca6c68$0950f6a0$1bf2e3e0$@thakershi@continuityhealth.com> <57CD54FA-C16E-4F36-9678-6CF1FB2D1A17@freeswitch.org> <012401ca6c6c$a19673a0$e4c35ae0$@thakershi@continuityhealth.com> Message-ID: <80AF7620-0980-4B75-A9B0-F046356DF591@jerris.com> Sounds like they don't want your business that much. You can try using mrcp with them , not sure if they have that released on their side or not. I think the build integration for mrcp client just went into the windows build earlier today. To be honest we used to have a pretty good relationship with them but we have had basically no response at all to any technical problems we have had with them in quite some time, so maybe they have decided to move on and not work with open source any more. It would appear so from their actions at least. Mike On Nov 23, 2009, at 1:41 PM, Malay Thakershi wrote: > Thank you for your responses. > > I did follow that web link to ask them as instructed but they declined. They asked me where I want to use it. > > I told them I wanted it to build FreeSwitch so that I can use Cepstral voices (to be purchased from them with it). Their response was they do not provide trial of the SDK. They do not support FreeSwitch. > > Malay Thakershi > > From: Brian West [mailto:brian at freeswitch.org] > Sent: Monday, November 23, 2009 12:14 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_flite sound profiles > > You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. > > /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/a4153509/attachment-0001.html From pjintheusa at gmail.com Mon Nov 23 11:24:18 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 23 Nov 2009 14:24:18 -0500 Subject: [Freeswitch-users] Simplest of Conference Setup questions Message-ID: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> Hi there, I have created a simple conference that works great. The only problem is, when a participant press # it exits the call. So when a user enters a conference with a PIN, and by habit they enter 12345 followed by pound, it puts them in and then straight out. So I edited conference.conf.xml so: and even assigned # to another function: and the same occurs. Pressing # exits the conference. What am I missing here? tia - phil Conf Setup: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/4ad7a208/attachment.html From anthony.minessale at gmail.com Mon Nov 23 11:35:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 13:35:54 -0600 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> Message-ID: <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> issue console loglevel debug from the cli then try again and see if there is any hint On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones wrote: > Hi there, > > I have created a simple conference that works great. The only problem is, > when a participant press # it exits the call. So when a user enters a > conference with a PIN, and by habit they enter 12345 followed by pound, it > puts them in and then straight out. > > So I edited conference.conf.xml so: > > > > and even assigned # to another function: > > > > and the same occurs. Pressing # exits the conference. > > What am I missing here? > > tia - phil > > > > Conf Setup: > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/c6e78263/attachment.html From anthony.minessale at gmail.com Mon Nov 23 11:40:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 13:40:22 -0600 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> Message-ID: <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> or open a new outbound connection at the end of your script so you can send your originate command. Since the channel hanging up will close your existing connection since it's only an outbound single session socket. On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins wrote: > > > On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> Hi, >> I'm using perl ESL to control the call in freeswitch. >> I'm having the following scenario, but not able to get it right. >> >> Dialplan: >> >> >> >> >> >> >> >> >> 1. User A calls to an extention (1000). >> 2. My ESL program will be running, and it answers the call. >> 3. Then the program will get a number from the user. >> 4. It will hangup the call. >> 5. The program has to call to the number that was given by the user. >> >> In the above scenario, I was able to do until the 4th step. After hangup >> the call, if I say originate it is not working. >> Any ideas on how to do this in ESL. >> >> > I want to make sure I understand what the script is supposed to be doing. > The caller will key in a phone number to your script and your script will > collect those digits. The script will then hangup on the caller and > originate a completely new call? Perhaps you could use sched_api to schedule > a new originate command for a few seconds into the future and then hangup? > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/70c790ad/attachment.html From anthony.minessale at gmail.com Mon Nov 23 11:45:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 13:45:34 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <4B0ADFE1.4070506@xpirio.com> References: <4B0ADFE1.4070506@xpirio.com> Message-ID: <191c3a030911231145j384e5bbat7208633895b3b7af@mail.gmail.com> you need to provide a FS console trace of your problem from your FS source dir (build root) cd libs/esl make perlmod cd perl perl logger.pl -pb christian reproduce then hit ctl-c and tell me the url it posted to. 2009/11/23 Christian L?schenkohl > hi > > our freeswitch server has to talk to a sonus ip-switch > when we want to setup a call we do get a "100 Trying" and then a "180 > Ringing" > within the "180 Ringing" we get a sdp with "a=sendonly" then our freeswitch > quits with a CANCEL message. > i simply don't get why our freeswitch aborts the session - i think it would > work > if no "a=sendonly" would be present in the sdp. > > my technical contact doesn't want to switch 180 to 183 on the sonus side - > this would > also work (i think). in fact he says that 180 ringing is vaild, he isn't > that wrong in > this case. > > our freeswitch works in proxy mode, we do use trunk 15396 > see a ngrep trace under http://pastebin.freeswitch.org/11235 > > 92.63.208.36 - freeswitch > 38.105.229.100 - sonus > > br > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/5a0fe41b/attachment.html From brian at freeswitch.org Mon Nov 23 11:48:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 13:48:18 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <4B0ADFE1.4070506@xpirio.com> References: <4B0ADFE1.4070506@xpirio.com> Message-ID: <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> Well its also G729 so I suspect you don't have G729 /b On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > hi > > our freeswitch server has to talk to a sonus ip-switch > when we want to setup a call we do get a "100 Trying" and then a > "180 Ringing" > within the "180 Ringing" we get a sdp with "a=sendonly" then our > freeswitch > quits with a CANCEL message. > i simply don't get why our freeswitch aborts the session - i think > it would work > if no "a=sendonly" would be present in the sdp. > > my technical contact doesn't want to switch 180 to 183 on the sonus > side - this would > also work (i think). in fact he says that 180 ringing is vaild, he > isn't that wrong in > this case. > > our freeswitch works in proxy mode, we do use trunk 15396 > see a ngrep trace under http://pastebin.freeswitch.org/11235 > > 92.63.208.36 - freeswitch > 38.105.229.100 - sonus > > br > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From robert.hadley at teotech.com Mon Nov 23 11:54:19 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Mon, 23 Nov 2009 11:54:19 -0800 Subject: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir In-Reply-To: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com> References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com> Message-ID: <8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com> Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test -d "$(switch_srcdir)/src/mod/$$confmoddir" ; then \ moddir = "$(switch_srcdir)/src/mod/$$confmoddir" ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test -f "$$moddir/Makefile" ; then \ <-- Yep, this will be true cd $$moddir && . && $(MAKE) I'm not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/4d01c933/attachment.html From anthony.minessale at gmail.com Mon Nov 23 11:57:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 13:57:43 -0600 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: <4B0266F4.8070602@savion.huji.ac.il> References: <4B0266F4.8070602@savion.huji.ac.il> Message-ID: <191c3a030911231157p44612c5dm3f0ee1e7b37e9cd3@mail.gmail.com> Let's just do this: r15629 or higher look for sip_profile_name On Tue, Nov 17, 2009 at 3:03 AM, Eli Hayun wrote: > Hi > We have more then one profile. To make a call I have to enter : bridge > sofia/profile/number at ip > The problem is when I use : "${use_profile}" I am getting the caller > profile, and I need the destination profile. > > How do I get this information? > > Thanks > > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/e961d695/attachment.html From msc at freeswitch.org Mon Nov 23 12:12:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 12:12:21 -0800 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> Message-ID: <87f2f3b90911231212x2467e0f3r44824e52f86773ea@mail.gmail.com> On Mon, Nov 23, 2009 at 11:24 AM, Phillip Jones wrote: > Hi there, > > I have created a simple conference that works great. The only problem is, > when a participant press # it exits the call. So when a user enters a > conference with a PIN, and by habit they enter 12345 followed by pound, it > puts them in and then straight out. > > So I edited conference.conf.xml so: > > > > and even assigned # to another function: > > > > and the same occurs. Pressing # exits the conference. > > What am I missing here? > > tia - phil > > > Phil, I recommend that you create a custom profile and a custom caller control group. Just copy the defaults and rename them to something meaningful. In conference.conf.xml you can add a new call control group like this: Then make a copy of the default profile changing the profile name and the caller-controls parameter: Give that a whirl and report back. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/1bd3bc70/attachment.html From pjintheusa at gmail.com Mon Nov 23 12:17:33 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 23 Nov 2009 15:17:33 -0500 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> Message-ID: <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> Thanks for replying. Well in the log I see: 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving conference, cause: NONE which make sense because just above I see: 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default caller control action 'hangup' bound to '#'. The question I have - is how do I change that default caller control action if it is not in conference.conf.xml ?? ... ** On Mon, Nov 23, 2009 at 2:35 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > issue > > console loglevel debug > from the cli > > then try again and see if there is any hint > > > On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones wrote: > >> Hi there, >> >> I have created a simple conference that works great. The only problem is, >> when a participant press # it exits the call. So when a user enters a >> conference with a PIN, and by habit they enter 12345 followed by pound, it >> puts them in and then straight out. >> >> So I edited conference.conf.xml so: >> >> >> >> and even assigned # to another function: >> >> >> >> and the same occurs. Pressing # exits the conference. >> >> What am I missing here? >> >> tia - phil >> >> >> >> Conf Setup: >> >> >> >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/a35891b1/attachment-0001.html From anthony.minessale at gmail.com Mon Nov 23 12:25:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 14:25:35 -0600 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> Message-ID: <191c3a030911231225i457a4329j3ec578d9f594db03@mail.gmail.com> see what happens if you set hangup to some other key or the word "event" On Mon, Nov 23, 2009 at 2:17 PM, Phillip Jones wrote: > Thanks for replying. > > Well in the log I see: > > 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 > 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving > conference, cause: NONE > > which make sense because just above I see: > > 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default > caller control action 'hangup' bound to '#'. > > The question I have - is how do I change that default caller control action > if it is not in conference.conf.xml ?? > > > > ... > > ** > > > On Mon, Nov 23, 2009 at 2:35 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> issue >> >> console loglevel debug >> from the cli >> >> then try again and see if there is any hint >> >> >> On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones wrote: >> >>> Hi there, >>> >>> I have created a simple conference that works great. The only problem is, >>> when a participant press # it exits the call. So when a user enters a >>> conference with a PIN, and by habit they enter 12345 followed by pound, it >>> puts them in and then straight out. >>> >>> So I edited conference.conf.xml so: >>> >>> >>> >>> and even assigned # to another function: >>> >>> >>> >>> and the same occurs. Pressing # exits the conference. >>> >>> What am I missing here? >>> >>> tia - phil >>> >>> >>> >>> Conf Setup: >>> >>> >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/37969033/attachment.html From msc at freeswitch.org Mon Nov 23 12:27:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 12:27:38 -0800 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> Message-ID: <87f2f3b90911231227s51f2a2f4r28ab93c77eb9ac61@mail.gmail.com> On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones wrote: > Thanks for replying. > > Well in the log I see: > > 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 > 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving > conference, cause: NONE > > which make sense because just above I see: > > 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default > caller control action 'hangup' bound to '#'. > > The question I have - is how do I change that default caller control action > if it is not in conference.conf.xml ?? > > > > ... > > ** > I believe that this is because the caller-controls param is commented out in the default profile config. I prefer not to mess w/ the default configs which is why I recommended the custom configs in my previous email... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/5ba7a184/attachment.html From Mailings at kh-dev.de Mon Nov 23 12:29:51 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Mon, 23 Nov 2009 21:29:51 +0100 Subject: [Freeswitch-users] FS dies after some minutes Message-ID: Hi, I did a new installation with the trunk from Saturday (21. Nov.) and it always dies with a core after 5-10 minutes. It happened several times. After that I did a new installation of 1.0.4 and this runs without problems on the same host. I'm using Ubuntu 8.04 Server with all patches. Anyone else experiencing this problem? Thanks, Klaus Here's the bt: #0 0x00007f7aa2eb22fc in sofia_reg_nonce_callback (pArg=0x40f3ca50, argc=, argv=0x7f7a9c006758, columnNames=) at ../../../../src/include/switch_utils.h:78 #1 0x00007f7aa91f4a12 in sqlite3_exec (db=0x7f7a9c00a6a0, zSql=0x7f7a9c006cd0 "select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8'", xCallback=0x7f7aa2eb22d0 , pArg=0x40f3ca50, pzErrMsg=0x40f3c680) at ./src/legacy.c:95 #2 0x00007f7aa917b98d in switch_core_db_exec (db=0x7f7a9c00a6a0, sql=0x7f7a9c006cd0 "select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8'", callback=0x7f7aa2eb22d0 , data=0x40f3ca50, errmsg=0x40f3c6e8) at src/switch_core_db.c:93 #3 0x00007f7aa2e985b1 in sofia_glue_execute_sql_callback (profile=0x72e940, mutex=0x0, sql=0x7f7a9c006cd0 "select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8'", callback=0x7f7aa2eb22d0 , pdata=0x40f3ca50) at sofia_glue.c:4297 #4 0x00007f7aa2ead8ec in sofia_reg_parse_auth (profile=0x72e940, authorization=0x7f7a9c078ad0, sip=0x7f7a9c0695d8, regstr=0x7f7aa2fe7137 "REGISTER", np=0x40f3d940 "b7ed6efa-d801-11de-a716-67cbb4a551f8", nplen=128, ip=0x40f3d840 "10.134.38.59", v_event=0x40f3d930, exptime=3600, regtype=REG_REGISTER, to_user=0x7f7a9c0dd18e "29", auth_params=0x40f3cd60, reg_count=0x40f3cd58) at sofia_reg.c:1704 #5 0x00007f7aa2eb004a in sofia_reg_handle_register (nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sip=0x7f7a9c0695d8, regtype=REG_REGISTER, key=0x40f3d940 "b7ed6efa-d801-11de-a716-67cbb4a551f8", keylen=0, v_event=0x40f3d930, is_nat=0x0) at sofia_reg.c:888 #6 0x00007f7aa2eb2f1c in sofia_reg_handle_sip_i_register (nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sofia_private=, sip=0x7f7a9c0695d8, tags=) at sofia_reg.c:1362 #7 0x00007f7aa2e9371c in sofia_event_callback (event=, status=100, phrase=0x7f7a9c071700 "Trying", nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sofia_private=0x0, sip=0x7f7a9c0695d8, tags=0x7f7a9c0716f0) at sofia.c:672 #8 0x00007f7aa2f1119e in nua_application_event (dummy=0x0, sumsg=0x40f3dd10, ee=0x7f7a9c0716c8) at nua_stack.c:393 #9 0x00007f7aa2f7e2c1 in su_base_port_execute_msgs (queue=0x0) at su_base_port.c:280 #10 0x00007f7aa2f7e039 in su_base_port_getmsgs (self=0x721320) at su_base_port.c:202 #11 0x00007f7aa2f7e5dc in su_base_port_step (self=0x721320, tout=0) at su_base_port.c:473 #12 0x00007f7aa2f7b68e in su_port_step (self=0x721320, tout=1000) at su_port.h:340 #13 0x00007f7aa2f7b656 in su_root_step (self=0x723320, tout=1000) at su_root.c:858 #14 0x00007f7aa2e8d40a in sofia_profile_thread_run (thread=, obj=) at sofia.c:1194 #15 0x00007f7aa88b63f7 in start_thread () from /lib/libpthread.so.0 #16 0x00007f7aa7e20b4d in clone () from /lib/libc.so.6 #17 0x0000000000000000 in ?? () -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/ca7d653d/attachment-0001.html From pjintheusa at gmail.com Mon Nov 23 12:32:39 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 23 Nov 2009 15:32:39 -0500 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <87f2f3b90911231212x2467e0f3r44824e52f86773ea@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <87f2f3b90911231212x2467e0f3r44824e52f86773ea@mail.gmail.com> Message-ID: <367751820911231232mea2e90dlf75590ca3f0ae839@mail.gmail.com> Michael that for the reply. I created a new group with # unbound and referenced it from the default profile: And that worked fine. Strangely though, changing the default group and referencing that from the default profile does not. Do you want me to test this on the latest trunk or is this as expected? Phil On Mon, Nov 23, 2009 at 3:12 PM, Michael Collins wrote: > > > On Mon, Nov 23, 2009 at 11:24 AM, Phillip Jones wrote: > >> Hi there, >> >> I have created a simple conference that works great. The only problem is, >> when a participant press # it exits the call. So when a user enters a >> conference with a PIN, and by habit they enter 12345 followed by pound, it >> puts them in and then straight out. >> >> So I edited conference.conf.xml so: >> >> >> >> and even assigned # to another function: >> >> >> >> and the same occurs. Pressing # exits the conference. >> >> What am I missing here? >> >> tia - phil >> >> >> > Phil, > > I recommend that you create a custom profile and a custom caller control > group. Just copy the defaults and rename them to something meaningful. In > conference.conf.xml you can add a new call control group like this: > > > digits="0"/> > > digits="*"/> > > digits="9"/> > > digits="8"/> > > digits="7"/> > > digits="3"/> > > digits="2"/> > > digits="1"/> > > digits="6"/> > > digits="5"/> > > digits="4"/> > > > > > Then make a copy of the default profile changing the profile name and the > caller-controls parameter: > > > > > > > > > > > Give that a whirl and report back. :) > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/5c8d6e2e/attachment.html From brian at freeswitch.org Mon Nov 23 12:34:36 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 14:34:36 -0600 Subject: [Freeswitch-users] FS dies after some minutes In-Reply-To: References: Message-ID: Update to SVN Trunk. /b On Nov 23, 2009, at 2:29 PM, Klaus Hochlehnert wrote: > Hi, > > I did a new installation with the trunk from Saturday (21. Nov.) and > it always dies with a core after 5-10 minutes. > It happened several times. > After that I did a new installation of 1.0.4 and this runs without > problems on the same host. > I?m using Ubuntu 8.04 Server with all patches. > > Anyone else experiencing this problem? > > Thanks, Klaus > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/968d050b/attachment.html From pjintheusa at gmail.com Mon Nov 23 12:42:50 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 23 Nov 2009 15:42:50 -0500 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <87f2f3b90911231227s51f2a2f4r28ab93c77eb9ac61@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> <87f2f3b90911231227s51f2a2f4r28ab93c77eb9ac61@mail.gmail.com> Message-ID: <367751820911231242o5d329480x5523a24696c6fa56@mail.gmail.com> Anthony - setting or does not make a difference, even when the default profile has un-commented. Looks to me like that default group is ignored even when specifically referred to? As Michael says though, creating a specific group: and adding in the default profile works a charm. I am good - but let me know if you want me to try anything else. Phil On Mon, Nov 23, 2009 at 3:27 PM, Michael Collins wrote: > > > On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones wrote: > >> Thanks for replying. >> >> Well in the log I see: >> >> 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 >> 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving >> conference, cause: NONE >> >> which make sense because just above I see: >> >> 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default >> caller control action 'hangup' bound to '#'. >> >> The question I have - is how do I change that default caller control >> action if it is not in conference.conf.xml ?? >> >> >> >> ... >> >> ** >> > > I believe that this is because the caller-controls param is commented out > in the default profile config. I prefer not to mess w/ the default configs > which is why I recommended the custom configs in my previous email... > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/1ec9d80b/attachment.html From christian.loeschenkohl at xpirio.com Mon Nov 23 12:54:21 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 23 Nov 2009 21:54:21 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <191c3a030911231145j384e5bbat7208633895b3b7af@mail.gmail.com> References: <4B0ADFE1.4070506@xpirio.com> <191c3a030911231145j384e5bbat7208633895b3b7af@mail.gmail.com> Message-ID: <4B0AF67D.6040707@xpirio.com> thank you for your answer the relevant part of the log is 2009-11-23 21:46:49.625130 [NOTICE] sofia.c:3693 Pre-Answer sofia/interconnect/24785214448370068 at 38.105.229.100! 2009-11-23 21:46:49.625130 [INFO] sofia.c:3706 Sending early media 2009-11-23 21:46:49.625130 [ERR] sofia_glue.c:2029 No audio codec available 2009-11-23 21:46:49.625130 [NOTICE] switch_channel.c:2048 Hangup sofia/interconnect/nobody at 81.94.55.100 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] it's the same with g729 and alaw (refering to brian) in my opinion the ringing here should be generated near end and no audio codec has to be used here (180 ringing) br On 2009-11-23 20:45, Anthony Minessale wrote: > you need to provide a FS console trace of your problem > > from your FS source dir (build root) > > cd libs/esl > make perlmod > cd perl > perl logger.pl -pb christian > > reproduce > > > then hit ctl-c and tell me the url it posted to. > > > > 2009/11/23 Christian L?schenkohl > > > hi > > our freeswitch server has to talk to a sonus ip-switch > when we want to setup a call we do get a "100 Trying" and then a > "180 Ringing" > within the "180 Ringing" we get a sdp with "a=sendonly" then our > freeswitch > quits with a CANCEL message. > i simply don't get why our freeswitch aborts the session - i think > it would work > if no "a=sendonly" would be present in the sdp. > > my technical contact doesn't want to switch 180 to 183 on the sonus > side - this would > also work (i think). in fact he says that 180 ringing is vaild, he > isn't that wrong in > this case. > > our freeswitch works in proxy mode, we do use trunk 15396 > see a ngrep trace under http://pastebin.freeswitch.org/11235 > > 92.63.208.36 - freeswitch > 38.105.229.100 - sonus > > br > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From christian.loeschenkohl at xpirio.com Mon Nov 23 12:56:15 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 23 Nov 2009 21:56:15 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> Message-ID: <4B0AF6EF.8070507@xpirio.com> thany ou for your answer we use g729 on all our other connections in passthrough mode and it also doesn't work with alaw. so i don't think it's related to this. br On 2009-11-23 20:48, Brian West wrote: > Well its also G729 so I suspect you don't have G729 > > /b > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > >> hi >> >> our freeswitch server has to talk to a sonus ip-switch >> when we want to setup a call we do get a "100 Trying" and then a >> "180 Ringing" >> within the "180 Ringing" we get a sdp with "a=sendonly" then our >> freeswitch >> quits with a CANCEL message. >> i simply don't get why our freeswitch aborts the session - i think >> it would work >> if no "a=sendonly" would be present in the sdp. >> >> my technical contact doesn't want to switch 180 to 183 on the sonus >> side - this would >> also work (i think). in fact he says that 180 ringing is vaild, he >> isn't that wrong in >> this case. >> >> our freeswitch works in proxy mode, we do use trunk 15396 >> see a ngrep trace under http://pastebin.freeswitch.org/11235 >> >> 92.63.208.36 - freeswitch >> 38.105.229.100 - sonus >> >> br >> >> -- >> Ing. Christian L?schenkohl >> Technische Leitung, Forschung& Entwicklung VoIP >> >> xpirio >> Telekommunikation& Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenkohl at xpirio.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From anthony.minessale at gmail.com Mon Nov 23 13:07:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 15:07:30 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <4B0AF6EF.8070507@xpirio.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> Message-ID: <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> do you have the ringback variable set on the channel? if so it will cause 180 to attempt to play inband ringback indication I have nothing left to say because I asked for the whole log with the siptrace enables not just 5 lines of it. If you still want help, give me the log to examine and I will tell you what your problem is. 2009/11/23 Christian L?schenkohl > thany ou for your answer > > we use g729 on all our other connections in passthrough mode and it also > doesn't work with alaw. > so i don't think it's related to this. > > br > > > On 2009-11-23 20:48, Brian West wrote: > > Well its also G729 so I suspect you don't have G729 > > > > /b > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > >> hi > >> > >> our freeswitch server has to talk to a sonus ip-switch > >> when we want to setup a call we do get a "100 Trying" and then a > >> "180 Ringing" > >> within the "180 Ringing" we get a sdp with "a=sendonly" then our > >> freeswitch > >> quits with a CANCEL message. > >> i simply don't get why our freeswitch aborts the session - i think > >> it would work > >> if no "a=sendonly" would be present in the sdp. > >> > >> my technical contact doesn't want to switch 180 to 183 on the sonus > >> side - this would > >> also work (i think). in fact he says that 180 ringing is vaild, he > >> isn't that wrong in > >> this case. > >> > >> our freeswitch works in proxy mode, we do use trunk 15396 > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > >> > >> 92.63.208.36 - freeswitch > >> 38.105.229.100 - sonus > >> > >> br > >> > >> -- > >> Ing. Christian L?schenkohl > >> Technische Leitung, Forschung& Entwicklung VoIP > >> > >> xpirio > >> Telekommunikation& Service GmbH > >> Lakeside B04 > >> 9020 Klagenfurt > >> Austria > >> > >> T +43 (0) 5 77 11 - 1000 > >> F +43 (0) 5 77 11 - 1002 > >> E christian.loeschenkohl at xpirio.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/12a10e08/attachment.html From rupa at rupa.com Mon Nov 23 13:08:55 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 23 Nov 2009 15:08:55 -0600 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231242o5d329480x5523a24696c6fa56@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> <87f2f3b90911231227s51f2a2f4r28ab93c77eb9ac61@mail.gmail.com> <367751820911231242o5d329480x5523a24696c6fa56@mail.gmail.com> Message-ID: The behavior of not being able to change the default caller controls are documented on the wiki: http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E *Reserved Group Names* - none - Use this name to prevent installing caller-controls for callers of a conference. - default - Use this name to utilize the hard-coded set of controls built-in to mod_conference. Do NOT name a custom set of conference-controls "default" as they will be overridden with the hard-coded set. The behavior of the "default" group is defined below: On Mon, Nov 23, 2009 at 2:42 PM, Phillip Jones wrote: > Anthony - setting > > > > or > > > > does not make a difference, even when the default profile has > > > > un-commented. > > > Looks to me like that default group is ignored even when specifically > referred to? > > As Michael says though, creating a specific group: > > > > and adding > > in the default profile > works a charm. > > I am good - but let me know if you want me to try anything else. > > Phil > > > > On Mon, Nov 23, 2009 at 3:27 PM, Michael Collins wrote: > >> >> >> On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones wrote: >> >>> Thanks for replying. >>> >>> Well in the log I see: >>> >>> 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 >>> 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving >>> conference, cause: NONE >>> >>> which make sense because just above I see: >>> >>> 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing >>> default caller control action 'hangup' bound to '#'. >>> >>> The question I have - is how do I change that default caller control >>> action if it is not in conference.conf.xml ?? >>> >>> >>> >>> ... >>> >>> ** >>> >> >> I believe that this is because the caller-controls param is commented out >> in the default profile config. I prefer not to mess w/ the default configs >> which is why I recommended the custom configs in my previous email... >> >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/f49bd641/attachment.html From mike at jerris.com Mon Nov 23 13:09:05 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 16:09:05 -0500 Subject: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir In-Reply-To: <8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com> References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com> <8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com> Message-ID: <99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com> In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: > Thanks Mike. > > modmake.rules is created in the $(switch_builddir)/build. > > What I see as the problem is in src/mod/Makefile.am > > There is a statement line 12 that points moddir to the source > if test ?d ?$(switch_srcdir)/src/mod/$$confmoddir? ; then \ > moddir = ?$(switch_srcdir)/src/mod/$$confmoddir? ; > > And then the statements starting around line 22 that cd to moddir (in src) and fire off make > if test ?f ?$$moddir/Makefile? ; then \ ? Yep, this will be true > cd $$moddir && ? && $(MAKE) > > I?m not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). > > Does anybody have a better idea? > > Thanks, > Robert > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Monday, November 23, 2009 11:16 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir > > The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. > > Mike > > On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: > > > I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/73d28cb1/attachment-0001.html From mike at jerris.com Mon Nov 23 13:14:27 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 16:14:27 -0500 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231242o5d329480x5523a24696c6fa56@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> <87f2f3b90911231227s51f2a2f4r28ab93c77eb9ac61@mail.gmail.com> <367751820911231242o5d329480x5523a24696c6fa56@mail.gmail.com> Message-ID: <0E108DAC-8A58-41D3-A194-F092AB4FBF87@jerris.com> Default controls are hard coded. If you want to change them you must use a name other than default. Mike On Nov 23, 2009, at 3:42 PM, Phillip Jones wrote: > Anthony - setting > > > > or > > > > does not make a difference, even when the default profile has > > > > un-commented. > > > Looks to me like that default group is ignored even when specifically referred to? > > As Michael says though, creating a specific group: > > > > and adding > > in the default profile works a charm. > > I am good - but let me know if you want me to try anything else. > > Phil > From robert.hadley at teotech.com Mon Nov 23 13:19:08 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Mon, 23 Nov 2009 13:19:08 -0800 Subject: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In-Reply-To: <99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com> References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com><8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com> <99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com> Message-ID: <6987A58F102E4AAE8A5F64581653ED78@greyhawk.tonecommander.com> In typical automake builds the configure step takes the Makefile.am from the srcdir and generates the Makefile in the builddir. Most src/mod subdirs are not using automake and/or configure. They just have a simple Makefile in with the source. Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 1:09 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test -d "$(switch_srcdir)/src/mod/$$confmoddir" ; then \ moddir = "$(switch_srcdir)/src/mod/$$confmoddir" ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test -f "$$moddir/Makefile" ; then \ <-- Yep, this will be true cd $$moddir && . && $(MAKE) I'm not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/a3fe6090/attachment.html From christian.loeschenkohl at xpirio.com Mon Nov 23 13:36:30 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 23 Nov 2009 22:36:30 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> Message-ID: <4B0B005E.4080202@xpirio.com> sorry about wasting your time (wasn't my intent) the log is at http://pastebin.freeswitch.org/11240 i called 5214448370068 (also other calls are in the log) they now have changed 180 to 183 on the sonus, but makes no difference here br On 2009-11-23 22:07, Anthony Minessale wrote: > do you have the ringback variable set on the channel? > if so it will cause 180 to attempt to play inband ringback indication > > I have nothing left to say because I asked for the whole log with the > siptrace enables not just 5 lines of it. > If you still want help, give me the log to examine and I will tell you > what your problem is. > > > > 2009/11/23 Christian L?schenkohl > > > thany ou for your answer > > we use g729 on all our other connections in passthrough mode and it > also doesn't work with alaw. > so i don't think it's related to this. > > br > > > On 2009-11-23 20:48, Brian West wrote: > > Well its also G729 so I suspect you don't have G729 > > > > /b > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > >> hi > >> > >> our freeswitch server has to talk to a sonus ip-switch > >> when we want to setup a call we do get a "100 Trying" and then a > >> "180 Ringing" > >> within the "180 Ringing" we get a sdp with "a=sendonly" then our > >> freeswitch > >> quits with a CANCEL message. > >> i simply don't get why our freeswitch aborts the session - i think > >> it would work > >> if no "a=sendonly" would be present in the sdp. > >> > >> my technical contact doesn't want to switch 180 to 183 on the sonus > >> side - this would > >> also work (i think). in fact he says that 180 ringing is vaild, he > >> isn't that wrong in > >> this case. > >> > >> our freeswitch works in proxy mode, we do use trunk 15396 > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > >> > >> 92.63.208.36 - freeswitch > >> 38.105.229.100 - sonus > >> > >> br > >> > >> -- > >> Ing. Christian L?schenkohl > >> Technische Leitung, Forschung& Entwicklung VoIP > >> > >> xpirio > >> Telekommunikation& Service GmbH > >> Lakeside B04 > >> 9020 Klagenfurt > >> Austria > >> > >> T +43 (0) 5 77 11 - 1000 > >> F +43 (0) 5 77 11 - 1002 > >> E christian.loeschenkohl at xpirio.com > > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From mike at jerris.com Mon Nov 23 13:42:24 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 16:42:24 -0500 Subject: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In-Reply-To: <6987A58F102E4AAE8A5F64581653ED78@greyhawk.tonecommander.com> References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com><8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com> <99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com> <6987A58F102E4AAE8A5F64581653ED78@greyhawk.tonecommander.com> Message-ID: I'll work on this, can you open me up a bug on http://jira.freeswitch.org in regards to this please. Mike On Nov 23, 2009, at 4:19 PM, Robert Hadley wrote: > In typical automake builds the configure step takes the Makefile.am from the srcdir and generates the Makefile in the builddir. > > Most src/mod subdirs are not using automake and/or configure. They just have a simple Makefile in with the source. > > Robert > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Monday, November 23, 2009 1:09 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir > > In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? > > Mike > > On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: > > > Thanks Mike. > > modmake.rules is created in the $(switch_builddir)/build. > > What I see as the problem is in src/mod/Makefile.am > > There is a statement line 12 that points moddir to the source > if test ?d ?$(switch_srcdir)/src/mod/$$confmoddir? ; then \ > moddir = ?$(switch_srcdir)/src/mod/$$confmoddir? ; > > And then the statements starting around line 22 that cd to moddir (in src) and fire off make > if test ?f ?$$moddir/Makefile? ; then \ ? Yep, this will be true > cd $$moddir && ? && $(MAKE) > > I?m not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). > > Does anybody have a better idea? > > Thanks, > Robert > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Monday, November 23, 2009 11:16 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir > > The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. > > Mike > > On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: > > > > I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/3704f258/attachment.html From rob4manhere at gmail.com Mon Nov 23 13:53:56 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Mon, 23 Nov 2009 15:53:56 -0600 Subject: [Freeswitch-users] Memory leak with mod_local_stream Message-ID: Hey guys, Having a problem with mod_local_stream. I recently did a "make current" from 15334 to the latest trunk (15630). After restarting, there now appears to be a memory leak. On a test system (CentOS 5.4, 64-bit) with no calls or registrations, Freeswitch gradually consumes all of the host memory (rate of about 200K/second), then swaps out, eventually rendering the system useless. I isolated it to mod_local_stream. If I unload mod_local_stream, the memory use stops climbing. If I re-load mod_local_stream, it starts again. I would submit the logs except they aren't any besides it starting. The system is just sitting there idle. Even valgrind didn't show much (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- check=full --leak-resolution=high --show-reachable=yes .libs/ freeswitch -vg Questions: * has anyone else seen this? * what is the best way I can assist troubleshooting this? I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could that have anything to do with it? Rob From jaybinks at gmail.com Mon Nov 23 14:12:07 2009 From: jaybinks at gmail.com (Jay Binks) Date: Tue, 24 Nov 2009 08:12:07 +1000 Subject: [Freeswitch-users] Memory leak with mod_local_stream In-Reply-To: References: Message-ID: <13396C7D-C89A-4ABC-A63F-EE5A3F8DBC50@gmail.com> if you suspect 15431 to have caused this, then revert to 15430 and see if the problem exists. if you can narrow do the bug to a specific svn revision, then you greatly assist in the resolution of the issue. apart from that im not much help sorry. maybe someone else can lab it up and see if they get the same result. ( Im on a train now, so not so easy :P ) J On 24/11/2009, at 7:53 AM, Rob Forman wrote: > Hey guys, > > Having a problem with mod_local_stream. > > I recently did a "make current" from 15334 to the latest trunk > (15630). After restarting, there now appears to be a memory leak. On > a test system (CentOS 5.4, 64-bit) with no calls or registrations, > Freeswitch gradually consumes all of the host memory (rate of about > 200K/second), then swaps out, eventually rendering the system useless. > > I isolated it to mod_local_stream. If I unload mod_local_stream, the > memory use stops climbing. If I re-load mod_local_stream, it starts > again. > > > I would submit the logs except they aren't any besides it starting. > The system is just sitting there idle. Even valgrind didn't show much > (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I > ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- > check=full --leak-resolution=high --show-reachable=yes .libs/ > freeswitch -vg > > Questions: > * has anyone else seen this? > * what is the best way I can assist troubleshooting this? > > I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could > that have anything to do with it? > > Rob > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Mon Nov 23 14:15:23 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 17:15:23 -0500 Subject: [Freeswitch-users] Memory leak with mod_local_stream In-Reply-To: References: Message-ID: <3BF17413-EC48-4691-8C99-61CC4661E2CA@jerris.com> That rev should have fixed that memory leak, could you test mod_local_stream.c from rev 15430 (http://fisheye.freeswitch.org/browse/~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/mod_local_stream.c) with your current fs version to confirm this is the cause please? Mike On Nov 23, 2009, at 4:53 PM, Rob Forman wrote: > Hey guys, > > Having a problem with mod_local_stream. > > I recently did a "make current" from 15334 to the latest trunk > (15630). After restarting, there now appears to be a memory leak. On > a test system (CentOS 5.4, 64-bit) with no calls or registrations, > Freeswitch gradually consumes all of the host memory (rate of about > 200K/second), then swaps out, eventually rendering the system useless. > > I isolated it to mod_local_stream. If I unload mod_local_stream, the > memory use stops climbing. If I re-load mod_local_stream, it starts > again. > > > I would submit the logs except they aren't any besides it starting. > The system is just sitting there idle. Even valgrind didn't show much > (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I > ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- > check=full --leak-resolution=high --show-reachable=yes .libs/ > freeswitch -vg > > Questions: > * has anyone else seen this? > * what is the best way I can assist troubleshooting this? > > I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could > that have anything to do with it? > > Rob > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From robert.hadley at teotech.com Mon Nov 23 14:21:08 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Mon, 23 Nov 2009 14:21:08 -0800 Subject: [Freeswitch-users] Building in a builddir using--srcdiroptionbut modules still build in srcdir In-Reply-To: References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com><8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com><99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com><6987A58F102E4AAE8A5F64581653ED78@greyhawk.tonecommander.com> Message-ID: Thanks Mike, How is the easy way to give you the changes I found so far? There are around 10 changes in 30 files (all configure.gnu files need a fix). Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 1:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using--srcdiroptionbut modules still build in srcdir I'll work on this, can you open me up a bug on http://jira.freeswitch.org in regards to this please. Mike On Nov 23, 2009, at 4:19 PM, Robert Hadley wrote: In typical automake builds the configure step takes the Makefile.am from the srcdir and generates the Makefile in the builddir. Most src/mod subdirs are not using automake and/or configure. They just have a simple Makefile in with the source. Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 1:09 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test -d "$(switch_srcdir)/src/mod/$$confmoddir" ; then \ moddir = "$(switch_srcdir)/src/mod/$$confmoddir" ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test -f "$$moddir/Makefile" ; then \ <-- Yep, this will be true cd $$moddir && . && $(MAKE) I'm not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/a3783727/attachment.html From brian at freeswitch.org Mon Nov 23 14:29:14 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 16:29:14 -0600 Subject: [Freeswitch-users] Building in a builddir using--srcdiroptionbut modules still build in srcdir In-Reply-To: References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com><8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com><99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com><6987A58F102E4AAE8A5F64581653ED78@greyhawk.tonecommander.com> Message-ID: <51270770-CA9E-4081-B28C-E11113EA4A04@freeswitch.org> go to the src root and type: svn diff > patch.diff then open a jira and attach patch.diff /b On Nov 23, 2009, at 4:21 PM, Robert Hadley wrote: > Thanks Mike, > > How is the easy way to give you the changes I found so far? There > are around 10 changes in 30 files (all configure.gnu files need a > fix). > > Robert > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/f4cadce8/attachment-0001.html From pjintheusa at gmail.com Mon Nov 23 14:34:27 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 23 Nov 2009 17:34:27 -0500 Subject: [Freeswitch-users] register timeout / cisco 7960 Message-ID: <367751820911231434j36b9846dk46d058ddb77c634@mail.gmail.com> hi there, I have set up some cisco 7960 up with fs. They work fine - but the only way I can keep them registered is to set the "timer_register_expires" in the Cisco cfg file to something really short like 10s. Does anyone know the default register timeout for fs? And where I might change this in fs? Thanks! Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/72fd7e4b/attachment.html From lists at redbonez.net Mon Nov 23 14:49:03 2009 From: lists at redbonez.net (Adam Ford) Date: Mon, 23 Nov 2009 15:49:03 -0700 Subject: [Freeswitch-users] FIFO Orgination_caller_id Message-ID: <005701ca6c8f$28eaa570$7abff050$@net> Is there any way to set the origination_caller_id for a FIFO outbound call to an on-hook agent? I can't find anything in the wiki about a FIFO or member variable to set this. It seems to be set to 'Queue' by default, and appears to be hardcoded in the module source. It would be nice to be able to change per FIFO queue. That way agents that handle multiple companies can more easily see which queue is calling and answer accordingly. It is not a big deal, since it does automatically set the origination_caller_id_number to 'fifo+'. However, depending on the phone, the caller ID number is not always readily shown, and must be looked for. Thanks to anyone who has some insight on this, -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/9d905f1e/attachment.html From siniypin at gmail.com Mon Nov 23 14:51:17 2009 From: siniypin at gmail.com (RobertT) Date: Tue, 24 Nov 2009 01:51:17 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> <2160023e0911211523k7998d048nced3af8fb805e770@mail.gmail.com> Message-ID: <2160023e0911231451v59c072adr126584534b1e4f76@mail.gmail.com> OK, this is what I've got. First, I've updated FreeSwitch from trunk to version 15630 and deployed it to my server. Performed a tets and again no magic happened. The link to SIP trace is below. Then I've installed 1.0.4 version to another server (virtual hosting), and performed tha same. And everything went OK. This server's log is below as well. Not working - http://pastebin.com/m2e97985d Working - http://pastebin.com/m3c1e6bfe Also in both cases there is a strange detail - clients' SIP ports are configured to be 5060 and 5061, but what can be seen in trace differs from these values whereas stun resolution shows that there is no NAT (clients connect with ADSL modem). Regards, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/d4d4d38e/attachment.html From rob4manhere at gmail.com Mon Nov 23 15:04:28 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Mon, 23 Nov 2009 17:04:28 -0600 Subject: [Freeswitch-users] Memory leak with mod_local_stream In-Reply-To: <3BF17413-EC48-4691-8C99-61CC4661E2CA@jerris.com> References: <3BF17413-EC48-4691-8C99-61CC4661E2CA@jerris.com> Message-ID: I tried mod_local_stream.c from rev 15430, did a make clean && make all && make install-- but it didn't fix it so it wasn't that patch. I'll make current and try valgrind again unless someone has other ideas. Rob On Nov 23, 2009, at 4:15 PM, Michael Jerris wrote: > That rev should have fixed that memory leak, could you test > mod_local_stream.c from rev 15430 (http://fisheye.freeswitch.org/browse/ > ~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/ > mod_local_stream.c) with your current fs version to confirm this is > the cause please? > > Mike > > > On Nov 23, 2009, at 4:53 PM, Rob Forman wrote: > >> Hey guys, >> >> Having a problem with mod_local_stream. >> >> I recently did a "make current" from 15334 to the latest trunk >> (15630). After restarting, there now appears to be a memory leak. >> On >> a test system (CentOS 5.4, 64-bit) with no calls or registrations, >> Freeswitch gradually consumes all of the host memory (rate of about >> 200K/second), then swaps out, eventually rendering the system >> useless. >> >> I isolated it to mod_local_stream. If I unload mod_local_stream, the >> memory use stops climbing. If I re-load mod_local_stream, it starts >> again. >> >> >> I would submit the logs except they aren't any besides it starting. >> The system is just sitting there idle. Even valgrind didn't show >> much >> (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I >> ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- >> check=full --leak-resolution=high --show-reachable=yes .libs/ >> freeswitch -vg >> >> Questions: >> * has anyone else seen this? >> * what is the best way I can assist troubleshooting this? >> >> I saw a patch to mod_local_stream (rev 15431) a few weeks back. >> Could >> that have anything to do with it? >> >> Rob >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From siniypin at gmail.com Mon Nov 23 15:09:21 2009 From: siniypin at gmail.com (RobertT) Date: Tue, 24 Nov 2009 02:09:21 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911231451v59c072adr126584534b1e4f76@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> <2160023e0911211523k7998d048nced3af8fb805e770@mail.gmail.com> <2160023e0911231451v59c072adr126584534b1e4f76@mail.gmail.com> Message-ID: <2160023e0911231509i6f3ebc2er30d1a24c8d25d1d3@mail.gmail.com> You know what, guys? I've just made it working be opening ALL tcp trafic in and out from server by adding two match-all ip filters into local security policy. I can't say I like this solution... Why did this problem appeared with policy matching exact (sofia profiles) ports? Regards, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/6fcc18ce/attachment.html From anthony.minessale at gmail.com Mon Nov 23 16:13:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 18:13:11 -0600 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <005701ca6c8f$28eaa570$7abff050$@net> References: <005701ca6c8f$28eaa570$7abff050$@net> Message-ID: <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> if you add {origination_caller_id_name=foo,origination_caller_id_number=123} before the static entries for the on hook agent it will prevail over the default one. If you are using 1.0.4, this feature is only available in trunk or one of the 1.0.5 pre releases. On Mon, Nov 23, 2009 at 4:49 PM, Adam Ford wrote: > Is there any way to set the origination_caller_id for a FIFO outbound > call to an on-hook agent? I can?t find anything in the wiki about a FIFO or > member variable to set this. It seems to be set to ?Queue? by default, and > appears to be hardcoded in the module source. It would be nice to be able > to change per FIFO queue. That way agents that handle multiple companies > can more easily see which queue is calling and answer accordingly. > > > > It is not a big deal, since it does automatically set the > origination_caller_id_number to ?fifo+?. However, depending on > the phone, the caller ID number is not always readily shown, and must be > looked for. > > > > Thanks to anyone who has some insight on this, > > -Adam > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/d986f313/attachment.html From msc at freeswitch.org Mon Nov 23 16:16:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 16:16:03 -0800 Subject: [Freeswitch-users] conference digits and conference control In-Reply-To: <200910151944464068246@gmail.com> References: <200910151944464068246@gmail.com> Message-ID: <87f2f3b90911231616i2ebf92ccs485d66f29f00c4f5@mail.gmail.com> On Thu, Oct 15, 2009 at 3:44 AM, god.nirvana wrote: > hi all > how can i get the digits when users in the conference?? > and,in conference.conf.xml > the "action" will set another > value?e.g:transfer? > thanks > > I'm not sure I understand your question, but the wiki covers actions on keystrokes. If you need the user to dial other digits after the caller control then route the call to an extension that asks the user for input, like with play_and_get_digits, and handle the call accordingly. As far as the question about about setting another value - can you expound upon that a bit? I'm not sure what you're trying to accomplish. -MC P.S. - http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/94a4d0c3/attachment-0001.html From abeka at greatiam.com Mon Nov 23 16:18:29 2009 From: abeka at greatiam.com (Otis) Date: Tue, 24 Nov 2009 00:18:29 +0000 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: <4B0ABC4F.1010103@greatiam.com> References: <4B086689.6080804@greatiam.com> <4B097A89.2050400@greatiam.com> <4B0ABC4F.1010103@greatiam.com> Message-ID: <4B0B2655.4010900@greatiam.com> Has anyone got any suggestion how I can set up a gateway to receive incoming call on extension 1001 please. Any generic conf file will do. my username with my gateway is s=say " qwerty" and password "ytrewq" I have used the intruction from the link below without success. Thanks. Otis wrote: > Hello > > Could anyone point out what I have missed please ? > At the moment I configured a gateway voipuser as described here > : > Any suggestion as to what path I can take will be highly welcome > > Thanks > . > > > > > Sam Abekah-Mensah wrote: >>
Hi Michael >> >> Thanks >> >> I had set it to send incoming calls to extension 1001. This is in the >> file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory. >> The contents are : >> >> >> >> >> >> >> >> >> Is there >> anything wrong with this please ? >> >> Thanks >> >> >> >> Michal Bielicki wrote: >>> >>> Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah: >>> >>>> >>>> I need help as I cannot receive calls through VOIPUSER. This is a >>>> learning setup Attached are my conf files. What is wrong with them >>>> ? When I dial from a landline I get a continuous beep. >>>> >>>> Attached are my gateway and the conf file to transfer. Sopfia >>>> Status is my screen message. I can see a FAIL and cannot make head >>>> or tail of all that message. Hopefully anyone using voipuser or in >>>> fact any of you clever folks can make sense of this. >>>> >>>> Thanks for your time. >>>> >>>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy >>>> RFC2833 Mode! >>>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec >>>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>> 2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec >>>> Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] >>>> 2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec >>>> sofia/external/nobody at 213.166.5.133 PCMA/8000 20 ms 160 samples >>>> 2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf >>>> payload to 101 >>>> 2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 >>>> (sofia/external/nobody at 213.166.5.133) State Change CS_NEW -> CS_INIT >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_INIT >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>>> (sofia/external/nobody at 213.166.5.133) State INIT >>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 >>>> sofia/external/nobody at 213.166.5.133 SOFIA INIT >>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 >>>> (sofia/external/nobody at 213.166.5.133) State Change CS_INIT -> >>>> CS_ROUTING >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>>> (sofia/external/nobody at 213.166.5.133) State INIT going to sleep >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_ROUTING >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 >>>> (sofia/external/nobody at 213.166.5.133) State ROUTING >>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 >>>> sofia/external/nobody at 213.166.5.133 SOFIA ROUTING >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 >>>> sofia/external/nobody at 213.166.5.133 Standard ROUTING >>>> 2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing >>>> anonymous->abeka in context public >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->unloop] continue=false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (PASS) [unloop] >>>> ${unroll_loops}(true) =~ /^true$/ break=on-false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [unloop] >>>> ${sip_looped_call}() =~ /^true$/ break=on-false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->outside_call] continue=true >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Absolute Condition >>>> [outside_call] >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Action >>>> set(outside_call=true) >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->call_debug] continue=true >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->public_extensions] continue=false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>> [public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/ >>>> break=on-false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->public_did] continue=false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>> [public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->sip at sip.voipuser.org] continue=false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>> [sip at sip.voipuser.org] destination_number(abeka) =~ /08715042951/ >>>> break=on-false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->Inbound-abeka at sip.voipuser.org]] continue=false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>> [Inbound-abeka at sip.voipuser.org]] destination_number(abeka) =~ >>>> /[08444846450]/ break=on-false >>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 >>>> (sofia/external/nobody at 213.166.5.133) State Change CS_ROUTING -> >>>> CS_EXECUTE >>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send >>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 >>>> (sofia/external/nobody at 213.166.5.133) State ROUTING going to sleep >>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 >>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_EXECUTE >>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 >>>> (sofia/external/nobody at 213.166.5.133) State EXECUTE >>>> 2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 >>>> sofia/external/nobody at 213.166.5.133 SOFIA EXECUTE >>>> 2009-11-21 22:07:15.706658 [DEBUG] switch_core_state_machine.c:151 >>>> sofia/external/nobody at 213.166.5.133 Standard EXECUTE >>>> EXECUTE sofia/external/nobody at 213.166.5.133 set(outside_call=true) >>>> 2009-11-21 22:07:15.728613 [DEBUG] mod_dptools.c:748 >>>> sofia/external/nobody at 213.166.5.133 SET [outside_call]=[true] >>>> 2009-11-21 22:07:15.728613 [NOTICE] switch_core_state_machine.c:179 >>>> Hangup sofia/external/nobody at 213.166.5.133 [CS_EXECUTE] >>>> [NORMAL_CLEARING] >>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_channel.c:1683 Send >>>> signal sofia/external/nobody at 213.166.5.133 [KILL] >>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_session.c:932 Send >>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:491 >>>> (sofia/external/nobody at 213.166.5.133) State EXECUTE going to sleep >>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:398 >>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_HANGUP >>>> 2009-11-21 22:07:15.735830 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/external/nobody at 213.166.5.133) State HANGUP >>>> 2009-11-21 22:07:15.735830 [DEBUG] mod_sofia.c:338 Channel >>>> sofia/external/nobody at 213.166.5.133 hanging up, cause: NORMAL_CLEARING >>>> 2009-11-21 22:07:15.737680 [DEBUG] mod_sofia.c:417 Responding to >>>> INVITE with: 480 >>>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:46 >>>> sofia/external/nobody at 213.166.5.133 Standard HANGUP, cause: >>>> NORMAL_CLEARING >>>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/external/nobody at 213.166.5.133) State HANGUP going to sleep >>>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_state_machine.c:476 >>>> (sofia/external/nobody at 213.166.5.133) State Change CS_HANGUP -> >>>> CS_REPORTING >>>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_session.c:932 Send >>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:398 >>>> (sofia/external/nobody at 213.166.5.133) Running State Change >>>> CS_REPORTING >>>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:612 >>>> (sofia/external/nobody at 213.166.5.133) State REPORTING >>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:53 >>>> sofia/external/nobody at 213.166.5.133 Standard REPORTING, cause: >>>> NORMAL_CLEARING >>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:612 >>>> (sofia/external/nobody at 213.166.5.133) State REPORTING going to sleep >>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:411 >>>> (sofia/external/nobody at 213.166.5.133) State Change CS_REPORTING -> >>>> CS_DESTROY >>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_session.c:1068 >>>> Session 2 (sofia/external/nobody at 213.166.5.133) Locked, Waiting on >>>> external entities >>>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1086 >>>> Session 2 (sofia/external/nobody at 213.166.5.133) Ended >>>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1088 >>>> Close Channel sofia/external/nobody at 213.166.5.133 [CS_DESTROY] >>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>>> (sofia/external/nobody at 213.166.5.133) State DESTROY >>>> 2009-11-21 22:07:15.802636 [DEBUG] mod_sofia.c:255 >>>> sofia/external/nobody at 213.166.5.133 SOFIA DESTROY >>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:60 >>>> sofia/external/nobody at 213.166.5.133 Standard DESTROY >>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>>> (sofia/external/nobody at 213.166.5.133) State DESTROY going to sleep >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> : >>> >>> >>> you seem to have not specified an extension where the call should go to >>> my voipuser.org setup looks like: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I am also surprised that your setup works with a from-domain of >>> sip.voipuser.org >>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>> >>> *Michal Bielicki* >>> HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa >>> t. +48228753290 | f. +48228753291 michal.bielicki at halokwadrat.pl >>> | w. >>> www.halokwadrat.pl >>> >>> >>> >>> *Knowledge & Low Prices. Guaranteed!* >>> >> >> >> >>
> > From john_platts at hotmail.com Mon Nov 23 16:19:59 2009 From: john_platts at hotmail.com (John Platts) Date: Mon, 23 Nov 2009 18:19:59 -0600 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH Message-ID: I actually checked out the latest version of FreeSWITCH in the SVN repository. I have the following configured in /usr/local/freeswitch/conf/dialplan/default.xml: ??? ??????? ??????????? ??????????? ??????????? ??????????? ??????? ??? I have the following configured in /usr/local/freeswitch/conf/vars.xml: ? ? Here is the SIP trace for the failing call: Nov 23 17:55:05.245 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:+19725357722 at ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24 Record-Route: v: SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236 record-route: f: ;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a t: i: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585 CSeq: 1 INVITE Max-Forwards: 16 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208 m: c: application/SDP l: 210 P-Asserted-Identity: Privacy: none v=0 o=- 641026559 641026559 IN IP4 199.173.111.147 s=- c=IN IP4 199.173.111.147 t=0 0 m=audio 33344 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 17:55:05.257 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208 From: ;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a To: Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 17:55:05.257 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357722 at 168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: ;tag=105BD148-201C To: Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 Supported: timer,resource-priority,replaces Min-SE:? 1800 Cisco-Guid: 1961129755-3619819998-2727664095-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259020505 Contact: Expires: 180 Allow-Events: telephone-event Max-Forwards: 15 P-Asserted-Identity: Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 5041 5861 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.147 t=0 0 m=audio 33344 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.147 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 17:55:05.261 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: ;tag=105BD148-201C To: Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Timestamp: 1259020505 0.000345 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Content-Length: 0 Nov 23 17:55:05.309 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 P-Asserted-Identity: "19725357722" Nov 23 17:55:05.309 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208 From: ;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a To: ;tag=105BD180-BD7 Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: Record-Route: , Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 202 P-Asserted-Identity: "19725357722" v=0 o=- 211627 211627 IN IP4 192.168.1.4 s=- c=IN IP4 173.57.44.212 t=0 0 m=audio 0 RTP/AVP 96 101 a=rtpmap:96 G729a/8000 a=fmtp:96 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357722 at 168.75.202.212:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659B6A From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:19725357722 at 168.75.202.212:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 P-Asserted-Identity: Timestamp: 1259020508 CSeq: 102 BYE Reason: Q.850;cause=65 Content-Length: 0 Nov 23 17:55:08.401 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 CSeq: 102 BYE Timestamp: 1259020508 0.000093 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Content-Length: 0 Nov 23 17:55:08.401 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 CSeq: 102 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 The SIP call actually fails. If I remove the following from /usr/local/freeswitch/conf/dialplan/default.xml: ??? ??????? ??????????? ??????????? ??????????? ??????????? ??????? ??? And I change this line in /usr/local/freeswitch/conf/vars.xml from to And I change this line in /usr/local/freeswitch/conf/vars.xml from to both inbound and outbound calls succeed. Here is a SIP trace of a successful call after I apply the above changes: Nov 23 18:16:51.844 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:+19725357722 at ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb Record-Route: v: SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205 record-route: f: ;tag=dc7-13c4-3da425-183eff65-3da425 t: i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Max-Forwards: 16 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 m: c: application/SDP l: 210 P-Asserted-Identity: Privacy: none v=0 o=- 654094598 654094598 IN IP4 199.173.111.138 s=- c=IN IP4 199.173.111.138 t=0 0 m=audio 31456 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 18:16:51.852 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 From: ;tag=dc7-13c4-3da425-183eff65-3da425 To: Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 18:16:51.856 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357722 at 168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: ;tag=106FC130-1578 To: Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 Supported: timer,resource-priority,replaces Min-SE:? 1800 Cisco-Guid: 2142226702-3620016606-2730940895-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259021811 Contact: Expires: 180 Allow-Events: telephone-event Max-Forwards: 15 P-Asserted-Identity: Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 9668 3852 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.138 t=0 0 m=audio 31456 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.138 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 18:16:51.856 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: ;tag=106FC130-1578 To: Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Timestamp: 1259021811 0.000356 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Content-Length: 0 Nov 23 18:16:51.908 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: ;tag=106FC130-1578 To: ;tag=mXNXUN859rKBa Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 P-Asserted-Identity: "19725357722" Nov 23 18:16:51.908 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 From: ;tag=dc7-13c4-3da425-183eff65-3da425 To: ;tag=106FC168-50D Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: Record-Route: , Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 18:16:54.408 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: ;tag=106FC130-1578 To: ;tag=mXNXUN859rKBa Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 251 P-Asserted-Identity: "19725357722" v=0 o=FreeSWITCH 1259003870 1259003871 IN IP4 168.75.202.212 s=FreeSWITCH c=IN IP4 168.75.202.212 t=0 0 m=audio 17544 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 Nov 23 18:16:54.412 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357722 at 168.75.202.212:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AE109A From: ;tag=106FC130-1578 To: ;tag=mXNXUN859rKBa Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 Nov 23 18:16:54.412 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 From: ;tag=dc7-13c4-3da425-183eff65-3da425 To: ;tag=106FC168-50D Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: Record-Route: , Supported: replaces Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 253 v=0 o=CiscoSystemsSIP-GW-UserAgent 7353 3710 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 168.75.202.212 t=0 0 m=audio 17544 RTP/AVP 0 101 c=IN IP4 168.75.202.212 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Nov 23 18:16:54.492 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: ACK sip:19725357722 at 168.75.202.246:5060 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKb17fac77c446113b9154e16639d30287.6be1820d v: SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK99af095cadf31f4291c6a809ef6a6e03.7c44d9fc;received=65.217.40.205 f: ;tag=dc7-13c4-3da425-183eff65-3da425 t: ;tag=106FC168-50D i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 ACK user-agent: CS2000_NGSS/9.0 Max-Forwards: 68 k: 100rel,replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da428-f0c93cc6-61239372;received=199.173.101.208 m: l: 0 Nov 23 18:17:00.636 CST: %FAN-3-FAN_FAILED: Fan 1 had a rotation error reported. Nov 23 18:17:06.748 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: BYE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKmNaaUQK5vrpXQ Max-Forwards: 70 From: ;tag=mXNXUN859rKBa To: ;tag=106FC130-1578 Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 CSeq: 123392377 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Nov 23 18:17:06.748 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKmNaaUQK5vrpXQ From: ;tag=mXNXUN859rKBa To: ;tag=106FC130-1578 Date: Tue, 24 Nov 2009 00:17:06 GMT Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 Server: Cisco-SIPGateway/IOS-12.x CSeq: 123392377 BYE Reason: Q.850;cause=16 Content-Length: 0 Nov 23 18:17:06.752 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:199.173.101.208:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AF4FA From: ;tag=106FC168-50D To: ;tag=dc7-13c4-3da425-183eff65-3da425 Date: Tue, 24 Nov 2009 00:16:54 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Route: , Timestamp: 1259021826 CSeq: 101 BYE Reason: Q.850;cause=16 Content-Length: 0 Nov 23 18:17:06.824 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK f: ;tag=106FC168-50D t: ;tag=dc7-13c4-3da425-183eff65-3da425 i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 101 BYE server: CS2000_NGSS/9.0 allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP 168.75.202.246:5060;received=168.75.202.246;branch=z9hG4bK65AF4FA l: 0 I compared the SIP messaging from the failed call to the SIP messaging from the good call. Both calls are inbound calls. Here is the session description for the failed inbound call: v=0 o=- 211627 211627 IN IP4 192.168.1.4 s=- c=IN IP4 173.57.44.212 t=0 0 m=audio 0 RTP/AVP 96 101 a=rtpmap:96 G729a/8000 a=fmtp:96 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Here is the session description for the good outbound call: v=0 o=FreeSWITCH 1259003870 1259003871 IN IP4 168.75.202.212 s=FreeSWITCH c=IN IP4 168.75.202.212 t=0 0 m=audio 17544 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 Here are the differences between the session descriptors of the failed call and the good call: - The c= line has the correct IP address for the failed call, which was using media bypass - The c= line has the correct IP address for the good call, because the media is being processed by FreeSWITCH in the good call - The m= line does not have the correct RTP port in the failed call - The m= line has the correct RTP port in the good call I noticed that the SDP media descriptor is incorrect in the failed call. Has this problem been fixed? I am running revision 15586 from the FreeSWITCH SVN trunk. _________________________________________________________________ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/ From brian at freeswitch.org Mon Nov 23 16:25:44 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 18:25:44 -0600 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH In-Reply-To: References: Message-ID: <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org> What rev exactly? /b On Nov 23, 2009, at 6:19 PM, John Platts wrote: > > I actually checked out the latest version of FreeSWITCH in the SVN > repository. > > I have the following configured in /usr/local/freeswitch/conf/ > dialplan/default.xml: > > > > > > > > From msc at freeswitch.org Mon Nov 23 16:28:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 16:28:08 -0800 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: <4B0B2655.4010900@greatiam.com> References: <4B086689.6080804@greatiam.com> <4B097A89.2050400@greatiam.com> <4B0ABC4F.1010103@greatiam.com> <4B0B2655.4010900@greatiam.com> Message-ID: <87f2f3b90911231628t7e44986ar10451f26a86e7df6@mail.gmail.com> On Mon, Nov 23, 2009 at 4:18 PM, Otis wrote: > Has anyone got any suggestion how I can set up a gateway to receive > incoming call on extension 1001 please. > > Any generic conf file will do. my username with my gateway is s=say " > qwerty" and password "ytrewq" > > I have used the intruction from the link below without success. > > Thanks. > > Get a debug log of an incoming call. Most likely it is hitting the public context and you don't have it properly routed. Look in conf/dialplan/public.xml for an example of how to route to a specific extension. Another example is in conf/dialplan/public/00_inbound_did.xml. The trick is to know what to match on in your condition tag. In a pinch you could route all incoming calls to the info app and then make a test call to see what's coming down the line. Just be sure to turn it off when you're done testing!! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/15d909d0/attachment.html From lists at redbonez.net Mon Nov 23 16:43:42 2009 From: lists at redbonez.net (Adam Ford) Date: Mon, 23 Nov 2009 17:43:42 -0700 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> Message-ID: <009201ca6c9f$2d18cb80$874a6280$@net> I actually tried that, as a guess, based on the configuration output of fifo list. However I am running a tarball release of 1.0.4, which would explain why it did not work for me. I appreciate the feedback, and will make a note to implement this when I update my installation. Are the svn-trunk updates pretty solid? I have not attempted an update yet, as it is a production system. Thanks, -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, November 23, 2009 5:13 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FIFO Orgination_caller_id if you add {origination_caller_id_name=foo,origination_caller_id_number=123} before the static entries for the on hook agent it will prevail over the default one. If you are using 1.0.4, this feature is only available in trunk or one of the 1.0.5 pre releases. On Mon, Nov 23, 2009 at 4:49 PM, Adam Ford wrote: Is there any way to set the origination_caller_id for a FIFO outbound call to an on-hook agent? I can't find anything in the wiki about a FIFO or member variable to set this. It seems to be set to 'Queue' by default, and appears to be hardcoded in the module source. It would be nice to be able to change per FIFO queue. That way agents that handle multiple companies can more easily see which queue is calling and answer accordingly. It is not a big deal, since it does automatically set the origination_caller_id_number to 'fifo+'. However, depending on the phone, the caller ID number is not always readily shown, and must be looked for. Thanks to anyone who has some insight on this, -Adam _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/2fe06353/attachment.html From anthony.minessale at gmail.com Mon Nov 23 16:48:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 18:48:51 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <4B0B005E.4080202@xpirio.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> <4B0B005E.4080202@xpirio.com> Message-ID: <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> You forgot to set freeswitch to debug loglevel You need both of the following: console loglevel debug sofia profile internal siptrace on 2009/11/23 Christian L?schenkohl > sorry about wasting your time (wasn't my intent) > > the log is at http://pastebin.freeswitch.org/11240 > i called 5214448370068 (also other calls are in the log) > > they now have changed 180 to 183 on the sonus, but makes no difference here > > br > > On 2009-11-23 22:07, Anthony Minessale wrote: > > do you have the ringback variable set on the channel? > > if so it will cause 180 to attempt to play inband ringback indication > > > > I have nothing left to say because I asked for the whole log with the > > siptrace enables not just 5 lines of it. > > If you still want help, give me the log to examine and I will tell you > > what your problem is. > > > > > > > > 2009/11/23 Christian L?schenkohl > > > > > > thany ou for your answer > > > > we use g729 on all our other connections in passthrough mode and it > > also doesn't work with alaw. > > so i don't think it's related to this. > > > > br > > > > > > On 2009-11-23 20:48, Brian West wrote: > > > Well its also G729 so I suspect you don't have G729 > > > > > > /b > > > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > > > >> hi > > >> > > >> our freeswitch server has to talk to a sonus ip-switch > > >> when we want to setup a call we do get a "100 Trying" and then a > > >> "180 Ringing" > > >> within the "180 Ringing" we get a sdp with "a=sendonly" then our > > >> freeswitch > > >> quits with a CANCEL message. > > >> i simply don't get why our freeswitch aborts the session - i > think > > >> it would work > > >> if no "a=sendonly" would be present in the sdp. > > >> > > >> my technical contact doesn't want to switch 180 to 183 on the > sonus > > >> side - this would > > >> also work (i think). in fact he says that 180 ringing is vaild, > he > > >> isn't that wrong in > > >> this case. > > >> > > >> our freeswitch works in proxy mode, we do use trunk 15396 > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > > >> > > >> 92.63.208.36 - freeswitch > > >> 38.105.229.100 - sonus > > >> > > >> br > > >> > > >> -- > > >> Ing. Christian L?schenkohl > > >> Technische Leitung, Forschung& Entwicklung VoIP > > >> > > >> xpirio > > >> Telekommunikation& Service GmbH > > >> Lakeside B04 > > >> 9020 Klagenfurt > > >> Austria > > >> > > >> T +43 (0) 5 77 11 - 1000 > > >> F +43 (0) 5 77 11 - 1002 > > >> E christian.loeschenkohl at xpirio.com > > > > >> > > >> _______________________________________________ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch- > > >> users > > >> http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung & Entwicklung VoIP > > > > xpirio > > Telekommunikation & Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/5d1a239e/attachment-0001.html From dujinfang at gmail.com Mon Nov 23 16:58:39 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Nov 2009 08:58:39 +0800 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> Message-ID: <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> And because it's static string for on-hook members, it's hard to set dynamically. For now, I'm using a callback way - whenever the sip client answered the call, it fetch the real connected number from a http server. That's not ideal because not only it add the complexity but also the callee have no idea what the number is before answer. The problem for on-hook agent is that it call the agent first, and then get one customer from the fifo queue, so it is not possible to let the agent know the real caller-id before answer. Ideas? 2009/11/24 Anthony Minessale > if you add > {origination_caller_id_name=foo,origination_caller_id_number=123} before the > static entries for the on hook agent it will prevail over the default one. > > If you are using 1.0.4, this feature is only available in trunk or one of > the 1.0.5 pre releases. > > > On Mon, Nov 23, 2009 at 4:49 PM, Adam Ford wrote: > >> Is there any way to set the origination_caller_id for a FIFO outbound >> call to an on-hook agent? I can?t find anything in the wiki about a FIFO or >> member variable to set this. It seems to be set to ?Queue? by default, and >> appears to be hardcoded in the module source. It would be nice to be able >> to change per FIFO queue. That way agents that handle multiple companies >> can more easily see which queue is calling and answer accordingly. >> >> >> >> It is not a big deal, since it does automatically set the >> origination_caller_id_number to ?fifo+?. However, depending on >> the phone, the caller ID number is not always readily shown, and must be >> looked for. >> >> >> >> Thanks to anyone who has some insight on this, >> >> -Adam >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/ccb99647/attachment.html From msc at freeswitch.org Mon Nov 23 17:01:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 17:01:51 -0800 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <009201ca6c9f$2d18cb80$874a6280$@net> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <009201ca6c9f$2d18cb80$874a6280$@net> Message-ID: <87f2f3b90911231701n7dd58b6fhe354e04890ada239@mail.gmail.com> On Mon, Nov 23, 2009 at 4:43 PM, Adam Ford wrote: > I actually tried that, as a guess, based on the configuration output of > fifo list. However I am running a tarball release of 1.0.4, which would > explain why it did not work for me. > > > > I appreciate the feedback, and will make a note to implement this when I > update my installation. Are the svn-trunk updates pretty solid? I have not > attempted an update yet, as it is a production system. > > > Trunk has been very solid with a few minor exceptions. Best bet is to back up everything and do the upgrade during down time. If you have a test system that you can use as a sandbox that would be even better... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/950fef76/attachment.html From msc at freeswitch.org Mon Nov 23 17:07:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 17:07:45 -0800 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> Message-ID: <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> On Mon, Nov 23, 2009 at 4:58 PM, Seven Du wrote: > And because it's static string for on-hook members, it's hard to set > dynamically. For now, I'm using a callback way - whenever the sip client > answered the call, it fetch the real connected number from a http server. > That's not ideal because not only it add the complexity but also the callee > have no idea what the number is before answer. > > The problem for on-hook agent is that it call the agent first, and then get > one customer from the fifo queue, so it is not possible to let the agent > know the real caller-id before answer. Ideas? > > Tony and Brian were discussing this today. They bring up a really good point: do you want to risk having calls remain on hold as they bounce around looking for an agent? This can happen if you pre-determine which caller goes to which agent and the agent doesn't answer. I do understand why this feature matters to many people - it's how old school ACD systems work. However, mod_fifo is more efficient. It's hard to justify decreasing call routing efficiency in order to display the caller's info to the on-hook agent prior to answering. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/0ca1c6bf/attachment.html From brian at freeswitch.org Mon Nov 23 17:12:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 19:12:20 -0600 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> Message-ID: You do realize that the whole concept is OLD skewl. You should be popping this info via external resources when the agent is bridged to the caller and the info is there before they are done saying "thanks for calling spacely sprockets, this is George how may I help you .... " /b On Nov 23, 2009, at 7:07 PM, Michael Collins wrote: > Tony and Brian were discussing this today. They bring up a really > good point: do you want to risk having calls remain on hold as they > bounce around looking for an agent? This can happen if you pre- > determine which caller goes to which agent and the agent doesn't > answer. I do understand why this feature matters to many people - > it's how old school ACD systems work. However, mod_fifo is more > efficient. It's hard to justify decreasing call routing efficiency > in order to display the caller's info to the on-hook agent prior to > answering. > > -MC From msc at freeswitch.org Mon Nov 23 17:18:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 17:18:08 -0800 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> Message-ID: <87f2f3b90911231718q46c51982j36067f1627dd759c@mail.gmail.com> On Mon, Nov 23, 2009 at 5:12 PM, Brian West wrote: > You do realize that the whole concept is OLD skewl. You should be > popping this info via external resources when the agent is bridged to > the caller and the info is there before they are done saying "thanks > for calling spacely sprockets, this is George how may I help you .... " > > /b > > Agreed! Screen pop should be easy in the 21st Century. If it's not then you've got MUCH bigger problems than caller ID being delivered to your FIFO agents... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/d04f1ea1/attachment.html From dujinfang at gmail.com Mon Nov 23 17:24:05 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Nov 2009 09:24:05 +0800 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> Message-ID: <23f91030911231724nc7dd28cs6a8c1ae738b3aae8@mail.gmail.com> Yes, that's what we are doing. 2009/11/24 Brian West > You do realize that the whole concept is OLD skewl. You should be > popping this info via external resources when the agent is bridged to > the caller and the info is there before they are done saying "thanks > for calling spacely sprockets, this is George how may I help you .... " > > /b > > > On Nov 23, 2009, at 7:07 PM, Michael Collins wrote: > > > Tony and Brian were discussing this today. They bring up a really > > good point: do you want to risk having calls remain on hold as they > > bounce around looking for an agent? This can happen if you pre- > > determine which caller goes to which agent and the agent doesn't > > answer. I do understand why this feature matters to many people - > > it's how old school ACD systems work. However, mod_fifo is more > > efficient. It's hard to justify decreasing call routing efficiency > > in order to display the caller's info to the on-hook agent prior to > > answering. > > > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b7cac2a2/attachment-0001.html From lists at redbonez.net Mon Nov 23 17:26:47 2009 From: lists at redbonez.net (Adam Ford) Date: Mon, 23 Nov 2009 18:26:47 -0700 Subject: [Freeswitch-users] Business/holiday hours routing Message-ID: <00be01ca6ca5$31f64ff0$95e2efd0$@net> Is there a standard module for FreeSWITCH out there that people use for routing calls based on business hours and a holiday schedule? Or is everyone just creating their own in the XML dialplan?(which seems pretty simple) I can't seem to find anything on the wiki, but might just be searching for the wrong thing. I am relatively new at FreeSWITCH and would rather follow what the community has decided is the best practice, instead of trying to reinvent the wheel myself. Thanks for any input, -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/0e7b0f5c/attachment.html From rob4manhere at gmail.com Mon Nov 23 17:30:03 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Mon, 23 Nov 2009 19:30:03 -0600 Subject: [Freeswitch-users] Memory leak with mod_local_stream In-Reply-To: <3BF17413-EC48-4691-8C99-61CC4661E2CA@jerris.com> References: <3BF17413-EC48-4691-8C99-61CC4661E2CA@jerris.com> Message-ID: <009439B2-B025-4C26-8407-A212A762A7F9@gmail.com> Ignore this. I'm an idiot. Rob :) On Nov 23, 2009, at 4:15 PM, Michael Jerris wrote: > That rev should have fixed that memory leak, could you test > mod_local_stream.c from rev 15430 (http://fisheye.freeswitch.org/browse/ > ~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/ > mod_local_stream.c) with your current fs version to confirm this is > the cause please? > > Mike > > > On Nov 23, 2009, at 4:53 PM, Rob Forman wrote: > >> Hey guys, >> >> Having a problem with mod_local_stream. >> >> I recently did a "make current" from 15334 to the latest trunk >> (15630). After restarting, there now appears to be a memory leak. >> On >> a test system (CentOS 5.4, 64-bit) with no calls or registrations, >> Freeswitch gradually consumes all of the host memory (rate of about >> 200K/second), then swaps out, eventually rendering the system >> useless. >> >> I isolated it to mod_local_stream. If I unload mod_local_stream, the >> memory use stops climbing. If I re-load mod_local_stream, it starts >> again. >> >> >> I would submit the logs except they aren't any besides it starting. >> The system is just sitting there idle. Even valgrind didn't show >> much >> (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I >> ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- >> check=full --leak-resolution=high --show-reachable=yes .libs/ >> freeswitch -vg >> >> Questions: >> * has anyone else seen this? >> * what is the best way I can assist troubleshooting this? >> >> I saw a patch to mod_local_stream (rev 15431) a few weeks back. >> Could >> that have anything to do with it? >> >> Rob >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Mon Nov 23 17:43:13 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Nov 2009 09:43:13 +0800 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <00be01ca6ca5$31f64ff0$95e2efd0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> Message-ID: <23f91030911231743m35928483vfcac709fce1cae4e@mail.gmail.com> XML has basic conditioning, but lua rocks. -- Time condition for sales 1 --session:setAutoHangup(false) function do_transfer(extn) --print(extn) session:transfer(extn, "XML", "sales") end now = os.date("%H:%M") w = tonumber(os.date("%w")) if w >= 1 and w <=5 then if ( now >= "09:00" and now < "20:30" ) then do_transfer("sales_fifo_1") elseif ( now >= "20:30" and now < "22:30" ) then do_transfer("sales_fifo_2") else do_transfer("sales_fifo_cellphone") end else if ( now >= "10:00" and now < "19:00" ) then do_transfer("sales_fifo_1") elseif (now >= "20:00" and now < "22:30" ) then do_transfer("sales_fifo_2") else do_transfer("sales_fifo_cellphone") end end 2009/11/24 Adam Ford > Is there a standard module for FreeSWITCH out there that people use for > routing calls based on business hours and a holiday schedule? Or is everyone > just creating their own in the XML dialplan?(which seems pretty simple) > > > > I can?t seem to find anything on the wiki, but might just be searching for > the wrong thing. I am relatively new at FreeSWITCH and would rather follow > what the community has decided is the best practice, instead of trying to > reinvent the wheel myself. > > > > Thanks for any input, > > -Adam > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/98e25335/attachment.html From brian at freeswitch.org Mon Nov 23 17:44:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 19:44:40 -0600 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <00be01ca6ca5$31f64ff0$95e2efd0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> Message-ID: <825B66B6-EE5B-4CA0-8CEF-3CF6A0E7789C@freeswitch.org> Please see default.xml dialplan at the top in SVN. /b On Nov 23, 2009, at 7:26 PM, Adam Ford wrote: > Is there a standard module for FreeSWITCH out there that people use > for routing calls based on business hours and a holiday schedule? Or > is everyone just creating their own in the XML dialplan?(which seems > pretty simple) > > I can?t seem to find anything on the wiki, but might just be > searching for the wrong thing. I am relatively new at FreeSWITCH > and would rather follow what the community has decided is the best > practice, instead of trying to reinvent the wheel myself. > > Thanks for any input, > -Adam > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/a73e6e4b/attachment.html From brian at freeswitch.org Mon Nov 23 17:45:46 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 19:45:46 -0600 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <23f91030911231724nc7dd28cs6a8c1ae738b3aae8@mail.gmail.com> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> <23f91030911231724nc7dd28cs6a8c1ae738b3aae8@mail.gmail.com> Message-ID: Its the proper way to do it. And do you know the second you're bridged... if you have a phone that isn't daft... snom or polycom... the display will update to the person your bridged to. /b On Nov 23, 2009, at 7:24 PM, Seven Du wrote: > Yes, that's what we are doing. > > 2009/11/24 Brian West > You do realize that the whole concept is OLD skewl. You should be > popping this info via external resources when the agent is bridged to > the caller and the info is there before they are done saying "thanks > for calling spacely sprockets, this is George how may I help > you .... " > > /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/1106afa6/attachment.html From andrew at hijacked.us Mon Nov 23 17:48:08 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 23 Nov 2009 20:48:08 -0500 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <00be01ca6ca5$31f64ff0$95e2efd0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> Message-ID: <20091124014808.GB3298@hijacked.us> On Mon, Nov 23, 2009 at 06:26:47PM -0700, Adam Ford wrote: > Is there a standard module for FreeSWITCH out there that people use for > routing calls based on business hours and a holiday schedule? Or is everyone > just creating their own in the XML dialplan?(which seems pretty simple) > > > > I can't seem to find anything on the wiki, but might just be searching for > the wrong thing. I am relatively new at FreeSWITCH and would rather follow > what the community has decided is the best practice, instead of trying to > reinvent the wheel myself. > I assume you've seen this: http://wiki.freeswitch.org/wiki/Time_of_Day_Routing I have a patch that'll let you specify the nth day of the nth week via wday="3,4" for the 4th tuesday in the month. This willl let you do vacations like thanksgiving, MLK day, etc as well. Andrew From lists at redbonez.net Mon Nov 23 18:13:48 2009 From: lists at redbonez.net (Adam Ford) Date: Mon, 23 Nov 2009 19:13:48 -0700 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <20091124014808.GB3298@hijacked.us> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> Message-ID: <00e101ca6cab$c3525240$49f6f6c0$@net> Yes, that wiki page is what I was referring to when I said it seems simple enough to do it in the XML dialplan. So other than the one lua suggestion, it seems the majority say XML is the way to go eh? -Adam Andrew - that does sound like a useful patch, is it in svn or unpublished? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Thompson Sent: Monday, November 23, 2009 6:48 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Business/holiday hours routing On Mon, Nov 23, 2009 at 06:26:47PM -0700, Adam Ford wrote: > Is there a standard module for FreeSWITCH out there that people use for > routing calls based on business hours and a holiday schedule? Or is everyone > just creating their own in the XML dialplan?(which seems pretty simple) > > > > I can't seem to find anything on the wiki, but might just be searching for > the wrong thing. I am relatively new at FreeSWITCH and would rather follow > what the community has decided is the best practice, instead of trying to > reinvent the wheel myself. > I assume you've seen this: http://wiki.freeswitch.org/wiki/Time_of_Day_Routing I have a patch that'll let you specify the nth day of the nth week via wday="3,4" for the 4th tuesday in the month. This willl let you do vacations like thanksgiving, MLK day, etc as well. Andrew _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From james at talent.com.au Mon Nov 23 18:16:34 2009 From: james at talent.com.au (James Budge) Date: Tue, 24 Nov 2009 12:16:34 +1000 Subject: [Freeswitch-users] os x compile failure Message-ID: The compile fails after this. i686-apple-darwin10-gcc-4.2.1: -bundle not allowed with -dynamiclib From brian at freeswitch.org Mon Nov 23 18:17:46 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 20:17:46 -0600 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <00e101ca6cab$c3525240$49f6f6c0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> <00e101ca6cab$c3525240$49f6f6c0$@net> Message-ID: <21CB5F92-98DE-4622-ADC5-013462A93BD2@freeswitch.org> He's working on it for SVN... I recommended the format and to add the phases of the moon and zodiac signs just for giggles. /b On Nov 23, 2009, at 8:13 PM, Adam Ford wrote: > > Andrew - that does sound like a useful patch, is it in svn or > unpublished? From brian at freeswitch.org Mon Nov 23 18:20:30 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 20:20:30 -0600 Subject: [Freeswitch-users] os x compile failure In-Reply-To: References: Message-ID: While I love that people report issues... can you elaborate on things a bit? OS X version? CPU? SVN Rev? /b On Nov 23, 2009, at 8:16 PM, James Budge wrote: > The compile fails after this. > > i686-apple-darwin10-gcc-4.2.1: -bundle not allowed with -dynamiclib From Prometheus001 at gmx.net Mon Nov 23 18:37:11 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 24 Nov 2009 03:37:11 +0100 Subject: [Freeswitch-users] Problems with Voicemail In-Reply-To: <4B099098.2040408@gmx.net> References: <4B073ACD.1090708@gmx.net> <976A0342-4F4B-4035-9201-D56F8625AE12@freeswitch.org> <4B07D999.4040004@gmx.net> <4B099098.2040408@gmx.net> Message-ID: <4B0B46D7.1050609@gmx.net> I sorted it out. Something went wrong with the odbc database. I deleted the voicemail database tables, restarted FS and let FS create the tables again. Now it works. I can even share the voicemails across 2 Freeswitch boxes. Best regards Peter Peter P GMX schrieb: > I now created a file inbox.PCMA and get the following: > > * inbox.PCMA is played > * the recorded voive mail file is not played (FS does not even try > to do that) > * then I hear > o "to listen to the recording press 1" > o "to save the recording press 2" > o ... > > Here's the debug output > 2009-11-22 20:17:43.701098 [DEBUG] switch_core_io.c:660 > sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] > 2009-11-22 20:17:44.278600 [DEBUG] switch_ivr_play_say.c:1428 done > playing file > 2009-11-22 20:17:44.386776 [INFO] mod_native_file.c:82 Opening File > [/usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA] 8000hz > 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1428 done > playing file > 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:118 No language > specified - Using [en] > 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-listen_to_recording.wav] (en:en) > 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1136 Codec > Activated L16 at 8000hz 1 channels 20ms > 2009-11-22 20:17:45.201099 [DEBUG] switch_core_io.c:660 > sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] > 2009-11-22 20:17:46.419933 [DEBUG] switch_ivr_play_say.c:1428 done > playing file > > nGrepping port 3306 I can see that the correct filenames are retrieved > from the mysql/odbc database: > 1258894746.0.200.sip1.mydomain.com$d11c2a74-d766-11de-997b-bd7aecdc2a16.Gor > Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_c57a5e84-d766-11de-997b-bd7aecdc2a16.wav.4..B_NORMAL.....47 > 1258897120.0.200.sip1.mydomain.com$580dafee-d76c-11de-84d4-a1cd7fa320b3.Gor > Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_4d484a7e-d76c-11de-84d4-a1cd7fa320b3.wav.5..B_NORMAL......... > Both filenames can be read. > > Best regards > Peter > > Peter P GMX schrieb: > >> I installed all sounds from SVN, but >> >> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA >> >> isn't there. I checked another, older installation and couldn't this >> file either. >> >> I think that freeswitch tries to build a sound path for the file to be >> played, and some parts of the path are missing. >> I expect it would play a recorded message at that time in >> /usr/local/freeswitch/storage/voicemail/default/${domain} and the >> defined format is "wav" not pcma. >> >> I also set "storage_dir" explicitely in the voicemail configs,but this >> also didn't help. >> >> Best regards >> Peter >> >> >> Brian West schrieb: >> >> >>> I'm going to venture to guess maybe the file was recorded in a >>> different codec and NOT pcma? >>> >>> /b >>> >>> On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: >>> >>> >>> >>> >>>> 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / >>>> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA >>>> >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From james at talent.com.au Mon Nov 23 18:44:30 2009 From: james at talent.com.au (James Budge) Date: Tue, 24 Nov 2009 12:44:30 +1000 Subject: [Freeswitch-users] os x compile failure In-Reply-To: References: Message-ID: <40B8DB3D-2F66-45BC-BB9E-9B773B707FA3@talent.com.au> 2GHz Intel Core Duo OS 10.6.2 Xcode 3.2.1 Updated to revision 15648. On 24/11/2009, at 12:20 PM, Brian West wrote: > While I love that people report issues... can you elaborate on things > a bit? OS X version? CPU? SVN Rev? > > /b > > On Nov 23, 2009, at 8:16 PM, James Budge wrote: > >> The compile fails after this. >> >> i686-apple-darwin10-gcc-4.2.1: -bundle not allowed with -dynamiclib > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Mon Nov 23 18:45:45 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Nov 2009 10:45:45 +0800 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <00e101ca6cab$c3525240$49f6f6c0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> <00e101ca6cab$c3525240$49f6f6c0$@net> Message-ID: <23f91030911231845w4247b7fcmc05f920a81dd288d@mail.gmail.com> 2009/11/24 Adam Ford > Yes, that wiki page is what I was referring to when I said it seems simple > enough to do it in the XML dialplan. So other than the one lua suggestion, > it seems the majority say XML is the way to go eh? > Yeah, I made that lua script when the latest XML time routing feature was unavailable. > > -Adam > > Andrew - that does sound like a useful patch, is it in svn or unpublished? > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew > Thompson > Sent: Monday, November 23, 2009 6:48 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Business/holiday hours routing > > On Mon, Nov 23, 2009 at 06:26:47PM -0700, Adam Ford wrote: > > Is there a standard module for FreeSWITCH out there that people use for > > routing calls based on business hours and a holiday schedule? Or is > everyone > > just creating their own in the XML dialplan?(which seems pretty simple) > > > > > > > > I can't seem to find anything on the wiki, but might just be searching > for > > the wrong thing. I am relatively new at FreeSWITCH and would rather > follow > > what the community has decided is the best practice, instead of trying to > > reinvent the wheel myself. > > > I assume you've seen this: > > http://wiki.freeswitch.org/wiki/Time_of_Day_Routing > > I have a patch that'll let you specify the nth day of the nth week via > wday="3,4" for the 4th tuesday in the month. This willl let you do > vacations like thanksgiving, MLK day, etc as well. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/791e0443/attachment.html From brian at freeswitch.org Mon Nov 23 18:47:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 20:47:27 -0600 Subject: [Freeswitch-users] os x compile failure In-Reply-To: <40B8DB3D-2F66-45BC-BB9E-9B773B707FA3@talent.com.au> References: <40B8DB3D-2F66-45BC-BB9E-9B773B707FA3@talent.com.au> Message-ID: <8CD390C6-D194-483C-8A0A-732B5BFFCE09@freeswitch.org> Ok 32bit... we are currently working on that as I type. /b On Nov 23, 2009, at 8:44 PM, James Budge wrote: > 2GHz Intel Core Duo > > OS 10.6.2 > > Xcode 3.2.1 > > Updated to revision 15648. From john_platts at hotmail.com Mon Nov 23 20:33:03 2009 From: john_platts at hotmail.com (John Platts) Date: Mon, 23 Nov 2009 22:33:03 -0600 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH In-Reply-To: <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org> References: , <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org> Message-ID: I was using revision 15586. ---------------------------------------- > From: brian at freeswitch.org > Date: Mon, 23 Nov 2009 18:25:44 -0600 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH > > What rev exactly? > > /b > > On Nov 23, 2009, at 6:19 PM, John Platts wrote: > >> >> I actually checked out the latest version of FreeSWITCH in the SVN >> repository. >> >> I have the following configured in /usr/local/freeswitch/conf/ >> dialplan/default.xml: >> >> >> >> >> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/ From talk2ram at gmail.com Mon Nov 23 21:54:19 2009 From: talk2ram at gmail.com (ram) Date: Mon, 23 Nov 2009 21:54:19 -0800 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <4B0AD655.9070507@greatiam.com> References: <4B0AA8B6.2080305@greatiam.com> <4B0AD655.9070507@greatiam.com> Message-ID: On Mon, Nov 23, 2009 at 10:37 AM, Otis wrote: > Thanks. > > I have to get a centos box I guess. > > Much appreciated > > Samuel 'Otis' > > > how about trying Fusionpbx.com ( GUI) Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/b7943aca/attachment-0001.html From thangappan143 at gmail.com Mon Nov 23 21:56:44 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Tue, 24 Nov 2009 11:26:44 +0530 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> Message-ID: <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> The reason for waiting only for DTMF event is to handle the time outs in the IVR concept like response and inter digit time out. Using our own logic we 10 voice files in each play back if the voice files are more than 10. Now it works fine. Now the new problem has been raised. The problem is we are filtering only for DTMF events but we are getting COMMAND event . Because of this the DTMF digits are missing at the time . I am not able to proceed further. We are in the critical situation. Why this command event is occurring? How can I restrict this? What are the information it has? How can I get all the information in it ? ( If command event has info) Help me............ On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: > I am waiting only for DTMF events. That's why I am setting freeswitch > variable for knowing whether the playback has done. > > My question is "why this freeswitch variable is not setting properly when I > play back more than 10 files using playback_delimiter option?". > > When I play back lesser than ten voice files the variable has been set > properly. What could be the reason? > > > > ---------- Forwarded message ---------- > From: Thangappan.M > Date: Sat, Nov 21, 2009 at 2:52 PM > Subject: Problem while playing more than 10 voice files using playback > To: freeswitch-users > > > Dear all, > > I am in the process of implementing IVR using event outbound > socket (async mode). > I have implemented using Perl language. > > I did the following steps: > => Set the playback_delimiter variable > => Set the playback_sleep_val variable > => Set the event lock as true > => Set the freeswitch ( my own) variable as zero > => Wait in the loop until the variable is been set as > zero > => Playback the voice files ( Here I combined the voice > files with the delimiter value if more than one voice files are there) > => Set the freeswitch(my own) variable as true ( This is > used to identify whether the voice files are played > successfully). > => Wait in the loop until the variable is been set as > one. > => Set the Event lock as false > > => Trying to get the DTMF digits ( Have a assurance > that all the voice files are played). > > The problem is, > > The above steps are working fine when the voice file count is > lesser than or equal to 10. After the voice files are played only the > variable(my own freeswitch) is set. Based on the variable I am doing further > things. > > But when I tried to give the voice files count of more than 10 > the variable has been set while starting to play back the first voice file > itself . Because of this I am not able to proceed further. > > *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* > > *NOTE*: I also referred mod_file_string documentation. In that they > specified 128 files can be used to play back the voice files using > playback_delimiter option. > > Please help me................? > Thanks in advance. > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b475fc39/attachment.html From yudha2008 at gmail.com Mon Nov 23 22:03:58 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 24 Nov 2009 11:33:58 +0530 Subject: [Freeswitch-users] DTMF javasript Message-ID: * Hi,* * * *I want to check value given to the javascript with conditions whether it is voicefile, extension or mobile Number when i press the dtmf value.* * * *Steps i need to check in javascript:* * * *When i Press the DTMF value 1 it should check the 3 condition* * * If the Value for argv[2]=vfsurya means it is a voice file so it should play the Voice file *If the Value for argv[2]=1001 means it is a extension. The call should Bridge the extension* *If the Value for argv[2]=9841799874 means it is a Mobile number. The call should Bridge the Mobile number* * * *var exit = false;* *var dtmf_digits = "";* *var repeat = 0;* *var argv[2]=vfsurya; // or var argv[2]=1001 or var argv[2]=Mobile Number* * * * * *function onInput( session, type, data, arg ) * *{* * if ( type == "dtmf" ) * * {* * console_log( "info", "Got digit " + data.digit + "\n" );* * if ( data.digit == "1" ) * * **{* * if(argv[2].startswith("vf"))* * **{* * **var voice2=voice.substring(2)+"
"* * **session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/"+voice2+".wav", onInput );* * **}* * **else if(argv[2].length==4)* * **{* * **console_log( "info", "Got voicefile " + argv[2] + "\n" );* * **session.execute("bridge", "sofia/internal/"+argv[2]+"%192.168.1.2", onInput ); * * **}* * **else* * **{* * **session.execute("bridge", "sofia/default/sip:"+argv[2]+"@ 192.168.1.135:5066", onInput ); * * **}* * }* * }* *}* * * *But if 1 is pressed there is no event trigger but it get the dtmf value as 1 in freeswitch console. * * * *can any one specify what is the error or correct me where i am wrong.* * -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/2da09a74/attachment.html From mike at jerris.com Mon Nov 23 22:23:00 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 01:23:00 -0500 Subject: [Freeswitch-users] Requesting testing. Message-ID: I have done quite a few changes to the build system and correcting build problems and other platform specific problems the last few days. Could everyone on the list please take a little time out of their day and do a clean fresh svn trunk checkout of FreeSWITCH and do a full build and report any errors you encounter (if not already reported) to http://jira.freeswitch.org. We have fixed things for many platforms including bsd, solaris, linux, and especially issues on OS X. Please try these out to make sure all works. Thanks Mike From andrew at hijacked.us Mon Nov 23 22:45:09 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 24 Nov 2009 01:45:09 -0500 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <21CB5F92-98DE-4622-ADC5-013462A93BD2@freeswitch.org> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> <00e101ca6cab$c3525240$49f6f6c0$@net> <21CB5F92-98DE-4622-ADC5-013462A93BD2@freeswitch.org> Message-ID: <20091124064509.GA6360@hijacked.us> On Mon, Nov 23, 2009 at 08:17:46PM -0600, Brian West wrote: > He's working on it for SVN... I recommended the format and to add the > phases of the moon and zodiac signs just for giggles. > I'll probably get a patch in this week (or early next) I'm thinking of changing the format so that "week of month" becomes its own value so you could compare against mweek as well as wday so thanksgiving + extension becomes something like If I really get ambitious I'd also like to allow wday="mon-fri" so I don't always forget that days are 1-indexed from sunday :) Andrew From velu.technical at gmail.com Mon Nov 23 23:22:30 2009 From: velu.technical at gmail.com (velusamy velu) Date: Tue, 24 Nov 2009 12:52:30 +0530 Subject: [Freeswitch-users] DTMF Event is not coming while using playback terminators. Message-ID: <1452e2980911232322j106fe5eoe3efad59199f36e4@mail.gmail.com> Dear All, I am using Perl ESL::IVR module to develop a simple IVR. I have filtered DTMF events. I have also set playback_terminators to cut the playback when giving the digits. I have faced problem that DTMF event has not come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. Why the 'COMMAND' event came event filter is on? How to avoid this event in ESL? Thanks, Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b777c286/attachment.html From siniypin at gmail.com Mon Nov 23 23:49:48 2009 From: siniypin at gmail.com (RobertT) Date: Tue, 24 Nov 2009 10:49:48 +0300 Subject: [Freeswitch-users] Requesting testing. In-Reply-To: References: Message-ID: <2160023e0911232349h6ef3a1f5m13c2cb21e12a70d2@mail.gmail.com> I've a problem building FS rev 15630 on Windows. One of mod_pocketsphinx related projects lack a code file. Regards, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/9ea7a70d/attachment-0001.html From mike at jerris.com Tue Nov 24 00:29:37 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 03:29:37 -0500 Subject: [Freeswitch-users] Requesting testing. In-Reply-To: <2160023e0911232349h6ef3a1f5m13c2cb21e12a70d2@mail.gmail.com> References: <2160023e0911232349h6ef3a1f5m13c2cb21e12a70d2@mail.gmail.com> Message-ID: <6C8B0117-27F5-4FEF-926A-6E6A97AA0309@jerris.com> please follow the procedures http://wiki.freeswitch.org/wiki/Reporting_Bugs to report bugs at http://jira.freeswitch.org. Also, you will need to provide far more detail than in this email for anyone to be able to have a possibility of fixing it. Please include details such as, what file is missing, what errors and warnings you get. How to reproduce it and preferably a patch to correct the problem if you can create one. Mike On Nov 24, 2009, at 2:49 AM, RobertT wrote: > I've a problem building FS rev 15630 on Windows. One of mod_pocketsphinx related projects lack a code file. > From mike at jerris.com Tue Nov 24 00:33:34 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 03:33:34 -0500 Subject: [Freeswitch-users] os x compile failure In-Reply-To: <8CD390C6-D194-483C-8A0A-732B5BFFCE09@freeswitch.org> References: <40B8DB3D-2F66-45BC-BB9E-9B773B707FA3@talent.com.au> <8CD390C6-D194-483C-8A0A-732B5BFFCE09@freeswitch.org> Message-ID: <363278A1-586F-4493-8A7E-BEEDAB036000@jerris.com> Please retest this with current svn trunk fresh checkout. Thanks Mike On Nov 23, 2009, at 9:47 PM, Brian West wrote: > Ok 32bit... we are currently working on that as I type. > > /b > > On Nov 23, 2009, at 8:44 PM, James Budge wrote: > >> 2GHz Intel Core Duo >> >> OS 10.6.2 >> >> Xcode 3.2.1 >> >> Updated to revision 15648. > From mike at jerris.com Tue Nov 24 00:39:16 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 03:39:16 -0500 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH In-Reply-To: References: , <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org> Message-ID: <6F2A2A62-CE26-477F-B402-358F313A3EC3@jerris.com> This was fixed in trunk yesterday about 8 hrs before you sent this message. (15619). Please update and try again. Mike On Nov 23, 2009, at 11:33 PM, John Platts wrote: > > I was using revision 15586. > > ---------------------------------------- >> From: brian at freeswitch.org >> Date: Mon, 23 Nov 2009 18:25:44 -0600 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH >> >> What rev exactly? >> >> /b >> >> On Nov 23, 2009, at 6:19 PM, John Platts wrote: >> >>> >>> I actually checked out the latest version of FreeSWITCH in the SVN >>> repository. >>> >>> I have the following configured in /usr/local/freeswitch/conf/ >>> dialplan/default.xml: >>> >>> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/3ece4be6/attachment.html From mike at jerris.com Tue Nov 24 00:41:25 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 03:41:25 -0500 Subject: [Freeswitch-users] DTMF javasript In-Reply-To: References: Message-ID: Your not telling anything to call your callback. On Nov 24, 2009, at 1:03 AM, Baskar wrote: > Hi, > > I want to check value given to the javascript with conditions whether it is voicefile, extension or mobile Number when i press the dtmf value. > > Steps i need to check in javascript: > > When i Press the DTMF value 1 it should check the 3 condition > > If the Value for argv[2]=vfsurya means it is a voice file so it should play the Voice file > If the Value for argv[2]=1001 means it is a extension. The call should Bridge the extension > If the Value for argv[2]=9841799874 means it is a Mobile number. The call should Bridge the Mobile number > > var exit = false; > var dtmf_digits = ""; > var repeat = 0; > var argv[2]=vfsurya; // or var argv[2]=1001 or var argv[2]=Mobile Number > > > function onInput( session, type, data, arg ) > { > if ( type == "dtmf" ) > { > console_log( "info", "Got digit " + data.digit + "\n" ); > if ( data.digit == "1" ) > { > if(argv[2].startswith("vf")) > { > var voice2=voice.substring(2)+"
" > session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/"+voice2+".wav", onInput ); > } > else if(argv[2].length==4) > { > console_log( "info", "Got voicefile " + argv[2] + "\n" ); > session.execute("bridge", "sofia/internal/"+argv[2]+"%192.168.1.2", onInput ); > } > else > { > session.execute("bridge", "sofia/default/sip:"+argv[2]+"@192.168.1.135:5066", onInput ); > } > } > } > } > > But if 1 is pressed there is no event trigger but it get the dtmf value as 1 in freeswitch console. > > can any one specify what is the error or correct me where i am wrong. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/70af6ed1/attachment.html From mike at jerris.com Tue Nov 24 00:42:07 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 03:42:07 -0500 Subject: [Freeswitch-users] DTMF Event is not coming while using playback terminators. In-Reply-To: <1452e2980911232322j106fe5eoe3efad59199f36e4@mail.gmail.com> References: <1452e2980911232322j106fe5eoe3efad59199f36e4@mail.gmail.com> Message-ID: <7EE00C1B-83CD-44C2-9CC0-F66120A9B534@jerris.com> async? On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: > Dear All, > I am using Perl ESL::IVR module to develop a simple IVR. I have filtered DTMF events. I have also set playback_terminators to cut the playback when giving the digits. I have faced problem that DTMF event has not come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. > > Why the 'COMMAND' event came event filter is on? > How to avoid this event in ESL? From velu.technical at gmail.com Tue Nov 24 00:59:29 2009 From: velu.technical at gmail.com (velusamy velu) Date: Tue, 24 Nov 2009 14:29:29 +0530 Subject: [Freeswitch-users] DTMF Event is not coming while using playback terminators. In-Reply-To: <7EE00C1B-83CD-44C2-9CC0-F66120A9B534@jerris.com> References: <1452e2980911232322j106fe5eoe3efad59199f36e4@mail.gmail.com> <7EE00C1B-83CD-44C2-9CC0-F66120A9B534@jerris.com> Message-ID: <1452e2980911240059n4223bd48u487a80b8306131da@mail.gmail.com> Yes, I am using async mode only.. On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris wrote: > async? > > On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: > > > Dear All, > > I am using Perl ESL::IVR module to develop a simple IVR. I have > filtered DTMF events. I have also set playback_terminators to cut the > playback when giving the digits. I have faced problem that DTMF event has > not come if DTMF given while playing voice files. I have received 'COMMAND' > event. I have the following questions. > > > > Why the 'COMMAND' event came event filter is on? > > How to avoid this event in ESL? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/2d65e5d2/attachment.html From info at daccii.it Tue Nov 24 01:15:45 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Tue, 24 Nov 2009 10:15:45 +0100 Subject: [Freeswitch-users] User who answer the bridge in a execute_answer In-Reply-To: <9133578A-F706-46C2-9653-6C22D6E056CB@jerris.com> References: <4B0A65C9.10509@daccii.it> <9133578A-F706-46C2-9653-6C22D6E056CB@jerris.com> Message-ID: <4B0BA441.9060905@daccii.it> Hi, thanks for the suggestion! In the end i updated freeswitch using lastest source in the trunk and callee_id_number worked! Best Regard, Daniele Michael Jerris ha scritto: > Try running the info app there and see if the info is anywhere in that output . > > Mike > > On Nov 23, 2009, at 5:36 AM, Albano Daniele Salvatore - Lavoro wrote: > >> Hi, >> >> i'm writing some dialplan parts that get executed on execute_on_answer. In this dialplan that get executed i need to make a directory to handle recordings for record_session and my folder structure is: >> USER/YEAR/MONTH/HOUR-MINUTE-SECOND-CALLER_NUMBER.wav >> >> ------ >> >> >> ------ >> >> The call flow is: >> Call from external -> IVR -> Transfer to Group -> Execute on Answer -> system/bind_meta_app >> >> >> Pratically, i need the number (or better the user) that answered the call: what variable should i check? >> >> I tried with sip_from_user, callee_id_number and some other. >> >> >> Thank for your help, >> >> Best Regards, >> Daniele> -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/8a3593f2/attachment-0001.vcf From stevendt at primrosebank.net Tue Nov 24 02:29:08 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 24 Nov 2009 10:29:08 -0000 Subject: [Freeswitch-users] Call Transfer Help Please Message-ID: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> Hi, I'm trying to setup call transfer for a phone without a transfer button. I was on IRC last night and got some pointers to how this is setup in dialplan.xml and features.xml and what "bind meta app" does. Once it became clear how the transfer is initiated and that the transfer, in the default config, can only be initiated by the "b" leg of the call, I was able to make this work as configured in the defaults, i.e, to initiate a transfer (for an internal call) from the dialled extension to a new extension. Now the problem . . . I have an incoming PSTN line that rings a group of extensions, what I want to be able to do is to give whoever answers the PSTN call ability to transfer the call on to another extension. There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch extension of 1000, it rings the extension phones in the group. I'd hoped that the default transfer setup would handle this without modification - the incoming call on extension 1000 would be the "a" leg, the answering extension would be the "b" leg and a transfer from "b" would work as per the default config. This does not work for me though. I'm struggling a bit with the "bind meta app" options and can't seem to make it do what I want. Could someone please confirm that what I'm trying to do is feasible and perhaps suggest the right parameters to use in dialplan.xml and features.xml please ? Relevant section in the "is_transfer" section in features.xml And in default.xml from to I've tried posting a call log to the Pastebin (11252/3) but there was an error - it looks like the dump was too big. Not sure what the maximum size on pastebin dumps is ? My understanding (or lack of) of "a" and "b" are in the scenario described is not helping ... Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? Is the answering extension the "b" leg ? What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ? What is the correct "transfer" data string in features.xml ? Or am I totally on the wrong track here ? If it is possible to do what I want, and changes are required to the dialplan.xml and/or features.xml files, is it possible to have different logic in there such that the actions are different whether it is the "a" leg or "b" leg that's requesting the transfer ? regards Dave FreeSwitch Version 1.0.4 (14460) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b78cef61/attachment.html From lakindia89 at gmail.com Tue Nov 24 02:57:05 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Tue, 24 Nov 2009 16:27:05 +0530 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> Message-ID: <7d79b3930911240257g4c22a09dhad954629ae49072d@mail.gmail.com> Yes Mr. Collins, I've tried with shed_api. But I was not able to control, if the user reject the call. I made a shed_api to originate a call to 1000 and If it is answered, I'll transfer the call to 9097 (So it comes to my program, refer the dialplan in my question). But what happens if the user 1000, reject the call. I can't control that. If the user 1000, reject the call, I need to call the user after some time. Any way to do this!! On Mon, Nov 23, 2009 at 11:21 PM, Michael Collins wrote: > > > On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> Hi, >> I'm using perl ESL to control the call in freeswitch. >> I'm having the following scenario, but not able to get it right. >> >> Dialplan: >> >> >> >> >> >> >> >> >> 1. User A calls to an extention (1000). >> 2. My ESL program will be running, and it answers the call. >> 3. Then the program will get a number from the user. >> 4. It will hangup the call. >> 5. The program has to call to the number that was given by the user. >> >> In the above scenario, I was able to do until the 4th step. After hangup >> the call, if I say originate it is not working. >> Any ideas on how to do this in ESL. >> >> > I want to make sure I understand what the script is supposed to be doing. > The caller will key in a phone number to your script and your script will > collect those digits. The script will then hangup on the caller and > originate a completely new call? Perhaps you could use sched_api to schedule > a new originate command for a few seconds into the future and then hangup? > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/d009686b/attachment.html From oscav at hotmail.fr Tue Nov 24 04:23:52 2009 From: oscav at hotmail.fr (Oscav) Date: Tue, 24 Nov 2009 04:23:52 -0800 (PST) Subject: [Freeswitch-users] Execute on Answer with JavaScript In-Reply-To: References: <26476532.post@talk.nabble.com> Message-ID: <26494996.post@talk.nabble.com> Hi Mike, I understand. I just need to not use the session.answer(). Many thanks. Michael Jerris wrote: > > This is done automatically when you bridge 2 sessions together. > > Mike > > On Nov 23, 2009, at 6:45 AM, Oscav wrote: >> How can we send the answer to the caller only when the callee answers, in >> JavaScript?? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Execute-on-Answer-with-JavaScript-tp26476532p26494996.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From lakindia89 at gmail.com Tue Nov 24 04:27:40 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Tue, 24 Nov 2009 17:57:40 +0530 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> Message-ID: <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> I've tried the following program as per the suggestion that you've told. But it seems, no success. Once the connection is closed, I created a new connection and I send originate to originate a new call. But it is not working. require ESL; use IO::Socket::INET; use Data::Dumper; my $ip = "192.168.1.222"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; my $make_call; my $con; my $type = "user/"; for(;;) { my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { close($new_sock); next; } my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s to %s\n", $uuid, $info->getHeader("caller-caller-id-number"), $info->getHeader("caller-destination-number"); $con->filter("Unique-Id", $uuid); $con->events("plain", "all"); $con->execute("answer"); $con->setEventLock("true"); my $number=$con->execute("read","2 4 /usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav accnt_number 5000 #"); while($con->connected()) { my $e=$con->recvEvent(); my $ename=$e->getHeader("Event-Name"); my $app=$e->getHeader("Application"); if($ename eq "CHANNEL_EXECUTE_COMPLETE" and $app eq "read") { my $num=$e->getHeader("variable_accnt_number"); print "$num\n"; $con->execute("hangup"); } } if(!$con->connected()) { print "Connection not exists\n"; $con = new ESL::ESLconnection($fd); $con->api("originate","user/1000 &park()"); print "Hai\n"; } print "Bye\n------------------------------------------------------------------\n"; close($new_sock); } Output: Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097 1000 Connection not exists Hai Bye ------------------------------------------------------------------ The freeswitch log is in http://pastebin.freeswitch.org/11258 I also noted that, if I don't receive any events, especially "SERVER_DISCONNECTED", then the connection is in established state, but once I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it correct?? On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > or open a new outbound connection at the end of your script so you can send > your originate command. > Since the channel hanging up will close your existing connection since it's > only an outbound single session socket. > > > On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins wrote: > >> >> >> On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < >> lakindia89 at gmail.com> wrote: >> >>> Hi, >>> I'm using perl ESL to control the call in freeswitch. >>> I'm having the following scenario, but not able to get it right. >>> >>> Dialplan: >>> >>> >>> >>> >>> >>> >>> >>> >>> 1. User A calls to an extention (1000). >>> 2. My ESL program will be running, and it answers the call. >>> 3. Then the program will get a number from the user. >>> 4. It will hangup the call. >>> 5. The program has to call to the number that was given by the user. >>> >>> In the above scenario, I was able to do until the 4th step. After hangup >>> the call, if I say originate it is not working. >>> Any ideas on how to do this in ESL. >>> >>> >> I want to make sure I understand what the script is supposed to be doing. >> The caller will key in a phone number to your script and your script will >> collect those digits. The script will then hangup on the caller and >> originate a completely new call? Perhaps you could use sched_api to schedule >> a new originate command for a few seconds into the future and then hangup? >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b7ad5841/attachment-0001.html From oscav at hotmail.fr Tue Nov 24 04:30:05 2009 From: oscav at hotmail.fr (Oscav) Date: Tue, 24 Nov 2009 04:30:05 -0800 (PST) Subject: [Freeswitch-users] sched_broadcast doesn't execute In-Reply-To: <191c3a030911181128g35ba0652w9fc575d5586367dc@mail.gmail.com> References: <26408422.post@talk.nabble.com> <191c3a030911181128g35ba0652w9fc575d5586367dc@mail.gmail.com> Message-ID: <26495078.post@talk.nabble.com> Hi Anthony, Now it works very well. Thank you so much for your help. I'm having a lot of fun with this platform. Regards. Anthony Minessale-2 wrote: > > is that your exact code? > > ${uuid} will not be expanded by javascript > > var uuid = session.getVariable(uuid); > > new_session.execute("sched_broadcast", "+20 alloted_timeout " + uuid + " > playback:ivr-welcome_to_freeswitch.wav"); > > On Wed, Nov 18, 2009 at 10:07 AM, Oscav wrote: > >> >> Hi, >> >> I'm writing a script in Javascript that plays a message during a bridge. >> I'm >> trying to use a sched_broadcast to do it. The job is scheduled and then >> deleted but I never hear the wav file and I don't get the "OK Message >> Scheduled" in the log. It even doesn't display any error message if I >> specify a wrong file name. Someone could help me on this issue ?? >> >> new_session.execute("sched_broadcast", "+20 alloted_timeout ${uuid} >> playback:ivr-welcome_to_freeswitch.wav"); >> >> I already did some posts but I got no answer. This is very difficult to >> progress without help. >> >> Many thanks >> -- >> View this message in context: >> http://old.nabble.com/sched_broadcast-doesn%27t-execute-tp26408422p26408422.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/sched_broadcast-doesn%27t-execute-tp26408422p26495078.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yehavi.bourvine at gmail.com Tue Nov 24 05:05:34 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 24 Nov 2009 15:05:34 +0200 Subject: [Freeswitch-users] How to find whether the destination extension supports encryption Message-ID: Hello, We have a mix of phones that support RTP encryption and those that do not. I have to support both types in the meanwhile, and would like to have encryption enabled on the relevant leg, even if the other leg does not support it (why? one of our ATAs either must have it unencrypted or have it encrypted, but cannot have both). How do I find whether the *destination* supports encryption? I do not want to manage an additional table in the database... Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/7e41e5a6/attachment.html From yehavi.bourvine at gmail.com Tue Nov 24 05:02:15 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 24 Nov 2009 15:02:15 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: <191c3a030911231157p44612c5dm3f0ee1e7b37e9cd3@mail.gmail.com> References: <4B0266F4.8070602@savion.huji.ac.il> <191c3a030911231157p44612c5dm3f0ee1e7b37e9cd3@mail.gmail.com> Message-ID: Hello Anthony, Indeed I see the reference to this channel variable in the code, but when trying to access it from the dial plan it is empty... I try to get the value of ${sip_profile_name} and it is empty. Thanks! __Yehavi: 2009/11/23 Anthony Minessale > Let's just do this: > > r15629 or higher > > look for sip_profile_name > > > > > On Tue, Nov 17, 2009 at 3:03 AM, Eli Hayun wrote: > >> Hi >> We have more then one profile. To make a call I have to enter : bridge >> sofia/profile/number at ip >> The problem is when I use : "${use_profile}" I am getting the caller >> profile, and I need the destination profile. >> >> How do I get this information? >> >> Thanks >> >> Eli >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b43fc10c/attachment.html From achaloyan at yahoo.com Tue Nov 24 05:19:09 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Tue, 24 Nov 2009 05:19:09 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <1258995685201-4052409.post@n2.nabble.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> <1258949788572-4048969.post@n2.nabble.com> <858430.90192.qm@web111301.mail.gq1.yahoo.com> <1258995685201-4052409.post@n2.nabble.com> Message-ID: <900524.79591.qm@web111310.mail.gq1.yahoo.com> Hi Jeff, All is good, I have looked at the x64 related changes you made and will merge them back to UniMRCP tree most probably during the next week. Thanks, Arsen. ________________________________ From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Mon, November 23, 2009 9:01:25 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Hi Arsen, I have merged your changes in now - thank you. Would you perhaps be able to look at the x64 changes I made to the projects and merge them back into your code to ease the future updating. Thanks Jeff Arsen Chaloyan wrote: > > Hi Jeff, > > > Your input would be very helpful, I just wanted to understand where the > problem is and contribute the way I can. > I see you're the assignee, so please go ahead and let me know if there is > anything left I can help with. > > Arsen. > > > > ________________________________ > From: Jeff Lenk > To: freeswitch-users at lists.freeswitch.org > Sent: Mon, November 23, 2009 8:16:28 AM > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > Hi Arsen, > > I would be happy to help with the FS integration if you want - please do > put your patch in a Jira. > > Jeff > > ________________________________ > Date: Sun, 22 Nov 2009 10:09:41 -0800 > From: [hidden email] > To: [hidden email] > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > > We discussed build integration related issues a few months ago with Mike > and seemed to find a solution which would work for both UniMRCP and > FreeSWITCH source trees. > > Now I've just got a chance to look into this a bit closer trying to > further complete VS2008 build integration in FreeSWITCH. So I've got it > working, the module is not only being built, but also is getting loaded. > Current build integration is not as seamless as I want it to be, but > probably we can start with what we have now and then discuss and identify > what can be done in the future. This concerns not only build integration > but overall integrity. > > So would you be interested in the patch? Where should I upload it? > I thought I had a Jira account, but not sure it exists any more. > > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org > > > > > > ________________________________ > From: Jeff Lenk <[hidden email]> > To: [hidden email] > Sent: Fri, November 20, 2009 7:59:28 PM > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > > That module is not currently being built for Windows. Also the library > unimrcp needs build integration work with FS to make that happen under > windows. > > > ss1 wrote: > >> >> Hi Everyone, >> >> Please help freeswitch experts... !!! >> >> i have been working on freeswitch from last 2 days. i have downloaded >> freeswitch and unimrcp (server + client) for windows. >> I tested the unimrcp client and server, which is running fine with the >> command: run synth and run recog. I got both synth.pcm & recog.pcm files. >> >> But my objective is to call Freeswitch through x-lite, where freeswitch >> should call unimrcp client and return the PCM files. >> >> I tried it alot, but unable to do it. after lots of reading i found that >> i >> do not have mod_unimrcp. i do not know from where to download it and how >> to merge it into freeswitch. >> >> I would be very thankful if you may help. >> >> Thanks, >> ss >> >> -- > View this message in context: > http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > > View message @ > http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html > To unsubscribe from Re: need help !! Problem with freeswitch & uniMRCP, > click here. > > ________________________________ > Hotmail: Trusted email with powerful SPAM protection. Sign up now. > ________________________________ > View this message in context: RE: [Freeswitch-users] need help !! Problem > with freeswitch & uniMRCP > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4052409.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/2072e68b/attachment-0001.html From steve at justfone.com Tue Nov 24 05:25:06 2009 From: steve at justfone.com (Steven Brown) Date: Tue, 24 Nov 2009 13:25:06 +0000 Subject: [Freeswitch-users] Noise with openzap Message-ID: <3e6d7b0c0911240525o747f7b05y5c1f50ec6afe1179@mail.gmail.com> Hi, I have an Ubuntu box running FS1.0.4 which has been processing a good volume of calls between local users with soft phones (Xlite) and GSM handsets via a number or Portech gateways, this has worked very well for some time and audio quality is very good. I've now added a Sangoma A200 with 4 ports hooked up to 4 PSTN lines, configured openzap and I can originate and answer calls on the the openzap lines fine, however these calls via opezap all seem to suffer from significant noise, the audio path works fine in both directions but noise seems particularly bad at the local soft phone end. Quality of all other calls through the box is fine though, any ideas appreciated ?, NB A regular handset plugged directly into the PSTN lines has no problems though Thanks Steve From lei.tlfly at gmail.com Tue Nov 24 06:12:31 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 24 Nov 2009 22:12:31 +0800 Subject: [Freeswitch-users] FS cluster and how to get sofia gateway health status? Message-ID: <50c41b4e0911240612w7506a2f1qed0f25143c0b65d2@mail.gmail.com> Hi everyone, I'm setting up FS cluster In my application, I plan to use two FS server as front and four FS as backend, the incoming calls first send to the front FS, then the front FS forward the call to backend FS server by return 302 to invite message. The front FS need to known the backend FS's status, so it won't forward calls to a server if it's down. The question is, how to check the backend FS's status. As I known, fs can add gateways to sofia profile, the endpoint will check gateways's state by send ping message, I think it is the function what I need if I can get the gateways's status from fs, does someone known how to do it? or can someone give me some suggestion about how to setup FS cluster? -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/57da1325/attachment.html From imthiyazg at gmail.com Tue Nov 24 06:54:23 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Tue, 24 Nov 2009 20:24:23 +0530 Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <1258732768082-4038514.post@n2.nabble.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> Message-ID: <8595daf70911240654y72f0440cm2ab1a50babc96f0f@mail.gmail.com> Hi Can we enable passive recording in freeswitch ,wanpipe ,openzap , we are using a sangoma tapping system with freeswitch. Thanks Imthiyaz From ovvenkatesan at gmail.com Tue Nov 24 04:49:34 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Tue, 24 Nov 2009 18:19:34 +0530 Subject: [Freeswitch-users] How to run IVR application Message-ID: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> Hi to all, I am very new this platform . I just downloaded freeswitch to my windows xp machine , compiled successfully and run. After that I dont have any idea what to do :( . I am not finding simple kind of tutorial on the net. could you please suggest me, how I have to proceed. My requirement is; I need to run IVR application on machine using SIP phone. I am very sorry to my bad English. Thanks and Regards, Venkat. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/595e26d5/attachment.html From rajesh.npnr at yahoo.com Tue Nov 24 07:01:04 2009 From: rajesh.npnr at yahoo.com (rex.alex) Date: Tue, 24 Nov 2009 07:01:04 -0800 (PST) Subject: [Freeswitch-users] SoftPhone Message-ID: <1259074864897-4058292.post@n2.nabble.com> Hello, I have been going through FreeSWITCH for quite sometime now. I would like to develop my own SIP Client soft-phone in Java/etc., how do I start?. Will I get any SDK/APIs for this. Please assist. Thanks, Rex -- View this message in context: http://n2.nabble.com/SoftPhone-tp4058292p4058292.html Sent from the freeswitch-users mailing list archive at Nabble.com. From juanbackson at gmail.com Tue Nov 24 07:22:22 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 24 Nov 2009 23:22:22 +0800 Subject: [Freeswitch-users] remote_media_ip variable not set Message-ID: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> Hi, I tried to use the variable remote_media_ip from within dialplan, but it is not returning anything. Does anyone know when this variable gets set and how to have this variable to be set as soon as an INVITE hit freeswitch? Thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/66c5d26e/attachment.html From edpimentl at gmail.com Tue Nov 24 07:30:48 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 24 Nov 2009 10:30:48 -0500 Subject: [Freeswitch-users] SoftPhone In-Reply-To: <1259074864897-4058292.post@n2.nabble.com> References: <1259074864897-4058292.post@n2.nabble.com> Message-ID: <9dc4a1670911240730q58f52e65o3b7586b30fd684e9@mail.gmail.com> Suggestion: Be one the first to integrate QuteCOm -E Gpro.ws -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/baefd770/attachment.html From mike at jerris.com Tue Nov 24 07:43:44 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 10:43:44 -0500 Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <8595daf70911240654y72f0440cm2ab1a50babc96f0f@mail.gmail.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <8595daf70911240654y72f0440cm2ab1a50babc96f0f@mail.gmail.com> Message-ID: <0636459B-CEAF-4332-84BD-D32DA0322A29@jerris.com> What does this have to do with uniMRCP? Mike On Nov 24, 2009, at 9:54 AM, Imthiyaz Ahmed wrote: > Hi > > Can we enable passive recording in freeswitch ,wanpipe ,openzap , we > are using a sangoma tapping system with freeswitch. From mrene_lists at avgs.ca Tue Nov 24 07:46:43 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 24 Nov 2009 10:46:43 -0500 Subject: [Freeswitch-users] remote_media_ip variable not set In-Reply-To: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> References: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> Message-ID: <2F929FDB-0E1B-49E0-A1E7-F4F1E2D548AD@avgs.ca> It gets set whenever the codec is negotiated. So it'll be NULL until (pre_)answer if you have late-negotiation on. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 24-Nov-09, at 10:22 AM, Juan Backson wrote: > Hi, > > I tried to use the variable remote_media_ip from within dialplan, > but it is not returning anything. > > Does anyone know when this variable gets set and how to have this > variable to be set as soon as an INVITE hit freeswitch? > > Thanks, > jb > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From juanbackson at gmail.com Tue Nov 24 07:56:17 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 24 Nov 2009 23:56:17 +0800 Subject: [Freeswitch-users] remote_media_ip variable not set In-Reply-To: <2F929FDB-0E1B-49E0-A1E7-F4F1E2D548AD@avgs.ca> References: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> <2F929FDB-0E1B-49E0-A1E7-F4F1E2D548AD@avgs.ca> Message-ID: <27c25bc40911240756k7842c80kd75be2d3d93441b9@mail.gmail.com> Hi, In the case of proxy_media=true, does it gets set at all then? thanks, jb On Tue, Nov 24, 2009 at 11:46 PM, Mathieu Rene wrote: > It gets set whenever the codec is negotiated. So it'll be NULL until > (pre_)answer if you have late-negotiation on. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 24-Nov-09, at 10:22 AM, Juan Backson wrote: > > > Hi, > > > > I tried to use the variable remote_media_ip from within dialplan, > > but it is not returning anything. > > > > Does anyone know when this variable gets set and how to have this > > variable to be set as soon as an INVITE hit freeswitch? > > > > Thanks, > > jb > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/30a1aeb9/attachment.html From anthony.minessale at gmail.com Tue Nov 24 11:24:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Nov 2009 13:24:58 -0600 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> Message-ID: <191c3a030911241124r49679ad7je630a964dd60a5c3@mail.gmail.com> 1) Did you ever supply a log of your problem? 2) Are you using ESL lib or did you make your own event socket client, (if you did maybe you implemented the protocol client wrong) You are not supplying any specific information like traces of the connection or the version of the code you are using, weather you have tried the latest release or not etc. And lastly you are not using the events I told you about to tell exactly when the commands in question are being executed. getting a variable in a loop is a non-scalable memory consuming bad idea in how to program over a socket. On Mon, Nov 23, 2009 at 11:56 PM, Thangappan.M wrote: > The reason for waiting only for DTMF event is to handle the time outs in > the IVR concept like response and inter digit time out. Using our own logic > we 10 voice files in each play back if the voice files are more than 10. Now > it works fine. > > Now the new problem has been raised. The problem is we are filtering only > for DTMF events but we are getting COMMAND event . Because of this the DTMF > digits are missing at the time . I am not able to proceed further. We are > in the critical situation. > > Why this command event is occurring? > How can I restrict this? > What are the information it has? > How can I get all the information in it ? ( If command event has info) > > Help me............ > > > On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: > >> I am waiting only for DTMF events. That's why I am setting freeswitch >> variable for knowing whether the playback has done. >> >> My question is "why this freeswitch variable is not setting properly when >> I play back more than 10 files using playback_delimiter option?". >> >> When I play back lesser than ten voice files the variable has been set >> properly. What could be the reason? >> >> >> >> ---------- Forwarded message ---------- >> From: Thangappan.M >> Date: Sat, Nov 21, 2009 at 2:52 PM >> Subject: Problem while playing more than 10 voice files using playback >> To: freeswitch-users >> >> >> Dear all, >> >> I am in the process of implementing IVR using event outbound >> socket (async mode). >> I have implemented using Perl language. >> >> I did the following steps: >> => Set the playback_delimiter variable >> => Set the playback_sleep_val variable >> => Set the event lock as true >> => Set the freeswitch ( my own) variable as zero >> => Wait in the loop until the variable is been set as >> zero >> => Playback the voice files ( Here I combined the >> voice files with the delimiter value if more than one voice files are there) >> => Set the freeswitch(my own) variable as true ( This >> is used to identify whether the voice files are played >> successfully). >> => Wait in the loop until the variable is been set as >> one. >> => Set the Event lock as false >> >> => Trying to get the DTMF digits ( Have a assurance >> that all the voice files are played). >> >> The problem is, >> >> The above steps are working fine when the voice file count is >> lesser than or equal to 10. After the voice files are played only the >> variable(my own freeswitch) is set. Based on the variable I am doing further >> things. >> >> But when I tried to give the voice files count of more than >> 10 the variable has been set while starting to play back the first voice >> file itself . Because of this I am not able to proceed further. >> >> *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* >> >> *NOTE*: I also referred mod_file_string documentation. In that they >> specified 128 files can be used to play back the voice files using >> playback_delimiter option. >> >> Please help me................? >> Thanks in advance. >> >> >> -- >> Regards, >> Thangappan.M >> >> >> >> -- >> Regards, >> Thangappan.M >> > > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/5c44a390/attachment-0001.html From john_platts at hotmail.com Tue Nov 24 11:41:15 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 24 Nov 2009 13:41:15 -0600 Subject: [Freeswitch-users] Patch to allow gateways to be defined without the password parameter set Message-ID: I have modified sofia.c in mod_sofia so that I can define gateways without having to specify the password parameter. This is because I am using a SIP gateway that does not require SIP registration. The modified version still requires the password to be set on any gateway for which register is set to true. Attached is the diff file for these changes. _________________________________________________________________ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search?q=restaurants&form=MFESRP&publ=WLHMTAG&crea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 -------------- next part -------------- A non-text attachment was scrubbed... Name: sofia_password_patch.diff Type: application/octet-stream Size: 815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/1757affb/attachment.obj From brian at freeswitch.org Tue Nov 24 11:58:12 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Nov 2009 13:58:12 -0600 Subject: [Freeswitch-users] Patch to allow gateways to be defined without the password parameter set In-Reply-To: References: Message-ID: John, If the remote end doesn't require a username or password then you don't need to create a gateway to send a call to that endpoint. You can simply do sofia/profile/number at remoteip and it'll work. Also can you put the patch on jira via http://jira.freeswitch.org /b On Nov 24, 2009, at 1:41 PM, John Platts wrote: > > I have modified sofia.c in mod_sofia so that I can define gateways > without having to specify the password parameter. This is because I > am using a SIP gateway that does not require SIP registration. The > modified version still requires the password to be set on any > gateway for which register is set to true. Attached is the diff file > for these changes. > > _________________________________________________________________ > Bing brings you maps, menus, and reviews organized in one place. > http://www.bing.com/search? > q > = > restaurants > &form > = > MFESRP > &publ > = > WLHMTAG > &crea > = > TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 > < > sofia_password_patch.diff > >_______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Nov 24 12:15:37 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 15:15:37 -0500 Subject: [Freeswitch-users] DTMF Event is not coming while using playback terminators. In-Reply-To: <1452e2980911240059n4223bd48u487a80b8306131da@mail.gmail.com> References: <1452e2980911232322j106fe5eoe3efad59199f36e4@mail.gmail.com> <7EE00C1B-83CD-44C2-9CC0-F66120A9B534@jerris.com> <1452e2980911240059n4223bd48u487a80b8306131da@mail.gmail.com> Message-ID: <6DBE01B2-75A5-4A0F-9BF9-C7434114DD92@jerris.com> 1. can you supply a trace of this esl communications. 2. is it inband or rfc2833 dtmf ? MIke On Nov 24, 2009, at 3:59 AM, velusamy velu wrote: > Yes, I am using async mode only.. > > On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris wrote: > async? > > On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: > > > Dear All, > > I am using Perl ESL::IVR module to develop a simple IVR. I have filtered DTMF events. I have also set playback_terminators to cut the playback when giving the digits. I have faced problem that DTMF event has not come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. > > > > Why the 'COMMAND' event came event filter is on? > > How to avoid this event in ESL? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/00974a44/attachment.html From mike at jerris.com Tue Nov 24 12:19:17 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 15:19:17 -0500 Subject: [Freeswitch-users] Call Transfer Help Please In-Reply-To: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> References: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> Message-ID: On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: > Hi, > > I'm trying to setup call transfer for a phone without a transfer button. I was on IRC last night and got some pointers to how this is setup in dialplan.xml and features.xml and what "bind meta app" does. > > Once it became clear how the transfer is initiated and that the transfer, in the default config, can only be initiated by the "b" leg of the call, I was able to make this work as configured in the defaults, i.e, to initiate a transfer (for an internal call) from the dialled extension to a new extension. > > Now the problem . . . > > I have an incoming PSTN line that rings a group of extensions, what I want to be able to do is to give whoever answers the PSTN call ability to transfer the call on to another extension. > > There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch extension of 1000, it rings the extension phones in the group. > > I'd hoped that the default transfer setup would handle this without modification - the incoming call on extension 1000 would be the "a" leg, the answering extension would be the "b" leg and a transfer from "b" would work as per the default config. This does not work for me though. > > I'm struggling a bit with the "bind meta app" options and can't seem to make it do what I want. > > Could someone please confirm that what I'm trying to do is feasible and perhaps suggest the right parameters to use in dialplan.xml and features.xml please ? > > Relevant section in the "is_transfer" section in features.xml > > > And in default.xml from > to > > I've tried posting a call log to the Pastebin (11252/3) but there was an error - it looks like the dump was too big. Not sure what the maximum size on pastebin dumps is ? > > > My understanding (or lack of) of "a" and "b" are in the scenario described is not helping ... > > Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? Yes, the calling leg > Is the answering extension the "b" leg ? Yes > What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ? I don't understand this question > What is the correct "transfer" data string in features.xml ? > ditto > Or am I totally on the wrong track here ? > You should just need to make sure that the bind meta is called in this scenario so the b leg is able to do it, thats it. > If it is possible to do what I want, and changes are required to the dialplan.xml and/or features.xml files, is it possible to have different logic in there such that the actions are different whether it is the "a" leg or "b" leg that's requesting the transfer ? > > regards > Dave > > FreeSwitch Version 1.0.4 (14460) also, try the latest 1.0.5. pre release or svn trunk to confirm this is not an issue that has already been fixed. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b1fe436d/attachment.html From Prometheus001 at gmx.net Tue Nov 24 12:54:50 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 24 Nov 2009 21:54:50 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> Message-ID: <4B0C481A.8030309@gmx.net> Hello, I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console. I have explicitely set in the internal profile. When a phone is set up I get the following Snom Phone REGISTER => OpenSIPS=> Freeswitch Freeswitch OK => OpenSIPS=>Snom Phone Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch Freeswitch 202 Accepted => OpenSIPS=>Snom Phone Snom Phone PUBLISH => OpenSIPS=> Freeswitch Freeswitch 200 OK => OpenSIPS=>Snom Phone So presence generally seems to work. But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there. Any idea how to force the NOTIFY messages? Best regards Peter Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it: freeswitch at sip11.mydomain.com> nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 401 Unauthorized for REGISTER (6) nua(0x905a80): recv signal r_destroy nta_leg_destroy((nil)) nta: received PUBLISH sip:100 at sip1.mydomain.com SIP/2.0 (CSeq 3) nta: PUBLISH (3) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905f10): event i_publish 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905f10): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x905f10): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua(0x905f10): sent signal r_destroy nta: sent 200 OK for PUBLISH (3) nua(0x905f10): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:mod_sofia at 192.168.178.200:5062 SIP/2.0 (CSeq 2) nta: canonizing sip:mod_sofia at 192.168.178.200:5062 with contact nta: SUBSCRIBE (2) going to existing leg nua: nua_stack_process_request: entering nta: sent 200 OK for SUBSCRIBE (2) nua(0x905560): event i_subscribe 200 OK nua: nua_application_event: entering nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 8) nta: REGISTER (8) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5dc073ba0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5dc073ba0): sent signal r_respond nua(0x7fd5dc073ba0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua: nua_handle_destroy: entering nua(0x7fd5dc073ba0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (8) nua(0x7fd5dc073ba0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x8fc3d0): sent signal r_respond nua(0x8fc3d0): recv signal r_respond 200 OK nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (7) nua(0x8fc3d0): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:100 at sip1.mydomain.com;user=phone SIP/2.0 (CSeq 1) nta: SUBSCRIBE (1) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nta_leg_tcreate(0x7fd5dc03add0) nua(0x7fd5dc078b70): adding notify usage with event message-summary nua(0x7fd5dc078b70): event i_subscribe 100 Trying nua: nua_application_event: entering nua(): refresh notify after 3600 seconds (in [3600..3600]) nua: nua_respond: entering nua(0x7fd5dc078b70): sent signal r_respond nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted nua: nua_stack_set_params: entering nta: sent 202 Accepted for SUBSCRIBE (1) mayamatakeshi schrieb: > > On 9/12/09, *mayamatakeshi* > wrote: > > > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris > wrote: > > Following up, did a bug get created for this issue? > > > Hello, > yes. > http://jira.freeswitch.org/browse/MODSOFIA-26 > > > Just to simplify things in case someone searches the list: > Issue was solved on rev 14851. > Thank you all. > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From john_platts at hotmail.com Tue Nov 24 13:05:48 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 24 Nov 2009 15:05:48 -0600 Subject: [Freeswitch-users] Call forwarding problem Message-ID: I was having trouble doing call forwarding from my SIP phone that is connected to FreeSWITCH. It turns out that my SIP phone is actually sending 302 Moved Temporarily responses, but my SIP gateway does not support 302 Moved Temporarily or SIP REFER messages. How do I get FreeSWITCH to forward calls without sending 302 Moved Temporarily or SIP REFER messages? Here is the SIP debug from our gateway: Received: INVITE sip:+19725357722 at ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0 Record-Route: v: SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236 record-route: f: ;tag=dc7-13c4-3ec95a-3ad03068-3ec95a t: i: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989 CSeq: 1 INVITE Max-Forwards: 18 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208 m: c: application/SDP l: 210 P-Asserted-Identity: Privacy: none v=0 o=- 540754816 540754816 IN IP4 199.173.111.141 s=- c=IN IP4 199.173.111.141 t=0 0 m=audio 30056 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208 From: ;tag=dc7-13c4-3ec95a-3ad03068-3ec95a To: Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357722 at 168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: ;tag=14E93594-2488 To: Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246 Supported: timer,resource-priority,replaces Min-SE:? 1800 Cisco-Guid: 1208058493-3631485406-2892683743-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259096880 Contact: Expires: 180 Allow-Events: telephone-event Max-Forwards: 17 P-Asserted-Identity: Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 2925 1780 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.141 t=0 0 m=audio 30056 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.141 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: ;tag=14E93594-2488 To: Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Timestamp: 1259096880 0.000342 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M Content-Length: 0 Nov 24 15:08:00.419 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: ;tag=14E93594-2488 To: ;tag=49aF8vtgHme2c Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 P-Asserted-Identity: "19725357722" Nov 24 15:08:00.427 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357722 at 168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: ;tag=14E93594-2488 To: ;tag=49aF8vtgHme2c Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 _________________________________________________________________ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 From john_platts at hotmail.com Tue Nov 24 13:28:09 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 24 Nov 2009 15:28:09 -0600 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH In-Reply-To: <6F2A2A62-CE26-477F-B402-358F313A3EC3@jerris.com> References: , , <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org>, , <6F2A2A62-CE26-477F-B402-358F313A3EC3@jerris.com> Message-ID: I actually checked out revision 15654 today, and I was still getting problems with proxy media and bypass media in FreeSWITCH. ________________________________ > From: mike at jerris.com > Date: Tue, 24 Nov 2009 03:39:16 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH > > > > This was fixed in trunk yesterday about 8 hrs before you sent this message. (15619). Please update and try again. > > > Mike > > On Nov 23, 2009, at 11:33 PM, John Platts wrote: > > > I was using revision 15586. > > ---------------------------------------- > From: brian at freeswitch.org > Date: Mon, 23 Nov 2009 18:25:44 -0600 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH > > What rev exactly? > > /b > > On Nov 23, 2009, at 6:19 PM, John Platts wrote: > > > I actually checked out the latest version of FreeSWITCH in the SVN > repository. > > I have the following configured in /usr/local/freeswitch/conf/ > dialplan/default.xml: > > > _________________________________________________________________ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/ From brian at freeswitch.org Tue Nov 24 13:32:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Nov 2009 15:32:44 -0600 Subject: [Freeswitch-users] Call forwarding problem In-Reply-To: References: Message-ID: <633E77B1-2EC0-41A2-90C9-E884B59AFC99@freeswitch.org> You'll have to hairpin the media thru your machine usually if they won't accept either of those. /b On Nov 24, 2009, at 3:05 PM, John Platts wrote: > How do I get FreeSWITCH to forward calls without sending 302 Moved > Temporarily or SIP REFER messages? From brian at freeswitch.org Tue Nov 24 13:33:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Nov 2009 15:33:06 -0600 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH In-Reply-To: References: , , <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org>, , <6F2A2A62-CE26-477F-B402-358F313A3EC3@jerris.com> Message-ID: Are you sure you did a make current? and can you outline the issue in more detail? /b On Nov 24, 2009, at 3:28 PM, John Platts wrote: > > I actually checked out revision 15654 today, and I was still getting > problems with proxy media and bypass media in FreeSWITCH. From anthony.minessale at gmail.com Tue Nov 24 13:59:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Nov 2009 15:59:42 -0600 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B0C481A.8030309@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> Message-ID: <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> connect to FS with fs_cli Issue the command: /events MESSAGE_QUERY MESSAGE_WAITING then leave some voice mails probably you have a mis-configuration where the user/domain/profile cannot be resolved to the correct sofia profile to send the notify The event starts out as a freeswitch event and is translated into the notify by mod_sofia but only if it can match the event to a real sip user On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX wrote: > Hello, > > I have a similar problem with Freeswitch behind OpenSIPS as a load > balancer: > When registering, Freeeswitch does not send a MWI NOTIFY message for a > Phone which has voicemails. Even after recording a new voicemail there > is no NOTIFY message sent. And there are no error messages on the console. > > I have explicitely set > in the internal profile. > > When a phone is set up I get the following > Snom Phone REGISTER => OpenSIPS=> Freeswitch > Freeswitch OK => OpenSIPS=>Snom Phone > > Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch > Freeswitch 202 Accepted => OpenSIPS=>Snom Phone > > Snom Phone PUBLISH => OpenSIPS=> Freeswitch > Freeswitch 200 OK => OpenSIPS=>Snom Phone > So presence generally seems to work. > > But ngrepping the Network traffic there's no MWI NOTIFY message coming > from Freeswitch to any phone. > FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before > should be already there. > > Any idea how to force the NOTIFY messages? > > > Best regards > Peter > > Here's the debug Level9 output for nta and nua when a phone with VMs > registers, seems like there is no error in it: > > freeswitch at sip11.mydomain.com> nta: received REGISTER > sip:sip1.mydomain.com SIP/2.0 (CSeq 7) > nta: REGISTER (7) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x7fd5d409c8f0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x7fd5d409c8f0): sent signal r_respond > nua: nua_handle_destroy: entering > nua(0x7fd5d409c8f0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized > nua: nua_stack_set_params: entering > nta: sent 401 Unauthorized for REGISTER (7) > nta: timer set to 32000 ms > nua(0x7fd5d409c8f0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 6) > nta: REGISTER (6) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x905a80): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x905a80): sent signal r_respond > nua: nua_handle_destroy: entering > nua(0x905a80): recv signal r_respond 401 Unauthorized > nua(0x905a80): sent signal r_destroy > nua: nua_stack_set_params: entering > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 401 Unauthorized for REGISTER (6) > nua(0x905a80): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received PUBLISH sip:100 at sip1.mydomain.comSIP/2.0 (CSeq 3) > nta: PUBLISH (3) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x905f10): event i_publish 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x905f10): sent signal r_respond > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nua(0x905f10): recv signal r_respond 200 OK > nua: nua_stack_set_params: entering > nua(0x905f10): sent signal r_destroy > nta: sent 200 OK for PUBLISH (3) > nua(0x905f10): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received SUBSCRIBE sip:mod_sofia at 192.168.178.200:5062 SIP/2.0 (CSeq > 2) > nta: canonizing sip:mod_sofia at 192.168.178.200:5062 with contact > nta: SUBSCRIBE (2) going to existing leg > nua: nua_stack_process_request: entering > nta: sent 200 OK for SUBSCRIBE (2) > nua(0x905560): event i_subscribe 200 OK > nua: nua_application_event: entering > nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 8) > nta: REGISTER (8) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x7fd5dc073ba0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x7fd5dc073ba0): sent signal r_respond > nua(0x7fd5dc073ba0): recv signal r_respond 200 OK > nua: nua_stack_set_params: entering > nua: nua_handle_destroy: entering > nua(0x7fd5dc073ba0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 200 OK for REGISTER (8) > nua(0x7fd5dc073ba0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) > nta: REGISTER (7) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x8fc3d0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x8fc3d0): sent signal r_respond > nua(0x8fc3d0): recv signal r_respond 200 OK > nua: nua_handle_destroy: entering > nua: nua_stack_set_params: entering > nua(0x8fc3d0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 200 OK for REGISTER (7) > nua(0x8fc3d0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received SUBSCRIBE sip:100 at sip1.mydomain.com;user=phone > SIP/2.0 > (CSeq 1) > nta: SUBSCRIBE (1) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nta_leg_tcreate(0x7fd5dc03add0) > nua(0x7fd5dc078b70): adding notify usage with event message-summary > nua(0x7fd5dc078b70): event i_subscribe 100 Trying > nua: nua_application_event: entering > nua(): refresh notify after 3600 seconds (in [3600..3600]) > nua: nua_respond: entering > nua(0x7fd5dc078b70): sent signal r_respond > nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted > nua: nua_stack_set_params: entering > nta: sent 202 Accepted for SUBSCRIBE (1) > > > > > > mayamatakeshi schrieb: > > > > On 9/12/09, *mayamatakeshi* > > wrote: > > > > > > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris > > wrote: > > > > Following up, did a bug get created for this issue? > > > > > > Hello, > > yes. > > http://jira.freeswitch.org/browse/MODSOFIA-26 > > > > > > Just to simplify things in case someone searches the list: > > Issue was solved on rev 14851. > > Thank you all. > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/8dabf774/attachment-0001.html From john_platts at hotmail.com Tue Nov 24 14:04:07 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 24 Nov 2009 16:04:07 -0600 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript Message-ID: I have considered writing JavaScript code to bridge two calls together. However, I would like to perform custom handling of the 302 Moved Temporarily response. How do I handle the 302 Moved Temporarily response if I use JavaScript? _________________________________________________________________ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search?q=restaurants&form=MFESRP&publ=WLHMTAG&crea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 From anthony.minessale at gmail.com Tue Nov 24 14:20:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Nov 2009 16:20:03 -0600 Subject: [Freeswitch-users] Noise with openzap In-Reply-To: <3e6d7b0c0911240525o747f7b05y5c1f50ec6afe1179@mail.gmail.com> References: <3e6d7b0c0911240525o747f7b05y5c1f50ec6afe1179@mail.gmail.com> Message-ID: <191c3a030911241420m7f9d649dicc05a13171f4a05f@mail.gmail.com> you may want to try the latest release of both wanpipe and FS openzap is still a moving target since its in constant development from both the hardware and software end On Tue, Nov 24, 2009 at 7:25 AM, Steven Brown wrote: > Hi, > > I have an Ubuntu box running FS1.0.4 which has been processing a good > volume of calls between local users with soft phones (Xlite) and GSM > handsets via a number or Portech gateways, this has worked very well > for some time and audio quality is very good. > > I've now added a Sangoma A200 with 4 ports hooked up to 4 PSTN lines, > configured openzap and I can originate and answer calls on the the > openzap lines fine, however these calls via opezap all seem to suffer > from significant noise, the audio path works fine in both directions > but noise seems particularly bad at the local soft phone end. Quality > of all other calls through the box is fine though, any ideas > appreciated ?, > > NB A regular handset plugged directly into the PSTN lines has no problems > though > > Thanks > > Steve > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/6473f8e2/attachment.html From stevendt at primrosebank.net Tue Nov 24 14:36:36 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 24 Nov 2009 22:36:36 -0000 Subject: [Freeswitch-users] Call Transfer Help Please References: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> Message-ID: <5A7C3038838142B5A07F181C193754F6@bp1.ad.bp.com> Hi Mike, thanks for the reply. I am using the pre-compiled Windows binary - is there a 1.0.5 pre-release of that yet ? FreeSwitch reports its version as 1.0.4 (14460) but this is not correct, I was sure that I had previously loaded a later SVN Version, but just did it again, unless I'm not doing it right, the version number does not seem to be getting updated. The current build in the precompiled binaries area is reported to be 15604 and I've downloaded and installed that - although when the installer runs it tells me that it is version 15376. Either way, the "Version" command in FreeSwitch reports 1.0.4 (14460). The Transfer still does not work for me from the extension which answers the call. Sorry if my earlier questions were unclear ... "What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ?" What is the correct "transfer" data string in features.xml ? I don't understand this question(s) I was looking for clarification of the second two arguments in the bind_meta_app data call, i.e, that the "b" and "s" were the correct values and also that the "is transfer" "transfer" data argument was "-bleg" That is, that the arguments in the default dialplan are correct for this scenario - which they appear to be based on your previous reply to my query. So, is there anything else that I can check to see why this is not working ? regards Dave ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 24, 2009 8:19 PM Subject: Re: [Freeswitch-users] Call Transfer Help Please On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: Hi, I'm trying to setup call transfer for a phone without a transfer button. I was on IRC last night and got some pointers to how this is setup in dialplan.xml and features.xml and what "bind meta app" does. Once it became clear how the transfer is initiated and that the transfer, in the default config, can only be initiated by the "b" leg of the call, I was able to make this work as configured in the defaults, i.e, to initiate a transfer (for an internal call) from the dialled extension to a new extension. Now the problem . . . I have an incoming PSTN line that rings a group of extensions, what I want to be able to do is to give whoever answers the PSTN call ability to transfer the call on to another extension. There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch extension of 1000, it rings the extension phones in the group. I'd hoped that the default transfer setup would handle this without modification - the incoming call on extension 1000 would be the "a" leg, the answering extension would be the "b" leg and a transfer from "b" would work as per the default config. This does not work for me though. I'm struggling a bit with the "bind meta app" options and can't seem to make it do what I want. Could someone please confirm that what I'm trying to do is feasible and perhaps suggest the right parameters to use in dialplan.xml and features.xml please ? Relevant section in the "is_transfer" section in features.xml And in default.xml from to I've tried posting a call log to the Pastebin (11252/3) but there was an error - it looks like the dump was too big. Not sure what the maximum size on pastebin dumps is ? My understanding (or lack of) of "a" and "b" are in the scenario described is not helping ... Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? Yes, the calling leg Is the answering extension the "b" leg ? Yes What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ? I don't understand this question What is the correct "transfer" data string in features.xml ? ditto Or am I totally on the wrong track here ? You should just need to make sure that the bind meta is called in this scenario so the b leg is able to do it, thats it. If it is possible to do what I want, and changes are required to the dialplan.xml and/or features.xml files, is it possible to have different logic in there such that the actions are different whether it is the "a" leg or "b" leg that's requesting the transfer ? regards Dave FreeSwitch Version 1.0.4 (14460) also, try the latest 1.0.5. pre release or svn trunk to confirm this is not an issue that has already been fixed. Mike ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/57d90a13/attachment.html From Prometheus001 at gmx.net Tue Nov 24 14:56:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 24 Nov 2009 23:56:25 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> Message-ID: <4B0C6499.4060504@gmx.net> Anthony, thanks for the hint, I receive events like the following RECV EVENT Event-Name: MESSAGE_WAITING Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 FreeSWITCH-Hostname: sip11.mydomain.com FreeSWITCH-IPv4: 192.168.178.200 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-24 23:33:13 Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT Event-Date-Timestamp: 1259101993918617 Event-Calling-File: mod_voicemail.c Event-Calling-Function: update_mwi Event-Calling-Line-Number: 1738 MWI-Messages-Waiting: yes MWI-Message-Account: 200 at sip1.mydomain.com MWI-Voice-Message: 4/1 (0/0) I think the problem may be the Freeswitch cluster we are working with. All phones register with realm (e.g. 200 at sip1.mydomain.com). The FS hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. With xml_curl we ensure that for both domain names a directory entry is passed back. That way it works nicely with registering phones, receiving voicemails, recording voicemails etc. but not for MWI. For recording and querying voicemails we use the realm instead of the domain name and that way it works. When a voicemail has finished recording - and at the time the above message occurs - I see 2 directory xml_curl requests with Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id One I expect is for retrieving the MWI data and the other one for sending the VM email (which is sucessfully sent). Any hint how we can workaround this? Or is there a parameter to tell mod_voicemail that is should use the realm instead of the local hostname for sending MWI? Best regards Peter Anthony Minessale schrieb: > connect to FS with fs_cli > > Issue the command: > > /events MESSAGE_QUERY MESSAGE_WAITING > > then leave some voice mails > > probably you have a mis-configuration where the user/domain/profile > cannot be resolved to the correct > sofia profile to send the notify > > The event starts out as a freeswitch event and is translated into the > notify by mod_sofia but only if it can > match the event to a real sip user > > > > > On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX > wrote: > > Hello, > > I have a similar problem with Freeswitch behind OpenSIPS as a load > balancer: > When registering, Freeeswitch does not send a MWI NOTIFY message for a > Phone which has voicemails. Even after recording a new voicemail there > is no NOTIFY message sent. And there are no error messages on the > console. > > I have explicitely set > in the internal > profile. > > When a phone is set up I get the following > Snom Phone REGISTER => OpenSIPS=> Freeswitch > Freeswitch OK => OpenSIPS=>Snom Phone > > Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch > Freeswitch 202 Accepted => OpenSIPS=>Snom Phone > > Snom Phone PUBLISH => OpenSIPS=> Freeswitch > Freeswitch 200 OK => OpenSIPS=>Snom Phone > So presence generally seems to work. > > But ngrepping the Network traffic there's no MWI NOTIFY message coming > from Freeswitch to any phone. > FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before > should be already there. > > Any idea how to force the NOTIFY messages? > > > Best regards > Peter > > Here's the debug Level9 output for nta and nua when a phone with VMs > registers, seems like there is no error in it: > > freeswitch at sip11.mydomain.com > > nta: received REGISTER > sip:sip1.mydomain.com SIP/2.0 (CSeq 7) > nta: REGISTER (7) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x7fd5d409c8f0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x7fd5d409c8f0): sent signal r_respond > nua: nua_handle_destroy: entering > nua(0x7fd5d409c8f0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized > nua: nua_stack_set_params: entering > nta: sent 401 Unauthorized for REGISTER (7) > nta: timer set to 32000 ms > nua(0x7fd5d409c8f0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received REGISTER sip:sip1.mydomain.com > SIP/2.0 (CSeq 6) > nta: REGISTER (6) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x905a80): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x905a80): sent signal r_respond > nua: nua_handle_destroy: entering > nua(0x905a80): recv signal r_respond 401 Unauthorized > nua(0x905a80): sent signal r_destroy > nua: nua_stack_set_params: entering > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 401 Unauthorized for REGISTER (6) > nua(0x905a80): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received PUBLISH sip:100 at sip1.mydomain.com > SIP/2.0 (CSeq 3) > nta: PUBLISH (3) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x905f10): event i_publish 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x905f10): sent signal r_respond > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nua(0x905f10): recv signal r_respond 200 OK > nua: nua_stack_set_params: entering > nua(0x905f10): sent signal r_destroy > nta: sent 200 OK for PUBLISH (3) > nua(0x905f10): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received SUBSCRIBE sip:mod_sofia at 192.168.178.200:5062 > SIP/2.0 (CSeq 2) > nta: canonizing sip:mod_sofia at 192.168.178.200:5062 > with contact > nta: SUBSCRIBE (2) going to existing leg > nua: nua_stack_process_request: entering > nta: sent 200 OK for SUBSCRIBE (2) > nua(0x905560): event i_subscribe 200 OK > nua: nua_application_event: entering > nta: received REGISTER sip:sip1.mydomain.com > SIP/2.0 (CSeq 8) > nta: REGISTER (8) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x7fd5dc073ba0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x7fd5dc073ba0): sent signal r_respond > nua(0x7fd5dc073ba0): recv signal r_respond 200 OK > nua: nua_stack_set_params: entering > nua: nua_handle_destroy: entering > nua(0x7fd5dc073ba0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 200 OK for REGISTER (8) > nua(0x7fd5dc073ba0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received REGISTER sip:sip1.mydomain.com > SIP/2.0 (CSeq 7) > nta: REGISTER (7) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x8fc3d0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x8fc3d0): sent signal r_respond > nua(0x8fc3d0): recv signal r_respond 200 OK > nua: nua_handle_destroy: entering > nua: nua_stack_set_params: entering > nua(0x8fc3d0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 200 OK for REGISTER (7) > nua(0x8fc3d0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received SUBSCRIBE sip:100 at sip1.mydomain.com > ;user=phone SIP/2.0 > (CSeq 1) > nta: SUBSCRIBE (1) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nta_leg_tcreate(0x7fd5dc03add0) > nua(0x7fd5dc078b70): adding notify usage with event message-summary > nua(0x7fd5dc078b70): event i_subscribe 100 Trying > nua: nua_application_event: entering > nua(): refresh notify after 3600 seconds (in [3600..3600]) > nua: nua_respond: entering > nua(0x7fd5dc078b70): sent signal r_respond > nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted > nua: nua_stack_set_params: entering > nta: sent 202 Accepted for SUBSCRIBE (1) > > > > > > mayamatakeshi schrieb: > > > > On 9/12/09, *mayamatakeshi* > > >> wrote: > > > > > > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris > > > >> wrote: > > > > Following up, did a bug get created for this issue? > > > > > > Hello, > > yes. > > http://jira.freeswitch.org/browse/MODSOFIA-26 > > > > > > Just to simplify things in case someone searches the list: > > Issue was solved on rev 14851. > > Thank you all. > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From john_platts at hotmail.com Tue Nov 24 15:24:23 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 24 Nov 2009 17:24:23 -0600 Subject: [Freeswitch-users] Call forwarding problem In-Reply-To: <633E77B1-2EC0-41A2-90C9-E884B59AFC99@freeswitch.org> References: , <633E77B1-2EC0-41A2-90C9-E884B59AFC99@freeswitch.org> Message-ID: Is there any way to tell FreeSWITCH to do the following when 302 Moved Temporarily is sent to FreeSWITCH: - End the session between FreeSWITCH and the phone - Bridge the original session with the number that the call is forwarded to ---------------------------------------- > From: brian at freeswitch.org > Date: Tue, 24 Nov 2009 15:32:44 -0600 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call forwarding problem > > You'll have to hairpin the media thru your machine usually if they > won't accept either of those. > > /b > > On Nov 24, 2009, at 3:05 PM, John Platts wrote: > >> How do I get FreeSWITCH to forward calls without sending 302 Moved >> Temporarily or SIP REFER messages? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/ From fanatikneo at gmx.de Tue Nov 24 14:35:17 2009 From: fanatikneo at gmx.de (Jan Thiemo Fricke) Date: Tue, 24 Nov 2009 23:35:17 +0100 Subject: [Freeswitch-users] mod_conference kick to abort invitations Message-ID: <000001ca6d56$66037c80$320a7580$@de> Hi members, I'm controlling freeswitch with the conference module via xmlrpc. Is it desired that the kick command can only kick users that are connected to the conference? Is there no chance abort an invitation? The kick command has no effect until the person I invited with the dial command is connected. Thanks in advance! Jan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/1dc073b9/attachment.html From stevendt at primrosebank.net Tue Nov 24 16:11:24 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 25 Nov 2009 00:11:24 -0000 Subject: [Freeswitch-users] Call Transfer Help Please References: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> <5A7C3038838142B5A07F181C193754F6@bp1.ad.bp.com> Message-ID: Hi again folks, I have posted a dump into the Pastebin (11276), could someone have a look and perhaps suggest where the problem might be please ? I'm sure you'll be able to work it out, but the log is for a call where :- incoming on PSTN Line (ext 1000) Group exts, 111, 1001, 1001 Answered on 111 and requested transfer to 1001 with no success regards Dave ----- Original Message ----- From: Dave Stevenson To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 24, 2009 10:36 PM Subject: Re: [Freeswitch-users] Call Transfer Help Please Hi Mike, thanks for the reply. I am using the pre-compiled Windows binary - is there a 1.0.5 pre-release of that yet ? FreeSwitch reports its version as 1.0.4 (14460) but this is not correct, I was sure that I had previously loaded a later SVN Version, but just did it again, unless I'm not doing it right, the version number does not seem to be getting updated. The current build in the precompiled binaries area is reported to be 15604 and I've downloaded and installed that - although when the installer runs it tells me that it is version 15376. Either way, the "Version" command in FreeSwitch reports 1.0.4 (14460). The Transfer still does not work for me from the extension which answers the call. Sorry if my earlier questions were unclear ... "What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ?" What is the correct "transfer" data string in features.xml ? I don't understand this question(s) I was looking for clarification of the second two arguments in the bind_meta_app data call, i.e, that the "b" and "s" were the correct values and also that the "is transfer" "transfer" data argument was "-bleg" That is, that the arguments in the default dialplan are correct for this scenario - which they appear to be based on your previous reply to my query. So, is there anything else that I can check to see why this is not working ? regards Dave ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 24, 2009 8:19 PM Subject: Re: [Freeswitch-users] Call Transfer Help Please On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: Hi, I'm trying to setup call transfer for a phone without a transfer button. I was on IRC last night and got some pointers to how this is setup in dialplan.xml and features.xml and what "bind meta app" does. Once it became clear how the transfer is initiated and that the transfer, in the default config, can only be initiated by the "b" leg of the call, I was able to make this work as configured in the defaults, i.e, to initiate a transfer (for an internal call) from the dialled extension to a new extension. Now the problem . . . I have an incoming PSTN line that rings a group of extensions, what I want to be able to do is to give whoever answers the PSTN call ability to transfer the call on to another extension. There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch extension of 1000, it rings the extension phones in the group. I'd hoped that the default transfer setup would handle this without modification - the incoming call on extension 1000 would be the "a" leg, the answering extension would be the "b" leg and a transfer from "b" would work as per the default config. This does not work for me though. I'm struggling a bit with the "bind meta app" options and can't seem to make it do what I want. Could someone please confirm that what I'm trying to do is feasible and perhaps suggest the right parameters to use in dialplan.xml and features.xml please ? Relevant section in the "is_transfer" section in features.xml And in default.xml from to I've tried posting a call log to the Pastebin (11252/3) but there was an error - it looks like the dump was too big. Not sure what the maximum size on pastebin dumps is ? My understanding (or lack of) of "a" and "b" are in the scenario described is not helping ... Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? Yes, the calling leg Is the answering extension the "b" leg ? Yes What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ? I don't understand this question What is the correct "transfer" data string in features.xml ? ditto Or am I totally on the wrong track here ? You should just need to make sure that the bind meta is called in this scenario so the b leg is able to do it, thats it. If it is possible to do what I want, and changes are required to the dialplan.xml and/or features.xml files, is it possible to have different logic in there such that the actions are different whether it is the "a" leg or "b" leg that's requesting the transfer ? regards Dave FreeSwitch Version 1.0.4 (14460) also, try the latest 1.0.5. pre release or svn trunk to confirm this is not an issue that has already been fixed. Mike ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/f4e74177/attachment-0001.html From lei.tlfly at gmail.com Tue Nov 24 18:03:04 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 25 Nov 2009 10:03:04 +0800 Subject: [Freeswitch-users] How to run IVR application In-Reply-To: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> References: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> Message-ID: <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> you can do this in follow steps: 1.edit default.xml diaplan config file in your fs config directory(FS/conf/dialplan/default.xml), and section 2. edit your ivr script, your can refer to http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in lua. 3. connect your sip phone to fs, and dial 114, this will launch your ivr application 2009/11/24 ovvenkat > Hi to all, > > I am very new this platform . I just downloaded freeswitch to my windows xp > machine , compiled successfully and run. After that I dont have any idea > what to do :( . I am not finding simple kind of tutorial on the net. could > you please suggest me, how I have to proceed. My requirement is; I need to > run IVR application on machine using SIP phone. I am very sorry to my bad > English. > > Thanks and Regards, > Venkat. > > -- > > If you have come to help me, you are wasting your time. > If you have come to because your liberation is bound up in mine, we can > work together. > > > Regards > Venkatesan OV. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/c8e49dd1/attachment.html From jlenk at frontiernet.net Tue Nov 24 19:54:28 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 24 Nov 2009 19:54:28 -0800 (PST) Subject: [Freeswitch-users] Call Transfer Help Please In-Reply-To: References: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> <5A7C3038838142B5A07F181C193754F6@bp1.ad.bp.com> Message-ID: <1259121268571-4062810.post@n2.nabble.com> I do not see the meta app getting added in your log -> Dialplan: sofia/internal/1000 at 192.168.1.50 Action bind_meta_app(* Without this no meta actions will occur Dave Stevenson wrote: > > Hi again folks, > > I have posted a dump into the Pastebin (11276), could someone have a look > and perhaps suggest where the problem might be please ? > > I'm sure you'll be able to work it out, but the log is for a call where :- > > incoming on PSTN Line (ext 1000) > Group exts, 111, 1001, 1001 > Answered on 111 and requested transfer to 1001 with no success > > regards > Dave > > > ----- Original Message ----- > From: Dave Stevenson > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, November 24, 2009 10:36 PM > Subject: Re: [Freeswitch-users] Call Transfer Help Please > > > Hi Mike, > > thanks for the reply. I am using the pre-compiled Windows binary - is > there a 1.0.5 pre-release of that yet ? > > FreeSwitch reports its version as 1.0.4 (14460) but this is not correct, > I was sure that I had previously loaded a later SVN Version, but just did > it again, unless I'm not doing it right, the version number does not seem > to be getting updated. The current build in the precompiled binaries area > is reported to be 15604 and I've downloaded and installed that - although > when the installer runs it tells me that it is version 15376. Either way, > the "Version" command in FreeSwitch reports 1.0.4 (14460). > > The Transfer still does not work for me from the extension which answers > the call. > > Sorry if my earlier questions were unclear ... > "What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml > ?" > What is the correct "transfer" data string in features.xml ? > I don't understand this question(s) > > I was looking for clarification of the second two arguments in the > bind_meta_app data call, i.e, that the "b" and "s" were the correct values > and also that the "is transfer" "transfer" data argument was "-bleg" > > That is, that the arguments in the default dialplan are correct for this > scenario - which they appear to be based on your previous reply to my > query. > > So, is there anything else that I can check to see why this is not > working ? > > > regards > Dave > > > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, November 24, 2009 8:19 PM > Subject: Re: [Freeswitch-users] Call Transfer Help Please > > > > > On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: > > > Hi, > > I'm trying to setup call transfer for a phone without a transfer > button. I was on IRC last night and got some pointers to how this is setup > in dialplan.xml and features.xml and what "bind meta app" does. > > Once it became clear how the transfer is initiated and that the > transfer, in the default config, can only be initiated by the "b" leg of > the call, I was able to make this work as configured in the defaults, i.e, > to initiate a transfer (for an internal call) from the dialled extension > to a new extension. > > Now the problem . . . > > I have an incoming PSTN line that rings a group of extensions, what > I want to be able to do is to give whoever answers the PSTN call ability > to transfer the call on to another extension. > > There is an ATA (Linksys SPA3101) set up on the PSTN line with a > FreeSwitch extension of 1000, it rings the extension phones in the group. > > I'd hoped that the default transfer setup would handle this without > modification - the incoming call on extension 1000 would be the "a" leg, > the answering extension would be the "b" leg and a transfer from "b" would > work as per the default config. This does not work for me though. > > I'm struggling a bit with the "bind meta app" options and can't seem > to make it do what I want. > > Could someone please confirm that what I'm trying to do is feasible > and perhaps suggest the right parameters to use in dialplan.xml and > features.xml please ? > > Relevant section in the "is_transfer" section in features.xml > > > And in default.xml from > to > > > I've tried posting a call log to the Pastebin (11252/3) but there > was an error - it looks like the dump was too big. Not sure what the > maximum size on pastebin dumps is ? > > > My understanding (or lack of) of "a" and "b" are in the scenario > described is not helping ... > > Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? > > > Yes, the calling leg > > > Is the answering extension the "b" leg ? > > > Yes > > > What are the correct LISTEN_TO and RESPOND_ON entries in > dialplan.xml ? > > > I don't understand this question > > > What is the correct "transfer" data string in features.xml ? > > > > ditto > > > Or am I totally on the wrong track here ? > > > > You should just need to make sure that the bind meta is called in this > scenario so the b leg is able to do it, thats it. > > > If it is possible to do what I want, and changes are required to the > dialplan.xml and/or features.xml files, is it possible to have different > logic in there such that the actions are different whether it is the "a" > leg or "b" leg that's requesting the transfer ? > > regards > Dave > > FreeSwitch Version 1.0.4 (14460) > > > also, try the latest 1.0.5. pre release or svn trunk to confirm this > is not an issue that has already been fixed. > > > Mike > > > > > ---------------------------------------------------------------------------- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------------ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Call-Transfer-Help-Please-tp4056930p4062810.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Tue Nov 24 19:55:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Nov 2009 19:55:31 -0800 Subject: [Freeswitch-users] How to run IVR application In-Reply-To: <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> References: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> Message-ID: <87f2f3b90911241955v4e726111ked993c8dbb556f99@mail.gmail.com> On Tue, Nov 24, 2009 at 6:03 PM, Lei Tang wrote: > you can do this in follow steps: > 1.edit default.xml diaplan config file in your fs config > directory(FS/conf/dialplan/default.xml), and section > > > > > > 2. edit your ivr script, your can refer to > http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in > lua. > 3. connect your sip phone to fs, and dial 114, this will launch your ivr > application > > You can also do IVRs with static XML. I recommend you try out the demo IVR by dialing 5000. Now go look at the two main files that we used to build that IVR: conf/autoload_configs/ivr.conf.xml (menu structure) conf/lang/en/demo/demo-ivr.xml (phrase macros) it's overwhelming at first, however once you get the hang of it you'll appreciate how powerful it is. The wiki and the sample XML config files have lots of information so be sure to read as much as you can and try things. You can't break anything. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/3e78aadb/attachment.html From jlenk at frontiernet.net Tue Nov 24 20:35:59 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 24 Nov 2009 20:35:59 -0800 (PST) Subject: [Freeswitch-users] register timeout / cisco 7960 In-Reply-To: <367751820911231434j36b9846dk46d058ddb77c634@mail.gmail.com> References: <367751820911231434j36b9846dk46d058ddb77c634@mail.gmail.com> Message-ID: <1259123759153-4062958.post@n2.nabble.com> People commonly use 60 sec registration refreshes to keep NAT routers happy Phillip Jones-2 wrote: > > hi there, > > I have set up some cisco 7960 up with fs. They work fine - but the only > way > I can keep them registered is to set the "timer_register_expires" in the > Cisco cfg file to something really short like 10s. > > Does anyone know the default register timeout for fs? And where I might > change this in fs? > > Thanks! > > > Phil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/register-timeout-cisco-7960-tp4054546p4062958.html Sent from the freeswitch-users mailing list archive at Nabble.com. From thangappan143 at gmail.com Tue Nov 24 22:09:24 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Wed, 25 Nov 2009 11:39:24 +0530 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> Message-ID: <7aa29e790911242209w7ee2912bhbde4b3475147628d@mail.gmail.com> FreeSWITCH version: freeswitch 1.0.4 I am using ESL library I attached the example Perl script which does the same steps that I posted already. ( Sample.pl) I supplied the log , Here I attached the output of the ESL log. (Output.txt) Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. But in the output I got only 2,4,5,4 ( DTMF 1 is missed) Output of Perl code could be like Wait for response time out EVENT [COMMAND] Wait for response time out EVENT [DTMF] DTMF digit 2 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 4 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 5 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 4 (2000) Wait for inter digit time out Buffer: 2454 BYE Why the first digit(1) is missed here? In ESL log there is no digit called 1 why? Why the COMMAND event is received instead of DTMF? How can I get all DTMF digits? On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M wrote: > The reason for waiting only for DTMF event is to handle the time outs in > the IVR concept like response and inter digit time out. Using our own logic > we 10 voice files in each play back if the voice files are more than 10. Now > it works fine. > > Now the new problem has been raised. The problem is we are filtering only > for DTMF events but we are getting COMMAND event . Because of this the DTMF > digits are missing at the time . I am not able to proceed further. We are > in the critical situation. > > Why this command event is occurring? > How can I restrict this? > What are the information it has? > How can I get all the information in it ? ( If command event has info) > > Help me............ > > > On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: > >> I am waiting only for DTMF events. That's why I am setting freeswitch >> variable for knowing whether the playback has done. >> >> My question is "why this freeswitch variable is not setting properly when >> I play back more than 10 files using playback_delimiter option?". >> >> When I play back lesser than ten voice files the variable has been set >> properly. What could be the reason? >> >> >> >> ---------- Forwarded message ---------- >> From: Thangappan.M >> Date: Sat, Nov 21, 2009 at 2:52 PM >> Subject: Problem while playing more than 10 voice files using playback >> To: freeswitch-users >> >> >> Dear all, >> >> I am in the process of implementing IVR using event outbound >> socket (async mode). >> I have implemented using Perl language. >> >> I did the following steps: >> => Set the playback_delimiter variable >> => Set the playback_sleep_val variable >> => Set the event lock as true >> => Set the freeswitch ( my own) variable as zero >> => Wait in the loop until the variable is been set as >> zero >> => Playback the voice files ( Here I combined the >> voice files with the delimiter value if more than one voice files are there) >> => Set the freeswitch(my own) variable as true ( This >> is used to identify whether the voice files are played >> successfully). >> => Wait in the loop until the variable is been set as >> one. >> => Set the Event lock as false >> >> => Trying to get the DTMF digits ( Have a assurance >> that all the voice files are played). >> >> The problem is, >> >> The above steps are working fine when the voice file count is >> lesser than or equal to 10. After the voice files are played only the >> variable(my own freeswitch) is set. Based on the variable I am doing further >> things. >> >> But when I tried to give the voice files count of more than >> 10 the variable has been set while starting to play back the first voice >> file itself . Because of this I am not able to proceed further. >> >> *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* >> >> *NOTE*: I also referred mod_file_string documentation. In that they >> specified 128 files can be used to play back the voice files using >> playback_delimiter option. >> >> Please help me................? >> Thanks in advance. >> >> >> -- >> Regards, >> Thangappan.M >> >> >> >> -- >> Regards, >> Thangappan.M >> > > > > -- > Regards, > Thangappan.M > -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/e557b247/attachment-0001.html -------------- next part -------------- [DEBUG] src/esl.c:995 esl_send() SEND myevents [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK Events Enabled] [DEBUG] src/esl.c:995 esl_send() SEND filter Event-Name DTMF [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK filter added. [Event-Name]=[DTMF]] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: answer event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: set execute-app-arg: playback_terminators=0123456789 event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: set execute-app-arg: playback_delimiter=! event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: set execute-app-arg: playback_sleep_val=1 event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: set execute-app-arg: IVRFlag=0 event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: playback execute-app-arg: /FreeSWITCH-IVR/VoiceFile/English/iconnectform.wav!/FreeSWITCH-IVR/VoiceFile/English/dial1.wav!/FreeSWITCH-IVR/VoiceFile/English/tcurrform.wav!/FreeSWITCH-IVR/VoiceFile/English/dial2.wav!/FreeSWITCH-IVR/VoiceFile/English/licpayform.wav!/FreeSWITCH-IVR/VoiceFile/English/dial3.wav!/FreeSWITCH-IVR/VoiceFile/English/bsnlform.wav!/FreeSWITCH-IVR/VoiceFile/English/dial4.wav!/FreeSWITCH-IVR/VoiceFile/English/encashform.wav!/FreeSWITCH-IVR/VoiceFile/English/dial5.wav event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: set execute-app-arg: IVRFlag=1 event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1516] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1516] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Name] = [DTMF] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Core-UUID] = [cfbc5248-d983-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-Hostname] = [debian] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv4] = [192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv6] = [::1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Local] = [2009-11-25 11:11:49] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-GMT] = [Wed, 25 Nov 2009 05:41:49 GMT] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Timestamp] = [1259127709436105] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-File] = [switch_channel.c] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Function] = [switch_channel_dequeue_dtmf] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Line-Number] = [388] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State] = [CS_EXECUTE] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State-Number] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Presence-Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Answer-State] = [answered] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Username] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Dialplan] = [XML] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Name] = [thangappan] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Number] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Network-Addr] = [192.168.8.100] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Destination-Number] = [200] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Source] = [mod_sofia] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Context] = [default] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Index] = [1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Answered-Time] = [1259127708528257] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Media-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Hangup-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Transfer-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Screen-Bit] = [true] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Name] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Number] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Digit] = [2] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Duration] = [2000] [DEBUG] src/esl.c:963 esl_recv_event() RECV EVENT Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-25 11:11:49 Event-Date-GMT: Wed, 25 Nov 2009 05:41:49 GMT Event-Date-Timestamp: 1259127709436105 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012 at 192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012 at 192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 2 DTMF-Duration: 2000 [DEBUG] src/esl.c:971 esl_recv_event() RECV MESSAGE Event-Name: COMMAND Content-Length: 1516 Content-Type: text/event-plain Content-Length: 1516 Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-11-25%2011%3A11%3A49 Event-Date-GMT: Wed,%2025%20Nov%202009%2005%3A41%3A49%20GMT Event-Date-Timestamp: 1259127709436105 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012%40192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012%40192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 2 DTMF-Duration: 2000 [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1516] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Name] = [DTMF] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Core-UUID] = [cfbc5248-d983-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-Hostname] = [debian] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv4] = [192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv6] = [::1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Local] = [2009-11-25 11:11:49] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-GMT] = [Wed, 25 Nov 2009 05:41:49 GMT] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Timestamp] = [1259127709636074] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-File] = [switch_channel.c] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Function] = [switch_channel_dequeue_dtmf] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Line-Number] = [388] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State] = [CS_EXECUTE] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State-Number] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Presence-Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Answer-State] = [answered] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Username] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Dialplan] = [XML] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Name] = [thangappan] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Number] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Network-Addr] = [192.168.8.100] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Destination-Number] = [200] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Source] = [mod_sofia] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Context] = [default] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Index] = [1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Answered-Time] = [1259127708528257] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Media-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Hangup-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Transfer-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Screen-Bit] = [true] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Name] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Number] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Digit] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Duration] = [2000] [DEBUG] src/esl.c:963 esl_recv_event() RECV EVENT Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-25 11:11:49 Event-Date-GMT: Wed, 25 Nov 2009 05:41:49 GMT Event-Date-Timestamp: 1259127709636074 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012 at 192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012 at 192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 4 DTMF-Duration: 2000 [DEBUG] src/esl.c:971 esl_recv_event() RECV MESSAGE Event-Name: COMMAND Content-Length: 1516 Content-Type: text/event-plain Content-Length: 1516 Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-11-25%2011%3A11%3A49 Event-Date-GMT: Wed,%2025%20Nov%202009%2005%3A41%3A49%20GMT Event-Date-Timestamp: 1259127709636074 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012%40192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012%40192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 4 DTMF-Duration: 2000 [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1516] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Name] = [DTMF] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Core-UUID] = [cfbc5248-d983-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-Hostname] = [debian] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv4] = [192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv6] = [::1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Local] = [2009-11-25 11:11:51] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-GMT] = [Wed, 25 Nov 2009 05:41:51 GMT] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Timestamp] = [1259127711031853] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-File] = [switch_channel.c] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Function] = [switch_channel_dequeue_dtmf] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Line-Number] = [388] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State] = [CS_EXECUTE] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State-Number] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Presence-Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Answer-State] = [answered] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Username] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Dialplan] = [XML] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Name] = [thangappan] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Number] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Network-Addr] = [192.168.8.100] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Destination-Number] = [200] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Source] = [mod_sofia] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Context] = [default] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Index] = [1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Answered-Time] = [1259127708528257] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Media-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Hangup-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Transfer-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Screen-Bit] = [true] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Name] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Number] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Digit] = [5] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Duration] = [2000] [DEBUG] src/esl.c:963 esl_recv_event() RECV EVENT Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-25 11:11:51 Event-Date-GMT: Wed, 25 Nov 2009 05:41:51 GMT Event-Date-Timestamp: 1259127711031853 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012 at 192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012 at 192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 5 DTMF-Duration: 2000 [DEBUG] src/esl.c:971 esl_recv_event() RECV MESSAGE Event-Name: COMMAND Content-Length: 1516 Content-Type: text/event-plain Content-Length: 1516 Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-11-25%2011%3A11%3A51 Event-Date-GMT: Wed,%2025%20Nov%202009%2005%3A41%3A51%20GMT Event-Date-Timestamp: 1259127711031853 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012%40192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012%40192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 5 DTMF-Duration: 2000 [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1516] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Name] = [DTMF] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Core-UUID] = [cfbc5248-d983-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-Hostname] = [debian] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv4] = [192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv6] = [::1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Local] = [2009-11-25 11:11:52] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-GMT] = [Wed, 25 Nov 2009 05:41:52 GMT] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Timestamp] = [1259127712831585] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-File] = [switch_channel.c] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Function] = [switch_channel_dequeue_dtmf] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Line-Number] = [388] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State] = [CS_EXECUTE] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State-Number] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Presence-Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Answer-State] = [answered] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Username] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Dialplan] = [XML] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Name] = [thangappan] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Number] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Network-Addr] = [192.168.8.100] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Destination-Number] = [200] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Source] = [mod_sofia] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Context] = [default] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Index] = [1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Answered-Time] = [1259127708528257] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Media-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Hangup-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Transfer-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Screen-Bit] = [true] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Name] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Number] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Digit] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Duration] = [2000] [DEBUG] src/esl.c:963 esl_recv_event() RECV EVENT Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-25 11:11:52 Event-Date-GMT: Wed, 25 Nov 2009 05:41:52 GMT Event-Date-Timestamp: 1259127712831585 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012 at 192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012 at 192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 4 DTMF-Duration: 2000 [DEBUG] src/esl.c:971 esl_recv_event() RECV MESSAGE Event-Name: COMMAND Content-Length: 1516 Content-Type: text/event-plain Content-Length: 1516 Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-11-25%2011%3A11%3A52 Event-Date-GMT: Wed,%2025%20Nov%202009%2005%3A41%3A52%20GMT Event-Date-Timestamp: 1259127712831585 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012%40192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012%40192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 4 DTMF-Duration: 2000 [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/disconnect-notice] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Controlled-Session-UUID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Disposition] = [disconnect] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [67] -------------- next part -------------- A non-text attachment was scrubbed... Name: Sample.pl Type: text/x-perl Size: 4984 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/e557b247/attachment-0001.bin From thangappan143 at gmail.com Tue Nov 24 22:18:27 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Wed, 25 Nov 2009 11:48:27 +0530 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911242209w7ee2912bhbde4b3475147628d@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> <7aa29e790911242209w7ee2912bhbde4b3475147628d@mail.gmail.com> Message-ID: <7aa29e790911242218l580b90eem3ec50676dfbc5536@mail.gmail.com> The example script is there in the following link http://pastebin.com/f332f2fda In the previous post I have attached it. But it was not shown. 2009/11/25 Thangappan.M > FreeSWITCH version: freeswitch 1.0.4 > I am using ESL library > I attached the example Perl script which does the same steps that I posted > already. ( Sample.pl) > I supplied the log , Here I attached the output of the ESL log. > (Output.txt) > > Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. > But in the output I got only 2,4,5,4 ( DTMF 1 is missed) > > Output of Perl code could be like > > Wait for response time out > EVENT [COMMAND] > Wait for response time out > EVENT [DTMF] > DTMF digit 2 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 5 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > Buffer: 2454 > BYE > > Why the first digit(1) is missed here? > In ESL log there is no digit called 1 why? > Why the COMMAND event is received instead of DTMF? > How can I get all DTMF digits? > > > > > > > > > > > > > > > > > On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M wrote: > >> The reason for waiting only for DTMF event is to handle the time outs in >> the IVR concept like response and inter digit time out. Using our own logic >> we 10 voice files in each play back if the voice files are more than 10. Now >> it works fine. >> >> Now the new problem has been raised. The problem is we are filtering only >> for DTMF events but we are getting COMMAND event . Because of this the DTMF >> digits are missing at the time . I am not able to proceed further. We are >> in the critical situation. >> >> Why this command event is occurring? >> How can I restrict this? >> What are the information it has? >> How can I get all the information in it ? ( If command event has info) >> >> Help me............ >> >> >> On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: >> >>> I am waiting only for DTMF events. That's why I am setting freeswitch >>> variable for knowing whether the playback has done. >>> >>> My question is "why this freeswitch variable is not setting properly when >>> I play back more than 10 files using playback_delimiter option?". >>> >>> When I play back lesser than ten voice files the variable has been set >>> properly. What could be the reason? >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Thangappan.M >>> Date: Sat, Nov 21, 2009 at 2:52 PM >>> Subject: Problem while playing more than 10 voice files using playback >>> To: freeswitch-users >>> >>> >>> Dear all, >>> >>> I am in the process of implementing IVR using event outbound >>> socket (async mode). >>> I have implemented using Perl language. >>> >>> I did the following steps: >>> => Set the playback_delimiter variable >>> => Set the playback_sleep_val variable >>> => Set the event lock as true >>> => Set the freeswitch ( my own) variable as zero >>> => Wait in the loop until the variable is been set as >>> zero >>> => Playback the voice files ( Here I combined the >>> voice files with the delimiter value if more than one voice files are there) >>> => Set the freeswitch(my own) variable as true ( This >>> is used to identify whether the voice files are played >>> successfully). >>> => Wait in the loop until the variable is been set as >>> one. >>> => Set the Event lock as false >>> >>> => Trying to get the DTMF digits ( Have a assurance >>> that all the voice files are played). >>> >>> The problem is, >>> >>> The above steps are working fine when the voice file count >>> is lesser than or equal to 10. After the voice files are played only the >>> variable(my own freeswitch) is set. Based on the variable I am doing further >>> things. >>> >>> But when I tried to give the voice files count of more than >>> 10 the variable has been set while starting to play back the first voice >>> file itself . Because of this I am not able to proceed further. >>> >>> *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* >>> >>> *NOTE*: I also referred mod_file_string documentation. In that they >>> specified 128 files can be used to play back the voice files using >>> playback_delimiter option. >>> >>> Please help me................? >>> Thanks in advance. >>> >>> >>> -- >>> Regards, >>> Thangappan.M >>> >>> >>> >>> -- >>> Regards, >>> Thangappan.M >>> >> >> >> >> -- >> Regards, >> Thangappan.M >> > > > > -- > Regards, > Thangappan.M > -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/043e14f4/attachment.html From mike at jerris.com Tue Nov 24 22:34:13 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 01:34:13 -0500 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911242218l580b90eem3ec50676dfbc5536@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> <7aa29e790911242209w7ee2912bhbde4b3475147628d@mail.gmail.com> <7aa29e790911242218l580b90eem3ec50676dfbc5536@mail.gmail.com> Message-ID: <4C8DCE59-7E49-4147-B930-227945D3D243@jerris.com> "you should use execute_complete events to tell when a command you tried to execute has finished and not poll the channel for a variable to be set because FreeSWITCH is an asynchronous application in the mode you are describing and you can never be sure of the timing." You are STILL polling for the variable. If you want help, perhaps you should at least attempt what is being suggested? Mike On Nov 25, 2009, at 1:18 AM, Thangappan.M wrote: > The example script is there in the following link > http://pastebin.com/f332f2fda > > In the previous post I have attached it. But it was not shown. > > 2009/11/25 Thangappan.M > FreeSWITCH version: freeswitch 1.0.4 > I am using ESL library > I attached the example Perl script which does the same steps that I posted already. ( Sample.pl) > I supplied the log , Here I attached the output of the ESL log. (Output.txt) > > Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. > But in the output I got only 2,4,5,4 ( DTMF 1 is missed) > > Output of Perl code could be like > > Wait for response time out > EVENT [COMMAND] > Wait for response time out > EVENT [DTMF] > DTMF digit 2 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 5 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > Buffer: 2454 > BYE > > Why the first digit(1) is missed here? > In ESL log there is no digit called 1 why? > Why the COMMAND event is received instead of DTMF? > How can I get all DTMF digits? > > > > > > > > > > > > > > > > > On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M wrote: > The reason for waiting only for DTMF event is to handle the time outs in the IVR concept like response and inter digit time out. Using our own logic we 10 voice files in each play back if the voice files are more than 10. Now it works fine. > > Now the new problem has been raised. The problem is we are filtering only for DTMF events but we are getting COMMAND event . Because of this the DTMF digits are missing at the time . I am not able to proceed further. We are in the critical situation. > > Why this command event is occurring? > How can I restrict this? > What are the information it has? > How can I get all the information in it ? ( If command event has info) > > Help me............ > > > On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: > I am waiting only for DTMF events. That's why I am setting freeswitch variable for knowing whether the playback has done. > > My question is "why this freeswitch variable is not setting properly when I play back more than 10 files using playback_delimiter option?". > > When I play back lesser than ten voice files the variable has been set properly. What could be the reason? > > > > ---------- Forwarded message ---------- > From: Thangappan.M > Date: Sat, Nov 21, 2009 at 2:52 PM > Subject: Problem while playing more than 10 voice files using playback > To: freeswitch-users > > > Dear all, > > I am in the process of implementing IVR using event outbound socket (async mode). > I have implemented using Perl language. > > I did the following steps: > => Set the playback_delimiter variable > => Set the playback_sleep_val variable > => Set the event lock as true > => Set the freeswitch ( my own) variable as zero > => Wait in the loop until the variable is been set as zero > => Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) > => Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played > successfully). > => Wait in the loop until the variable is been set as one. > => Set the Event lock as false > > => Trying to get the DTMF digits ( Have a assurance that all the voice files are played). > > The problem is, > > The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. > > But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. > > DID I MAKE ANY MISTAKE IN THE ABOVE STEPS? > > NOTE: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. > > Please help me................? > Thanks in advance. > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/4aad26d9/attachment-0001.html From ovvenkatesan at gmail.com Tue Nov 24 22:36:20 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Wed, 25 Nov 2009 12:06:20 +0530 Subject: [Freeswitch-users] How to connect SIP phone to freeswitch Message-ID: <47d63d920911242236m2c4720a8g7c900fe5f02c05aa@mail.gmail.com> Hi . Could you please tell me, How to connect sip phone (which one is more friendly with freeswitch) to freeswitch. How I can check whether connection is properly established or not? -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/94fb5dc2/attachment.html From mike at jerris.com Tue Nov 24 22:49:43 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 01:49:43 -0500 Subject: [Freeswitch-users] How to connect SIP phone to freeswitch In-Reply-To: <47d63d920911242236m2c4720a8g7c900fe5f02c05aa@mail.gmail.com> References: <47d63d920911242236m2c4720a8g7c900fe5f02c05aa@mail.gmail.com> Message-ID: <7465AD63-4C50-4D9C-993B-81B6621F98F4@jerris.com> http://wiki.freeswitch.org/wiki/Getting_Started_Guide http://wiki.freeswitch.org/wiki/Interop_List On Nov 25, 2009, at 1:36 AM, ovvenkat wrote: > Hi . > > Could you please tell me, How to connect sip phone (which one is more friendly with freeswitch) to freeswitch. How I can check whether connection is properly established or not? > From mcampbellsmith at gmail.com Tue Nov 24 23:46:11 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 25 Nov 2009 18:46:11 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> Message-ID: <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> Hi there Itamar, Does the SPA3102 support TLS or only SRTP? And what about Brians comments that 'It uses a sick twisted method of doing SRTP'. Do you have it working using SRTP together with FS? What about TLS? Otherwise are there any other ATA's that support TLS & SRTP? On Sun, Nov 22, 2009 at 8:41 PM, Itamar Reis Peixoto wrote: > it's support SRTP > > > On Sun, Nov 22, 2009 at 7:21 AM, Mark Campbell-Smith > wrote: >> Do LInksys devices support TLS and SRTP that FS supports? ?3102 at >> least doesn't according to this post > > > > > > -- > ------------ > > Itamar Reis Peixoto > > e-mail/msn/google talk/sip: itamar at ispbrasil.com.br > skype: itamarjp > icq: 81053601 > +55 11 4063 5033 > +55 34 3221 8599 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Wed Nov 25 00:14:53 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 25 Nov 2009 19:14:53 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> Message-ID: <20091125081453.GA28340@jdc.jasonjgw.net> Mark Campbell-Smith wrote: > Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... From ovvenkatesan at gmail.com Wed Nov 25 00:29:28 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Wed, 25 Nov 2009 13:59:28 +0530 Subject: [Freeswitch-users] How Register soft sip phones to FreeSWITCH with extension number. Message-ID: <47d63d920911250029v61538c37veefc1bd44e1bd072@mail.gmail.com> Hi to All, Any one please tell me , How to configure soft sip phone to freeswitch with extension number. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/dc8631a4/attachment.html From mcampbellsmith at gmail.com Wed Nov 25 00:34:29 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 25 Nov 2009 19:34:29 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <20091125081453.GA28340@jdc.jasonjgw.net> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> Message-ID: <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> The only ATA mentioned on the WIKI that supports TLS/SRTP is the Grandstream HandyTone 503. But, again according to the wiki, that doesn't seem to behave to well with TLS ... On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: > Mark Campbell-Smith wrote: >> Does the SPA3102 support TLS or only SRTP? > > I don't know, but supporting only SRTP would be ridiculous, since the keys > would then be transmitted in the clear and therefore amenable to interception. > SRTP requires the SIP channel to be encrypted by TLS in order to be secure. > ZRTP, on the other hand, doesn't have this limitation: it works entirely in > RTP. > > I would be rather surprised were a hardware manufacturer to implement SRTP > without TLS for the SIP traffic. On the other hand, we've seen often in this > forum that some manufacturers are really clueless... > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mcampbellsmith at gmail.com Wed Nov 25 00:36:41 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 25 Nov 2009 19:36:41 +1100 Subject: [Freeswitch-users] How Register soft sip phones to FreeSWITCH with extension number. In-Reply-To: <47d63d920911250029v61538c37veefc1bd44e1bd072@mail.gmail.com> References: <47d63d920911250029v61538c37veefc1bd44e1bd072@mail.gmail.com> Message-ID: <33c87fa30911250036k4c0820d0pd26ef96d7971b024@mail.gmail.com> Didn't Michael already answer this? Best read the FS wiki and the softphone user guide for help with this. http://wiki.freeswitch.org/wiki/Getting_Started_Guide http://wiki.freeswitch.org/wiki/Interop_List On Wed, Nov 25, 2009 at 7:29 PM, ovvenkat wrote: > Hi to All, > > Any one please tell me , How to configure soft sip phone to freeswitch with > extension number. > > -- > > If you have come to help me, you are wasting your time. > If you have come to because your liberation is bound up in mine, we can work > together. > > > Regards > Venkatesan OV. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From stevendt at primrosebank.net Wed Nov 25 02:27:51 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 25 Nov 2009 10:27:51 -0000 Subject: [Freeswitch-users] Call Transfer Help Please References: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com><5A7C3038838142B5A07F181C193754F6@bp1.ad.bp.com> <1259121268571-4062810.post@n2.nabble.com> Message-ID: <19A54F6B1B3F419E8EB581FEA7F44725@bp1.ad.bp.com> Jeff, thanks very much for picking this up. You quickly spotted my mistake - I had the bind_meta_data call in the local extensions but not added it to the group extension (100). Appreciate you taking the time to have a look and point out my silly mistake - all working now, regards Dave ----- Original Message ----- From: "Jeff Lenk" To: Sent: Wednesday, November 25, 2009 3:54 AM Subject: Re: [Freeswitch-users] Call Transfer Help Please > > I do not see the meta app getting added in your log > -> > Dialplan: sofia/internal/1000 at 192.168.1.50 Action bind_meta_app(* > > Without this no meta actions will occur > > > > Dave Stevenson wrote: >> >> Hi again folks, >> >> I have posted a dump into the Pastebin (11276), could someone have a look >> and perhaps suggest where the problem might be please ? >> >> I'm sure you'll be able to work it out, but the log is for a call where >> :- >> >> incoming on PSTN Line (ext 1000) >> Group exts, 111, 1001, 1001 >> Answered on 111 and requested transfer to 1001 with no success >> >> regards >> Dave >> >> >> ----- Original Message ----- >> From: Dave Stevenson >> To: freeswitch-users at lists.freeswitch.org >> Sent: Tuesday, November 24, 2009 10:36 PM >> Subject: Re: [Freeswitch-users] Call Transfer Help Please >> >> >> Hi Mike, >> >> thanks for the reply. I am using the pre-compiled Windows binary - is >> there a 1.0.5 pre-release of that yet ? >> >> FreeSwitch reports its version as 1.0.4 (14460) but this is not >> correct, >> I was sure that I had previously loaded a later SVN Version, but just did >> it again, unless I'm not doing it right, the version number does not seem >> to be getting updated. The current build in the precompiled binaries area >> is reported to be 15604 and I've downloaded and installed that - although >> when the installer runs it tells me that it is version 15376. Either way, >> the "Version" command in FreeSwitch reports 1.0.4 (14460). >> >> The Transfer still does not work for me from the extension which >> answers >> the call. >> >> Sorry if my earlier questions were unclear ... >> "What are the correct LISTEN_TO and RESPOND_ON entries in >> dialplan.xml >> ?" >> What is the correct "transfer" data string in features.xml ? >> I don't understand this question(s) >> >> I was looking for clarification of the second two arguments in the >> bind_meta_app data call, i.e, that the "b" and "s" were the correct >> values >> and also that the "is transfer" "transfer" data argument was "-bleg" >> >> That is, that the arguments in the default dialplan are correct for >> this >> scenario - which they appear to be based on your previous reply to my >> query. >> >> So, is there anything else that I can check to see why this is not >> working ? >> >> >> regards >> Dave >> >> >> >> ----- Original Message ----- >> From: Michael Jerris >> To: freeswitch-users at lists.freeswitch.org >> Sent: Tuesday, November 24, 2009 8:19 PM >> Subject: Re: [Freeswitch-users] Call Transfer Help Please >> >> >> >> >> On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: >> >> >> Hi, >> >> I'm trying to setup call transfer for a phone without a transfer >> button. I was on IRC last night and got some pointers to how this is >> setup >> in dialplan.xml and features.xml and what "bind meta app" does. >> >> Once it became clear how the transfer is initiated and that the >> transfer, in the default config, can only be initiated by the "b" leg of >> the call, I was able to make this work as configured in the defaults, >> i.e, >> to initiate a transfer (for an internal call) from the dialled extension >> to a new extension. >> >> Now the problem . . . >> >> I have an incoming PSTN line that rings a group of extensions, what >> I want to be able to do is to give whoever answers the PSTN call ability >> to transfer the call on to another extension. >> >> There is an ATA (Linksys SPA3101) set up on the PSTN line with a >> FreeSwitch extension of 1000, it rings the extension phones in the group. >> >> I'd hoped that the default transfer setup would handle this without >> modification - the incoming call on extension 1000 would be the "a" leg, >> the answering extension would be the "b" leg and a transfer from "b" >> would >> work as per the default config. This does not work for me though. >> >> I'm struggling a bit with the "bind meta app" options and can't >> seem >> to make it do what I want. >> >> Could someone please confirm that what I'm trying to do is feasible >> and perhaps suggest the right parameters to use in dialplan.xml and >> features.xml please ? >> >> Relevant section in the "is_transfer" section in features.xml >> >> >> And in default.xml from >> to >> >> >> I've tried posting a call log to the Pastebin (11252/3) but there >> was an error - it looks like the dump was too big. Not sure what the >> maximum size on pastebin dumps is ? >> >> >> My understanding (or lack of) of "a" and "b" are in the scenario >> described is not helping ... >> >> Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? >> >> >> Yes, the calling leg >> >> >> Is the answering extension the "b" leg ? >> >> >> Yes >> >> >> What are the correct LISTEN_TO and RESPOND_ON entries in >> dialplan.xml ? >> >> >> I don't understand this question >> >> >> What is the correct "transfer" data string in features.xml ? >> >> >> >> ditto >> >> >> Or am I totally on the wrong track here ? >> >> >> >> You should just need to make sure that the bind meta is called in >> this >> scenario so the b leg is able to do it, thats it. >> >> >> If it is possible to do what I want, and changes are required to >> the >> dialplan.xml and/or features.xml files, is it possible to have different >> logic in there such that the actions are different whether it is the "a" >> leg or "b" leg that's requesting the transfer ? >> >> regards >> Dave >> >> FreeSwitch Version 1.0.4 (14460) >> >> >> also, try the latest 1.0.5. pre release or svn trunk to confirm this >> is not an issue that has already been fixed. >> >> >> Mike >> >> >> >> >> ---------------------------------------------------------------------------- >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> ------------------------------------------------------------------------------ >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Call-Transfer-Help-Please-tp4056930p4062810.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mattdfong at gmail.com Wed Nov 25 06:53:55 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 25 Nov 2009 06:53:55 -0800 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> Message-ID: <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> I tried removing the codec file extension from uuid_record and session_record but I'm still unable to record a file in native format for a bridged call. record WORKS!, but uuid_record and session_record do not want to record in native format. do uuid_record and session_record work with native format? or is it not going to be possible to record a bridged call in native format?...maybe because there are two different channels with a bridged call? If it isn't going to be possible, what's the best format to record bridged calls in that conserves the most processing power? .wav? Thanks. --matt DEBUG logs from console: http://pastebin.freeswitch.org/11283 Lua script: api = freeswitch.API(); --record = api:execute("sched_api", '+1 none uuid_record '..session:getVariable("uuid")..' start /tmp/my_recording'); --session:execute("record", "/tmp/my_recording"); session:execute("record_session", "/tmp/my_recording"); session:execute("playback", "somefile.wav"); On Mon, Nov 23, 2009 at 6:42 AM, Brian West wrote: > If you're doing native file you DO NOT put an extension on the file > name. > > /b > > On Nov 22, 2009, at 5:54 PM, Seven Du wrote: > > > did you try without any .wav or .PCMU? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/48bc296e/attachment-0001.html From kokoska.rokoska at post.cz Wed Nov 25 07:11:05 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 25 Nov 2009 16:11:05 +0100 Subject: [Freeswitch-users] how to enable short recordings Message-ID: <4B0D4909.7030009@post.cz> Hello all, is there a way how to enable very short recordings (1-3 seconds) in FreeSWITCH other than editing source code and recompiling? Thanks for your time! Best regards, kokoska.rokoska From brian at freeswitch.org Wed Nov 25 07:18:15 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 09:18:15 -0600 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> Message-ID: <133568C3-6EE1-40E2-8C0F-2CB174C2D94D@freeswitch.org> These two options attach media bugs on to the session. Which doesn't work with native files as far as I know. /b On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote: > record WORKS!, but uuid_record and session_record do not want to > record in native format. do uuid_record and session_record work with > native format? or is it not going to be possible to record a bridged > call in native format?...maybe because there are two different > channels with a bridged call? From brian at freeswitch.org Wed Nov 25 07:18:42 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 09:18:42 -0600 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <4B0D4909.7030009@post.cz> References: <4B0D4909.7030009@post.cz> Message-ID: <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> Is this standard recording? or voicemail? /b On Nov 25, 2009, at 9:11 AM, kokoska rokoska wrote: > Hello all, > > is there a way how to enable very short recordings (1-3 seconds) in > FreeSWITCH other than editing source code and recompiling? > > Thanks for your time! > > Best regards, > > kokoska.rokoska From jeff at jefflenk.com Wed Nov 25 07:32:17 2009 From: jeff at jefflenk.com (Jeff Lenk ) Date: Wed, 25 Nov 2009 15:32:17 +0000 Subject: [Freeswitch-users] how to enable short recordings Message-ID: Is this for vm? If so set min-record-len on the profile -----Original Message----- From: kokoska rokoska Sent: 11/25/2009 3:11:05 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] how to enable short recordings Hello all, is there a way how to enable very short recordings (1-3 seconds) in FreeSWITCH other than editing source code and recompiling? Thanks for your time! Best regards, kokoska.rokoska _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/1f1b152d/attachment.html From imthiyazg at gmail.com Wed Nov 25 07:42:01 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Wed, 25 Nov 2009 21:12:01 +0530 Subject: [Freeswitch-users] passive recording Message-ID: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> hi is it possibe to enable passive recording in sangoma tdm interface in feeswich. pls advice Best Regards G.Imthiyaz Ahmed From mattdfong at gmail.com Wed Nov 25 07:44:41 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 25 Nov 2009 07:44:41 -0800 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <133568C3-6EE1-40E2-8C0F-2CB174C2D94D@freeswitch.org> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> <133568C3-6EE1-40E2-8C0F-2CB174C2D94D@freeswitch.org> Message-ID: <4256bf830911250744i3453f961h1e2c35f65222cab8@mail.gmail.com> so is using session_record with .wav my best option for recording bridged calls? --matt On Wed, Nov 25, 2009 at 7:18 AM, Brian West wrote: > These two options attach media bugs on to the session. Which doesn't > work with native files as far as I know. > > /b > > On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote: > > > record WORKS!, but uuid_record and session_record do not want to > > record in native format. do uuid_record and session_record work with > > native format? or is it not going to be possible to record a bridged > > call in native format?...maybe because there are two different > > channels with a bridged call? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/e2bde16f/attachment.html From kokoska.rokoska at post.cz Wed Nov 25 07:46:37 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 25 Nov 2009 16:46:37 +0100 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> Message-ID: <4B0D515D.60805@post.cz> Thank you very much, Brian, for your interest! It is standard recording: Best regards, kokoska.rokoska Brian West napsal(a): > Is this standard recording? or voicemail? > > /b > > On Nov 25, 2009, at 9:11 AM, kokoska rokoska wrote: > >> Hello all, >> >> is there a way how to enable very short recordings (1-3 seconds) in >> FreeSWITCH other than editing source code and recompiling? >> >> Thanks for your time! >> >> Best regards, >> >> kokoska.rokoska > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Nov 25 07:51:56 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 09:51:56 -0600 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <4B0D515D.60805@post.cz> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> Message-ID: Really you want to keep 1-3 second files around? /b On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > Thank you very much, Brian, for your interest! > > It is standard recording: > > > > Best regards, > > kokoska.rokoska From rupa at rupa.com Wed Nov 25 07:56:14 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 25 Nov 2009 09:56:14 -0600 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <4256bf830911250744i3453f961h1e2c35f65222cab8@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> <133568C3-6EE1-40E2-8C0F-2CB174C2D94D@freeswitch.org> <4256bf830911250744i3453f961h1e2c35f65222cab8@mail.gmail.com> Message-ID: Yes On Wed, Nov 25, 2009 at 9:44 AM, Matthew Fong wrote: > so is using session_record with .wav my best option for recording bridged > calls? > > --matt > > > On Wed, Nov 25, 2009 at 7:18 AM, Brian West wrote: > >> These two options attach media bugs on to the session. Which doesn't >> work with native files as far as I know. >> >> /b >> >> On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote: >> >> > record WORKS!, but uuid_record and session_record do not want to >> > record in native format. do uuid_record and session_record work with >> > native format? or is it not going to be possible to record a bridged >> > call in native format?...maybe because there are two different >> > channels with a bridged call? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/a9948b61/attachment.html From kokoska.rokoska at post.cz Wed Nov 25 08:03:45 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 25 Nov 2009 17:03:45 +0100 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> Message-ID: <4B0D5561.4020009@post.cz> Yes, Brian, I need them :-) They don't contain speech - instead, they contain few "computer generated" tones and I should store them in max quality for later proccessing (i.e. analysis)... Best regards, kokoska.rokoska Brian West napsal(a): > Really you want to keep 1-3 second files around? > > /b > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > >> Thank you very much, Brian, for your interest! >> >> It is standard recording: >> >> >> >> Best regards, >> >> kokoska.rokoska > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From juanbackson at gmail.com Wed Nov 25 08:10:30 2009 From: juanbackson at gmail.com (Juan Backson) Date: Thu, 26 Nov 2009 00:10:30 +0800 Subject: [Freeswitch-users] modify SDP for 200 OK Message-ID: <27c25bc40911250810x783fb1cbg49f50e624353bd51@mail.gmail.com> Hi, If I am using proxy_media=true, bypass_media=false, is there anyway of modifying o= and c= so that it won't show the IP of the far-end B leg? I am using fs as b2b2a and I want to hide the far-end ip as much as possible. I got to hide the IP for invite by modifying the sdp within C code, but I don't know how to do that for 200 OK. Any idea? Thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/76a73c27/attachment.html From juliano.duque at terra.com.br Wed Nov 25 05:18:07 2009 From: juliano.duque at terra.com.br (Juliano - Terra) Date: Wed, 25 Nov 2009 11:18:07 -0200 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B0C6499.4060504@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> Message-ID: <4B0D2E8F.1030200@terra.com.br> Peter, I had a similar problem, the way I found to make it work was setting the mailbox ID in the phone to match the FS domain/hostname. For instance using a Linksys SPA962, I set the "Voice Mail Server" in the extension tab to extension at domain using FS hostname as the domain. Regards, Juliano Peter P GMX escreveu: > Anthony, thanks for the hint, > > I receive events like the following > RECV EVENT > Event-Name: MESSAGE_WAITING > Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 > FreeSWITCH-Hostname: sip11.mydomain.com > FreeSWITCH-IPv4: 192.168.178.200 > FreeSWITCH-IPv6: ::1 > Event-Date-Local: 2009-11-24 23:33:13 > Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT > Event-Date-Timestamp: 1259101993918617 > Event-Calling-File: mod_voicemail.c > Event-Calling-Function: update_mwi > Event-Calling-Line-Number: 1738 > MWI-Messages-Waiting: yes > MWI-Message-Account: 200 at sip1.mydomain.com > MWI-Voice-Message: 4/1 (0/0) > > I think the problem may be the Freeswitch cluster we are working with. > All phones register with realm (e.g. 200 at sip1.mydomain.com). The FS > hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. > With xml_curl we ensure that for both domain names a directory entry is > passed back. That way it works nicely with registering phones, receiving > voicemails, recording voicemails etc. but not for MWI. For recording and > querying voicemails we use the realm instead of the domain name and that > way it works. > > When a voicemail has finished recording - and at the time the above > message occurs - I see 2 directory xml_curl requests with > Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id > One I expect is for retrieving the MWI data and the other one for > sending the VM email (which is sucessfully sent). > > Any hint how we can workaround this? Or is there a parameter to tell > mod_voicemail that is should use the realm instead of the local hostname > for sending MWI? > > Best regards > Peter > > Anthony Minessale schrieb: > >> connect to FS with fs_cli >> >> Issue the command: >> >> /events MESSAGE_QUERY MESSAGE_WAITING >> >> then leave some voice mails >> >> probably you have a mis-configuration where the user/domain/profile >> cannot be resolved to the correct >> sofia profile to send the notify >> >> The event starts out as a freeswitch event and is translated into the >> notify by mod_sofia but only if it can >> match the event to a real sip user >> >> >> >> >> On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX > > wrote: >> >> Hello, >> >> I have a similar problem with Freeswitch behind OpenSIPS as a load >> balancer: >> When registering, Freeeswitch does not send a MWI NOTIFY message for a >> Phone which has voicemails. Even after recording a new voicemail there >> is no NOTIFY message sent. And there are no error messages on the >> console. >> >> I have explicitely set >> in the internal >> profile. >> >> When a phone is set up I get the following >> Snom Phone REGISTER => OpenSIPS=> Freeswitch >> Freeswitch OK => OpenSIPS=>Snom Phone >> >> Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch >> Freeswitch 202 Accepted => OpenSIPS=>Snom Phone >> >> Snom Phone PUBLISH => OpenSIPS=> Freeswitch >> Freeswitch 200 OK => OpenSIPS=>Snom Phone >> So presence generally seems to work. >> >> But ngrepping the Network traffic there's no MWI NOTIFY message coming >> from Freeswitch to any phone. >> FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before >> should be already there. >> >> Any idea how to force the NOTIFY messages? >> >> >> Best regards >> Peter >> >> Here's the debug Level9 output for nta and nua when a phone with VMs >> registers, seems like there is no error in it: >> >> freeswitch at sip11.mydomain.com >> > nta: received REGISTER >> sip:sip1.mydomain.com SIP/2.0 (CSeq 7) >> nta: REGISTER (7) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x7fd5d409c8f0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x7fd5d409c8f0): sent signal r_respond >> nua: nua_handle_destroy: entering >> nua(0x7fd5d409c8f0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized >> nua: nua_stack_set_params: entering >> nta: sent 401 Unauthorized for REGISTER (7) >> nta: timer set to 32000 ms >> nua(0x7fd5d409c8f0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 6) >> nta: REGISTER (6) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x905a80): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x905a80): sent signal r_respond >> nua: nua_handle_destroy: entering >> nua(0x905a80): recv signal r_respond 401 Unauthorized >> nua(0x905a80): sent signal r_destroy >> nua: nua_stack_set_params: entering >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 401 Unauthorized for REGISTER (6) >> nua(0x905a80): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received PUBLISH sip:100 at sip1.mydomain.com >> SIP/2.0 (CSeq 3) >> nta: PUBLISH (3) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x905f10): event i_publish 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x905f10): sent signal r_respond >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nua(0x905f10): recv signal r_respond 200 OK >> nua: nua_stack_set_params: entering >> nua(0x905f10): sent signal r_destroy >> nta: sent 200 OK for PUBLISH (3) >> nua(0x905f10): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received SUBSCRIBE sip:mod_sofia at 192.168.178.200:5062 >> SIP/2.0 (CSeq 2) >> nta: canonizing sip:mod_sofia at 192.168.178.200:5062 >> with contact >> nta: SUBSCRIBE (2) going to existing leg >> nua: nua_stack_process_request: entering >> nta: sent 200 OK for SUBSCRIBE (2) >> nua(0x905560): event i_subscribe 200 OK >> nua: nua_application_event: entering >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 8) >> nta: REGISTER (8) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x7fd5dc073ba0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x7fd5dc073ba0): sent signal r_respond >> nua(0x7fd5dc073ba0): recv signal r_respond 200 OK >> nua: nua_stack_set_params: entering >> nua: nua_handle_destroy: entering >> nua(0x7fd5dc073ba0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 200 OK for REGISTER (8) >> nua(0x7fd5dc073ba0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 7) >> nta: REGISTER (7) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x8fc3d0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x8fc3d0): sent signal r_respond >> nua(0x8fc3d0): recv signal r_respond 200 OK >> nua: nua_handle_destroy: entering >> nua: nua_stack_set_params: entering >> nua(0x8fc3d0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 200 OK for REGISTER (7) >> nua(0x8fc3d0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received SUBSCRIBE sip:100 at sip1.mydomain.com >> ;user=phone SIP/2.0 >> (CSeq 1) >> nta: SUBSCRIBE (1) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nta_leg_tcreate(0x7fd5dc03add0) >> nua(0x7fd5dc078b70): adding notify usage with event message-summary >> nua(0x7fd5dc078b70): event i_subscribe 100 Trying >> nua: nua_application_event: entering >> nua(): refresh notify after 3600 seconds (in [3600..3600]) >> nua: nua_respond: entering >> nua(0x7fd5dc078b70): sent signal r_respond >> nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted >> nua: nua_stack_set_params: entering >> nta: sent 202 Accepted for SUBSCRIBE (1) >> >> >> >> >> >> mayamatakeshi schrieb: >> > >> > On 9/12/09, *mayamatakeshi* > >> > > >> wrote: >> > >> > >> > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris >> >> > >> wrote: >> > >> > Following up, did a bug get created for this issue? >> > >> > >> > Hello, >> > yes. >> > http://jira.freeswitch.org/browse/MODSOFIA-26 >> > >> > >> > Just to simplify things in case someone searches the list: >> > Issue was solved on rev 14851. >> > Thank you all. >> > >> ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From brian at freeswitch.org Wed Nov 25 08:20:29 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 10:20:29 -0600 Subject: [Freeswitch-users] modify SDP for 200 OK In-Reply-To: <27c25bc40911250810x783fb1cbg49f50e624353bd51@mail.gmail.com> References: <27c25bc40911250810x783fb1cbg49f50e624353bd51@mail.gmail.com> Message-ID: You know FreeSWITCH will proxy media already if you turn off proxy_media and disable transcoding you'll get the same results and the IP's will be correct. Proxy media is for one purpose... T.38, it gains you NOTHING otherwise. /b On Nov 25, 2009, at 10:10 AM, Juan Backson wrote: > Hi, > > If I am using proxy_media=true, bypass_media=false, is there anyway > of modifying o= and c= so that it won't show the IP of the far-end B > leg? > > I am using fs as b2b2a and I want to hide the far-end ip as much as > possible. > > I got to hide the IP for invite by modifying the sdp within C code, > but I don't know how to do that for 200 OK. Any idea? > > Thanks, > jb From srinivas.ksvreddy at gmail.com Wed Nov 25 08:53:34 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Wed, 25 Nov 2009 22:23:34 +0530 Subject: [Freeswitch-users] Bypass_media and re_invite Message-ID: Hi All, goodmorning to all, i have a scenario, two pjsua clients are connected with Freeswitch and they are in call and bypass_media=true. i close the Freeswitch server, still they are in call, again i started the Freeswitch, and registerd these two endpoints, now how can i end the call(estabilished by the first Freeswitch)? if i call re_invite will it estabilish the call between two endpoints? any idea? Thanks Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/71317773/attachment.html From imthiyazg at gmail.com Wed Nov 25 09:00:43 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Wed, 25 Nov 2009 22:30:43 +0530 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> Message-ID: <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> hi is it possibe to enable passive recording in sangoma tdm interface in feeswich. pls advice Best Regards G.Imthiyaz Ahmed -- Best Regards G.Imthiyaz Ahmed PeopleTech systems (P) ltd http://peopletech.co.in From anthony.minessale at gmail.com Wed Nov 25 09:07:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 11:07:04 -0600 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <4B0D5561.4020009@post.cz> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> <4B0D5561.4020009@post.cz> Message-ID: <191c3a030911250907x38aca94cs9128629fc2a1ba7c@mail.gmail.com> use the variable RECORD_MIN_SEC This was added in revision 15271 so if you are below that I recommend updating to latest trunk. On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska wrote: > Yes, Brian, I need them :-) > > They don't contain speech - instead, they contain few "computer > generated" tones and I should store them in max quality for later > proccessing (i.e. analysis)... > > Best regards, > > kokoska.rokoska > > > Brian West napsal(a): > > Really you want to keep 1-3 second files around? > > > > /b > > > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > > > >> Thank you very much, Brian, for your interest! > >> > >> It is standard recording: > >> > >> > >> > >> Best regards, > >> > >> kokoska.rokoska > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/d72acaf6/attachment-0001.html From anthony.minessale at gmail.com Wed Nov 25 09:13:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 11:13:39 -0600 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> Message-ID: <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> What do you mean by passive encoding? On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed wrote: > hi > is it possibe to enable passive recording in sangoma tdm interface > in feeswich. pls advice > Best Regards > G.Imthiyaz Ahmed > > > > -- > Best Regards > G.Imthiyaz Ahmed > PeopleTech systems (P) ltd > http://peopletech.co.in > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/1725495a/attachment.html From anthony.minessale at gmail.com Wed Nov 25 09:13:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 11:13:47 -0600 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> Message-ID: <191c3a030911250913i41ec7571t17396b9af247eb2f@mail.gmail.com> What do you mean by passive encoding? On Wed, Nov 25, 2009 at 11:13 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > What do you mean by passive encoding? > > > On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed wrote: > >> hi >> is it possibe to enable passive recording in sangoma tdm interface >> in feeswich. pls advice >> Best Regards >> G.Imthiyaz Ahmed >> >> >> >> -- >> Best Regards >> G.Imthiyaz Ahmed >> PeopleTech systems (P) ltd >> http://peopletech.co.in >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/39e55637/attachment.html From mshepet at gmail.com Mon Nov 23 09:09:33 2009 From: mshepet at gmail.com (Michael Shepet) Date: Mon, 23 Nov 2009 12:09:33 -0500 Subject: [Freeswitch-users] Server give-away Message-ID: We at Swifcore Technologies, a telephony and server management team, would like your help in reviewing our latest product. We have created a hosting platform around the FreeSWITCH engine (for obvious reasons of stability and extensibility) and would like your feedback so we continue to improve our service. To facilitate this, we are giving away ten (10) hosting packages (pre-configured with latest FreeSWITCH compiled trunk) along with a web diagnostic dashboard for a free 60 day trial, no strings attached (or credit card required)! Each private server comes with CentOS 5.4 64-bit, 512MB RAM, its own public IP (no NAT problems), a shared test DID number, 100 minutes/month call credit, and ssh access. All you have to do is tell us why you love FreeSWITCH and something creative you have done or plan on doing with it. The 10 best responses will be awarded servers. If you are interested, email your response to giveaway at swifcore.com. You can also go to http://www.swifcore.com/products/myswitch for more product details. Thank you! Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/1b98a0b3/attachment.html From kokoska.rokoska at post.cz Wed Nov 25 09:20:06 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 25 Nov 2009 18:20:06 +0100 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <191c3a030911250907x38aca94cs9128629fc2a1ba7c@mail.gmail.com> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> <4B0D5561.4020009@post.cz> <191c3a030911250907x38aca94cs9128629fc2a1ba7c@mail.gmail.com> Message-ID: <4B0D6746.8060907@post.cz> Thank you very much, Anthony, for your help! I'm nearly at current trunk (15653) and works great :-) Many thanks once more! Best regards, kokoska.rokoska Anthony Minessale napsal(a): > use the variable RECORD_MIN_SEC > > This was added in revision 15271 so if you are below that I recommend > updating to latest trunk. > > > On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska > > wrote: > > Yes, Brian, I need them :-) > > They don't contain speech - instead, they contain few "computer > generated" tones and I should store them in max quality for later > proccessing (i.e. analysis)... > > Best regards, > > kokoska.rokoska > > > Brian West napsal(a): > > Really you want to keep 1-3 second files around? > > > > /b > > > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > > > >> Thank you very much, Brian, for your interest! > >> > >> It is standard recording: > >> > >> > >> > >> Best regards, > >> > >> kokoska.rokoska > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From imthiyazg at gmail.com Wed Nov 25 09:29:03 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Wed, 25 Nov 2009 22:59:03 +0530 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> Message-ID: <8595daf70911250929w26eeb3aboae0f95042f35393b@mail.gmail.com> I mean to tap tx and rx of a PRI line using sangoma tap and record the call information and actual calls without distrubing the existing line . freeswitch will work in passive mode like trunk side call recorder. Thanks Imthiyaz On Wed, Nov 25, 2009 at 10:43 PM, Anthony Minessale wrote: > What do you mean by passive encoding? > > On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed > wrote: >> >> hi >> ?is it possibe to enable passive recording in sangoma tdm interface >> in feeswich. pls advice >> Best Regards >> G.Imthiyaz Ahmed >> >> >> >> -- >> Best Regards >> G.Imthiyaz Ahmed >> PeopleTech systems (P) ltd >> http://peopletech.co.in >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Best Regards G.Imthiyaz Ahmed PeopleTech systems (P) ltd http://peopletech.co.in From dujinfang at gmail.com Wed Nov 25 09:32:51 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 26 Nov 2009 01:32:51 +0800 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> <133568C3-6EE1-40E2-8C0F-2CB174C2D94D@freeswitch.org> <4256bf830911250744i3453f961h1e2c35f65222cab8@mail.gmail.com> Message-ID: <23f91030911250932m65d22333sd299702b881c1891@mail.gmail.com> http://code.google.com/p/mod-recpld/ It's out-dated. I originally wrote it to record raw G.729 codec on passthrough mode. It worked before and then we abandoned that since We felt G729 cannot deliver good sound particularly on a cross-continent network. The code is written when I don't know much about FS internals, perhaps it's easier to write some mod with indicate no-transcoding in switch_rtp.c. 2009/11/25 Rupa Schomaker > Yes > > > On Wed, Nov 25, 2009 at 9:44 AM, Matthew Fong wrote: > >> so is using session_record with .wav my best option for recording bridged >> calls? >> >> --matt >> >> >> On Wed, Nov 25, 2009 at 7:18 AM, Brian West wrote: >> >>> These two options attach media bugs on to the session. Which doesn't >>> work with native files as far as I know. >>> >>> /b >>> >>> On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote: >>> >>> > record WORKS!, but uuid_record and session_record do not want to >>> > record in native format. do uuid_record and session_record work with >>> > native format? or is it not going to be possible to record a bridged >>> > call in native format?...maybe because there are two different >>> > channels with a bridged call? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/29ce5598/attachment-0001.html From dujinfang at gmail.com Wed Nov 25 09:36:20 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 26 Nov 2009 01:36:20 +0800 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <4B0D6746.8060907@post.cz> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> <4B0D5561.4020009@post.cz> <191c3a030911250907x38aca94cs9128629fc2a1ba7c@mail.gmail.com> <4B0D6746.8060907@post.cz> Message-ID: <23f91030911250936q65dc7c25oe8c3746d6caed9e7@mail.gmail.com> And you may also would like to update the wiki as well if the var is not there. 2009/11/26 kokoska rokoska > Thank you very much, Anthony, for your help! > > I'm nearly at current trunk (15653) and > > works great :-) > > Many thanks once more! > > Best regards, > > kokoska.rokoska > > > Anthony Minessale napsal(a): > > use the variable RECORD_MIN_SEC > > > > This was added in revision 15271 so if you are below that I recommend > > updating to latest trunk. > > > > > > On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska > > > wrote: > > > > Yes, Brian, I need them :-) > > > > They don't contain speech - instead, they contain few "computer > > generated" tones and I should store them in max quality for later > > proccessing (i.e. analysis)... > > > > Best regards, > > > > kokoska.rokoska > > > > > > Brian West napsal(a): > > > Really you want to keep 1-3 second files around? > > > > > > /b > > > > > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > > > > > >> Thank you very much, Brian, for your interest! > > >> > > >> It is standard recording: > > >> > > >> > > >> > > >> Best regards, > > >> > > >> kokoska.rokoska > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/72720bd3/attachment.html From mike at jerris.com Wed Nov 25 09:39:10 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:39:10 -0500 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B0C6499.4060504@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> Message-ID: <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> Try an alias on the sip profile. Mike On Nov 24, 2009, at 5:56 PM, Peter P GMX wrote: > Anthony, thanks for the hint, > > I receive events like the following > RECV EVENT > Event-Name: MESSAGE_WAITING > Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 > FreeSWITCH-Hostname: sip11.mydomain.com > FreeSWITCH-IPv4: 192.168.178.200 > FreeSWITCH-IPv6: ::1 > Event-Date-Local: 2009-11-24 23:33:13 > Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT > Event-Date-Timestamp: 1259101993918617 > Event-Calling-File: mod_voicemail.c > Event-Calling-Function: update_mwi > Event-Calling-Line-Number: 1738 > MWI-Messages-Waiting: yes > MWI-Message-Account: 200 at sip1.mydomain.com > MWI-Voice-Message: 4/1 (0/0) > > I think the problem may be the Freeswitch cluster we are working with. > All phones register with realm (e.g. 200 at sip1.mydomain.com). The FS > hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. > With xml_curl we ensure that for both domain names a directory entry is > passed back. That way it works nicely with registering phones, receiving > voicemails, recording voicemails etc. but not for MWI. For recording and > querying voicemails we use the realm instead of the domain name and that > way it works. > > When a voicemail has finished recording - and at the time the above > message occurs - I see 2 directory xml_curl requests with > Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id > One I expect is for retrieving the MWI data and the other one for > sending the VM email (which is sucessfully sent). > > Any hint how we can workaround this? Or is there a parameter to tell > mod_voicemail that is should use the realm instead of the local hostname > for sending MWI? > > Best regards > Peter > > Anthony Minessale schrieb: >> connect to FS with fs_cli >> >> Issue the command: >> >> /events MESSAGE_QUERY MESSAGE_WAITING >> >> then leave some voice mails >> >> probably you have a mis-configuration where the user/domain/profile >> cannot be resolved to the correct >> sofia profile to send the notify >> >> The event starts out as a freeswitch event and is translated into the >> notify by mod_sofia but only if it can >> match the event to a real sip user >> >> >> >> >> On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX > > wrote: >> >> Hello, >> >> I have a similar problem with Freeswitch behind OpenSIPS as a load >> balancer: >> When registering, Freeeswitch does not send a MWI NOTIFY message for a >> Phone which has voicemails. Even after recording a new voicemail there >> is no NOTIFY message sent. And there are no error messages on the >> console. >> >> I have explicitely set >> in the internal >> profile. >> >> When a phone is set up I get the following >> Snom Phone REGISTER => OpenSIPS=> Freeswitch >> Freeswitch OK => OpenSIPS=>Snom Phone >> >> Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch >> Freeswitch 202 Accepted => OpenSIPS=>Snom Phone >> >> Snom Phone PUBLISH => OpenSIPS=> Freeswitch >> Freeswitch 200 OK => OpenSIPS=>Snom Phone >> So presence generally seems to work. >> >> But ngrepping the Network traffic there's no MWI NOTIFY message coming >> from Freeswitch to any phone. >> FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before >> should be already there. >> >> Any idea how to force the NOTIFY messages? >> >> >> Best regards >> Peter >> >> Here's the debug Level9 output for nta and nua when a phone with VMs >> registers, seems like there is no error in it: >> >> freeswitch at sip11.mydomain.com >> > nta: received REGISTER >> sip:sip1.mydomain.com SIP/2.0 (CSeq 7) >> nta: REGISTER (7) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x7fd5d409c8f0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x7fd5d409c8f0): sent signal r_respond >> nua: nua_handle_destroy: entering >> nua(0x7fd5d409c8f0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized >> nua: nua_stack_set_params: entering >> nta: sent 401 Unauthorized for REGISTER (7) >> nta: timer set to 32000 ms >> nua(0x7fd5d409c8f0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 6) >> nta: REGISTER (6) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x905a80): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x905a80): sent signal r_respond >> nua: nua_handle_destroy: entering >> nua(0x905a80): recv signal r_respond 401 Unauthorized >> nua(0x905a80): sent signal r_destroy >> nua: nua_stack_set_params: entering >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 401 Unauthorized for REGISTER (6) >> nua(0x905a80): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received PUBLISH sip:100 at sip1.mydomain.com >> SIP/2.0 (CSeq 3) >> nta: PUBLISH (3) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x905f10): event i_publish 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x905f10): sent signal r_respond >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nua(0x905f10): recv signal r_respond 200 OK >> nua: nua_stack_set_params: entering >> nua(0x905f10): sent signal r_destroy >> nta: sent 200 OK for PUBLISH (3) >> nua(0x905f10): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received SUBSCRIBE sip:mod_sofia at 192.168.178.200:5062 >> SIP/2.0 (CSeq 2) >> nta: canonizing sip:mod_sofia at 192.168.178.200:5062 >> with contact >> nta: SUBSCRIBE (2) going to existing leg >> nua: nua_stack_process_request: entering >> nta: sent 200 OK for SUBSCRIBE (2) >> nua(0x905560): event i_subscribe 200 OK >> nua: nua_application_event: entering >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 8) >> nta: REGISTER (8) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x7fd5dc073ba0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x7fd5dc073ba0): sent signal r_respond >> nua(0x7fd5dc073ba0): recv signal r_respond 200 OK >> nua: nua_stack_set_params: entering >> nua: nua_handle_destroy: entering >> nua(0x7fd5dc073ba0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 200 OK for REGISTER (8) >> nua(0x7fd5dc073ba0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 7) >> nta: REGISTER (7) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x8fc3d0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x8fc3d0): sent signal r_respond >> nua(0x8fc3d0): recv signal r_respond 200 OK >> nua: nua_handle_destroy: entering >> nua: nua_stack_set_params: entering >> nua(0x8fc3d0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 200 OK for REGISTER (7) >> nua(0x8fc3d0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received SUBSCRIBE sip:100 at sip1.mydomain.com >> ;user=phone SIP/2.0 >> (CSeq 1) >> nta: SUBSCRIBE (1) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nta_leg_tcreate(0x7fd5dc03add0) >> nua(0x7fd5dc078b70): adding notify usage with event message-summary >> nua(0x7fd5dc078b70): event i_subscribe 100 Trying >> nua: nua_application_event: entering >> nua(): refresh notify after 3600 seconds (in [3600..3600]) >> nua: nua_respond: entering >> nua(0x7fd5dc078b70): sent signal r_respond >> nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted >> nua: nua_stack_set_params: entering >> nta: sent 202 Accepted for SUBSCRIBE (1) >> >> >> >> >> >> mayamatakeshi schrieb: >>> >>> On 9/12/09, *mayamatakeshi* > >>> > >> wrote: >>> >>> >>> On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris >> >>> >> wrote: >>> >>> Following up, did a bug get created for this issue? >>> >>> >>> Hello, >>> yes. >>> http://jira.freeswitch.org/browse/MODSOFIA-26 >>> >>> >>> Just to simplify things in case someone searches the list: >>> Issue was solved on rev 14851. >>> Thank you all. >>> >> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Wed Nov 25 09:44:05 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:44:05 -0500 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: FreeSWITCH will kill the calls when you shut it down, if you intentionally kill the network without shutting down FreeSWITCH the only thing you can do is enable session timers or rtp timers in the soft phones to kill the call when FreeSWITCH dies or when the call is over. Mike On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > Hi All, > > goodmorning to all, i have a scenario, two pjsua clients are connected with Freeswitch and they are in call and bypass_media=true. i close the Freeswitch server, still they are in call, again i started the Freeswitch, and registerd these two endpoints, now how can i end the call(estabilished by the first Freeswitch)? if i call re_invite will it estabilish the call between two endpoints? > any idea? From mike at jerris.com Wed Nov 25 09:44:46 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:44:46 -0500 Subject: [Freeswitch-users] mod_conference kick to abort invitations In-Reply-To: <000001ca6d56$66037c80$320a7580$@de> References: <000001ca6d56$66037c80$320a7580$@de> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B@jerris.com> Its a feature we don't have, patches welcome. Mike On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > Hi members, > I?m controlling freeswitch with the conference module via xmlrpc. > > Is it desired that the kick command can only kick users that are connected to the conference? > Is there no chance abort an invitation? > The kick command has no effect until the person I invited with the dial command is connected. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment.html From mike at jerris.com Wed Nov 25 09:45:50 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:45:50 -0500 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: Message-ID: In trunk there is a sofia profile setting to allow dialplan processing of 302 responses. This won't get you back into your same javascript, but you can probably do something clever from there. Mike On Nov 24, 2009, at 5:04 PM, John Platts wrote: > > I have considered writing JavaScript code to bridge two calls together. However, I would like to perform custom handling of the 302 Moved Temporarily response. How do I handle the 302 Moved Temporarily response if I use JavaScript? > From brian at freeswitch.org Wed Nov 25 09:46:05 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 11:46:05 -0600 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> Yes an alias will be required for every domain you run on the profile so it can find it. /b On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > Try an alias on the sip profile. > > Mike From mike at jerris.com Wed Nov 25 09:47:37 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:47:37 -0500 Subject: [Freeswitch-users] remote_media_ip variable not set In-Reply-To: <27c25bc40911240756k7842c80kd75be2d3d93441b9@mail.gmail.com> References: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> <2F929FDB-0E1B-49E0-A1E7-F4F1E2D548AD@avgs.ca> <27c25bc40911240756k7842c80kd75be2d3d93441b9@mail.gmail.com> Message-ID: It's possible it does not. I just added some code to set it on auto-adjust so it might be there sometimes now. You might need to add some code in mod_sofia to add it other times. Maybe it makes sense to move that var setting down to switch_rtp.c. Patches for this would be welcome. Thanks Mike On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > Hi, > > In the case of proxy_media=true, does it gets set at all then? From mike at jerris.com Wed Nov 25 09:48:39 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:48:39 -0500 Subject: [Freeswitch-users] How to find whether the destination extension supports encryption In-Reply-To: References: Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40@jerris.com> You can send the call with secure enabled and if it supports it it will use it. Mike On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > Hello, > > We have a mix of phones that support RTP encryption and those that do not. I have to support both types in the meanwhile, and would like to have encryption enabled on the relevant leg, even if the other leg does not support it (why? one of our ATAs either must have it unencrypted or have it encrypted, but cannot have both). > > How do I find whether the destination supports encryption? I do not want to manage an additional table in the database... > From srinivas.ksvreddy at gmail.com Wed Nov 25 09:55:01 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Wed, 25 Nov 2009 23:25:01 +0530 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: HI, thanks for your reply, my requirement is i am doing failover stuff with freeswitch. i dont want cut the calls when freeswitch dies, when failover happens mean one freeswitch dies we are going to start the second freeswitch, i dont want close call intiated by the first freeswtich, they are communicating with meida(bypass media). when one endpoing try to end the call at that time i want to close the call for the other end also. srinivas On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris wrote: > FreeSWITCH will kill the calls when you shut it down, if you intentionally > kill the network without shutting down FreeSWITCH the only thing you can do > is enable session timers or rtp timers in the soft phones to kill the call > when FreeSWITCH dies or when the call is over. > > Mike > > On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > > > Hi All, > > > > goodmorning to all, i have a scenario, two pjsua clients are connected > with Freeswitch and they are in call and bypass_media=true. i close the > Freeswitch server, still they are in call, again i started the Freeswitch, > and registerd these two endpoints, now how can i end the call(estabilished > by the first Freeswitch)? if i call re_invite will it estabilish the call > between two endpoints? > > any idea? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment.html From stevecrozz at gmail.com Wed Nov 25 10:01:14 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 25 Nov 2009 10:01:14 -0800 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: Message-ID: <11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com> Surprisingly, I've found no way to access the HTTP response status code using mod_spidermonkey_curl. I'd love to see this feature added or discussed if it already exists and I'm missing it. --Stephen On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris wrote: > In trunk there is a sofia profile setting to allow dialplan processing of > 302 responses. This won't get you back into your same javascript, but you > can probably do something clever from there. > > Mike > > On Nov 24, 2009, at 5:04 PM, John Platts wrote: > > > > > I have considered writing JavaScript code to bridge two calls together. > However, I would like to perform custom handling of the 302 Moved > Temporarily response. How do I handle the 302 Moved Temporarily response if > I use JavaScript? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment.html From tculjaga at gmail.com Wed Nov 25 10:04:56 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 25 Nov 2009 19:04:56 +0100 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: Message-ID: <65d96fc80911251004l401d5efbl8df3a2ac920207b8@mail.gmail.com> this is how i do it from the dialplan: On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris wrote: > In trunk there is a sofia profile setting to allow dialplan processing of > 302 responses. This won't get you back into your same javascript, but you > can probably do something clever from there. > > Mike > > On Nov 24, 2009, at 5:04 PM, John Platts wrote: > > > > > I have considered writing JavaScript code to bridge two calls together. > However, I would like to perform custom handling of the 302 Moved > Temporarily response. How do I handle the 302 Moved Temporarily response if > I use JavaScript? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment-0001.html From kokoska.rokoska at post.cz Wed Nov 25 10:09:38 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 25 Nov 2009 19:09:38 +0100 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <23f91030911250936q65dc7c25oe8c3746d6caed9e7@mail.gmail.com> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> <4B0D5561.4020009@post.cz> <191c3a030911250907x38aca94cs9128629fc2a1ba7c@mail.gmail.com> <4B0D6746.8060907@post.cz> <23f91030911250936q65dc7c25oe8c3746d6caed9e7@mail.gmail.com> Message-ID: <4B0D72E2.10607@post.cz> It was the first I want to do - update wiki :-) But someone was much faster (00:14, 30 October 2009 :-) http://wiki.freeswitch.org/wiki/Variable_record_min_sec Last time I looked for some hint about the recording (few months ago), this page (and even the variable) didn't exist... Best regards, kokoska.rokoska Seven Du napsal(a): > And you may also would like to update the wiki as well if the var is not > there. > > 2009/11/26 kokoska rokoska > > > Thank you very much, Anthony, for your help! > > I'm nearly at current trunk (15653) and > > works great :-) > > Many thanks once more! > > Best regards, > > kokoska.rokoska > > > Anthony Minessale napsal(a): > > use the variable RECORD_MIN_SEC > > > > This was added in revision 15271 so if you are below that I recommend > > updating to latest trunk. > > > > > > On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska > > > >> > wrote: > > > > Yes, Brian, I need them :-) > > > > They don't contain speech - instead, they contain few "computer > > generated" tones and I should store them in max quality for later > > proccessing (i.e. analysis)... > > > > Best regards, > > > > kokoska.rokoska > > > > > > Brian West napsal(a): > > > Really you want to keep 1-3 second files around? > > > > > > /b > > > > > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > > > > > >> Thank you very much, Brian, for your interest! > > >> > > >> It is standard recording: > > >> > > >> > > >> > > >> Best regards, > > >> > > >> kokoska.rokoska > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Wed Nov 25 10:17:21 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 13:17:21 -0500 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: For that you would need to fully exchange session state into the sip library, something that is not available in that lib at this time. On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: > HI, > thanks for your reply, my requirement is i am doing failover stuff with freeswitch. i dont want cut the calls when freeswitch dies, when failover happens mean one freeswitch dies we are going to start the second freeswitch, i dont want close call intiated by the first freeswtich, they are communicating with meida(bypass media). when one endpoing try to end the call at that time i want to close the call for the other end also. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/4bac4fdf/attachment.html From lists at redbonez.net Wed Nov 25 10:21:25 2009 From: lists at redbonez.net (Adam Ford) Date: Wed, 25 Nov 2009 11:21:25 -0700 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <20091124064509.GA6360@hijacked.us> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> <00e101ca6cab$c3525240$49f6f6c0$@net> <21CB5F92-98DE-4622-ADC5-013462A93BD2@freeswitch.org> <20091124064509.GA6360@hijacked.us> Message-ID: <016701ca6dfc$1a8e9ae0$4fabd0a0$@net> Awesome, thanks Andrew, I will have to keep an eye out for that patch. To continue, last night I decided to tackle the business hours and holiday routing on my FreeSWITCH system. It turned out to not be quite as simple with the XML dialplan as I thought. After being up until 1am banging my head against the wall, I finally got the results I was after. I decided I would share what I tried, why I tried them, and the solution I ended with, in hopes of helping other newcomers such as myself. Goal: To have a single area to modify in order to affect business hours and holiday routing across all extensions. Sources: http://wiki.freeswitch.org/wiki/Dialplan_XML http://wiki.freeswitch.org/wiki/Time_of_Day_routing http://svn.freeswitch.org/svn/freeswitch/trunk/conf/dialplan/default.xml After reading the above documentation of time of day routing, my first thought was that following the main example on http://wiki.freeswitch.org/wiki/Time_of_Day_routing would be wildly inefficient for me. I didn't want to create 3 extensions for every 1 real extension, nor did I want to edit each individual extension if there were adjustments to the hours or holiday schedule. I was filled with hope when I re-read the top of the default.xml dialplan. As it implies that I could simply set a variable at the top of the dialplan that would be accessible to all following extensions. You veterans immediately realize this is silly, but that is what is implied to a newcomer reading the top of the default.xml diaplan. After lots of playing with different ways of trying to get this to work, I went back and re-read http://wiki.freeswitch.org/wiki/Time_of_Day_routing. The bottom of which points out why it won't work the way that it is in the default.xml dialplan, a "classic case of dialplan is parsed all at once." My next attempt was to make a catchall extension at the top of default.xml that would set the ${Status} variable, then pass the originally dial extension into a transfer application, as suggested by the bottom of the Time_of_Day_routing wiki page. This obviously didn't work with a catchall, as it just continued to loop through the catchall. However, it did set the variable and pass through the dialed extension, so I felt I was on the right track. After trying a few different things with no avail, I realized the catchall extension would work if I just had it jump contexts on transfer! So I moved my catchall to the public.xml, adjusted my OpenZAP context to be public instead of default (which is apparently the default), and viola everything worked flawlessly. Now I just had to add the ability to set ${Status} to closed on holidays. There could be better ways of organizing this, but I just created a holidays directory and included the xml, which added override conditions for holidays. ---------------------------------------------------------------------- public.xml - ---------------------------------------------------------------------- holidays/thanksgiving.xml ---------------------------------------------------------------------- default context extension - ---------------------------------------------------------------------- Call flow breakdown (for those who are new, so you can easily follow what is going on) - In this example, if someone calls the DID ending in 5651, the call is processed by the catchall 'business_hours' extension in the public.xml. The weekday, and hour of day, is checked and ${Status} set accordingly. The holiday conditions that were included are then processed, overriding the ${Status}, setting it to 'closed' if today meets the criteria of a holiday. The call is then transferred to extension 5651, in the default context, of the XML dialplan. In the default context dialplan, extension 5651 is processed, which checks to see if ${Status} = open. If ${Status}=open is true, the extension passes the call to the IVR 'coral_ivr'. If ${Status}=open is false, the extension passes the call to the IVR 'coral_after_hours_ivr'. The catchall in the public context works well for me, because my connection to the PSTN is direct, through OpenZAP. I do not have to use the public context to provide extension security, as the incoming dialed extensions is limited to my known DIDs. I imagine this is not recommended for those who use SIP providers, and need the public context to provide internal and external segregation of your extensions. For these situations, you could probably create a middle-man context with the business hours logic, in which to send all transfers from public to default through. To those experienced users, if I missed something, and am making this much more complicated that it needs to be, please let me know. Or if you see potential errors or problems with my configuration, please let me know. I am new to this after all. Thanks all, especially you FreeSWITCH developers/contributors. I love the clean, efficient, logical design of FreeSWITCH. -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Thompson Sent: Monday, November 23, 2009 11:45 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Business/holiday hours routing On Mon, Nov 23, 2009 at 08:17:46PM -0600, Brian West wrote: > He's working on it for SVN... I recommended the format and to add the > phases of the moon and zodiac signs just for giggles. > I'll probably get a patch in this week (or early next) I'm thinking of changing the format so that "week of month" becomes its own value so you could compare against mweek as well as wday so thanksgiving + extension becomes something like If I really get ambitious I'd also like to allow wday="mon-fri" so I don't always forget that days are 1-indexed from sunday :) Andrew _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From afritzlists at fritztech.com Wed Nov 25 10:36:43 2009 From: afritzlists at fritztech.com (Andrew Fritz) Date: Wed, 25 Nov 2009 12:36:43 -0600 Subject: [Freeswitch-users] Patch: VMD Configurable MIN_TIME Message-ID: <4B0D793B.5040700@fritztech.com> I've created a patch to override the value of MIN_TIME in the vmd modules using a channel variable. In this way, it can be configured on a call by call basis. The channel variable is name "vmd_min_time". I didn't add the other detection parameters, but doing so would be straight forward. So, in our app, we can catch T-Mobile and the other problematic cell carriers beeps. I did this because in our app, we would rather have false positives than miss the start of recording on a voice mail system. This way, anyone using the VMD module can configure the vmd module to be as touchy or hard to trigger as they would like. I not sure how to implement it (at least in the vmd module code), but a way to make mod_vmd more robust to false positives, especially with short beeps would be to have it look for short silence immediately proceeding and/or following the beep. I've noticed that it tends to trigger on noise if there is tone in the noise, if for example I extend a syllable in a word or I have music on in the background. However on a voice mail system there will likely be a short near silence before the tone and an indefinite silence after it. In fact, background noise should be non-existent, except for line noise which should be Gaussian and not look like a structured tone. Looking for a beep + near silence after it for some period should eliminate many false positive where tones are embedded in other sounds (e.g. music or someone holding a vowel for longer than normal). Andrew -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: mod_vmd.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/653e39fa/attachment.txt From rupa at rupa.com Wed Nov 25 11:06:20 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 25 Nov 2009 13:06:20 -0600 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: <11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com> References: <11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com> Message-ID: Stephen, I think you've jumped into the middle of a thread about sip 302, not about http. Anyway, you might want to look at using mod_curl instead of mod_spidermonkey_curl. mod_curl can give you a json response which you can then parse easily in javascript or any other language. The json response has the http response code, all headers, and the body. On Wed, Nov 25, 2009 at 12:01 PM, Stephen Crosby wrote: > Surprisingly, I've found no way to access the HTTP response status code > using mod_spidermonkey_curl. I'd love to see this feature added or discussed > if it already exists and I'm missing it. > > --Stephen > > > On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris wrote: > >> In trunk there is a sofia profile setting to allow dialplan processing of >> 302 responses. This won't get you back into your same javascript, but you >> can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >> > >> > I have considered writing JavaScript code to bridge two calls together. >> However, I would like to perform custom handling of the 302 Moved >> Temporarily response. How do I handle the 302 Moved Temporarily response if >> I use JavaScript? >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/5fa958fc/attachment-0001.html From anthony.minessale at gmail.com Wed Nov 25 11:13:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 13:13:25 -0600 Subject: [Freeswitch-users] Patch: VMD Configurable MIN_TIME In-Reply-To: <4B0D793B.5040700@fritztech.com> References: <4B0D793B.5040700@fritztech.com> Message-ID: <191c3a030911251113t777a4516y1b23824c147f1d45@mail.gmail.com> can you submit your patch to jira as an improvement under the "application modules" section please. On Wed, Nov 25, 2009 at 12:36 PM, Andrew Fritz wrote: > I've created a patch to override the value of MIN_TIME in the vmd modules > using a channel variable. In this way, it can be configured on a call by > call basis. The channel variable is name "vmd_min_time". I didn't add the > other detection parameters, but doing so would be straight forward. So, in > our app, we can catch T-Mobile and the other problematic cell carriers > beeps. > > I did this because in our app, we would rather have false positives than > miss the start of recording on a voice mail system. This way, anyone using > the VMD module can configure the vmd module to be as touchy or hard to > trigger as they would like. > > I not sure how to implement it (at least in the vmd module code), but a way > to make mod_vmd more robust to false positives, especially with short beeps > would be to have it look for short silence immediately proceeding and/or > following the beep. I've noticed that it tends to trigger on noise if there > is tone in the noise, if for example I extend a syllable in a word or I have > music on in the background. > > However on a voice mail system there will likely be a short near silence > before the tone and an indefinite silence after it. In fact, background > noise should be non-existent, except for line noise which should be Gaussian > and not look like a structured tone. Looking for a beep + near silence after > it for some period should eliminate many false positive where tones are > embedded in other sounds (e.g. music or someone holding a vowel for longer > than normal). > > Andrew > > Index: src/mod/applications/mod_vmd/mod_vmd.c > =================================================================== > --- src/mod/applications/mod_vmd/mod_vmd.c (revision 15668) > +++ src/mod/applications/mod_vmd/mod_vmd.c (working copy) > @@ -162,6 +162,8 @@ > /*! A count of how long a distinct beep was detected > * by the discreet energy separation algorithm. */ > switch_size_t timestamp; > + /*! The MIN_TIME to use for this call */ > + int minTime; > } vmd_session_info_t; > > static switch_bool_t process_data(vmd_session_info_t * vmd_info, > switch_frame_t * frame); > @@ -312,7 +314,7 @@ > > if (c < (POINTS - MAX_CHIRP)) { > vmd_info->state = BEEP_NOT_DETECTED; > - if (vmd_info->timestamp < MIN_TIME) { > + if (vmd_info->timestamp < vmd_info->minTime) { > break; > } > > @@ -541,6 +543,7 @@ > switch_channel_t *channel; > vmd_session_info_t *vmd_info; > int i; > + const char *minTimeString; > > if (session == NULL) > return; > @@ -588,6 +591,14 @@ > > switch_channel_set_private(channel, "_vmd_", bug); > > + minTimeString = switch_channel_get_variable(channel, > "vmd_min_time"); > + if (minTimeString != 0) { > + sscanf(minTimeString,"%d",&(vmd_info->minTime)); > + } else { > + vmd_info->minTime = MIN_TIME; > + } > + > + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "MIN_TIME for > call: %d\n",vmd_info->minTime); > } > > /*! \brief Called when the module shuts down > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/1df26015/attachment.html From stevecrozz at gmail.com Wed Nov 25 11:26:49 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 25 Nov 2009 11:26:49 -0800 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: <11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com> Message-ID: <11990ade0911251126w1a6937fdga6ee6da79342305d@mail.gmail.com> My apologies, and thanks for the info. --Stephen On Wed, Nov 25, 2009 at 11:06 AM, Rupa Schomaker wrote: > Stephen, I think you've jumped into the middle of a thread about sip 302, > not about http. > > Anyway, you might want to look at using mod_curl instead of > mod_spidermonkey_curl. mod_curl can give you a json response which you can > then parse easily in javascript or any other language. The json response > has the http response code, all headers, and the body. > > > On Wed, Nov 25, 2009 at 12:01 PM, Stephen Crosby wrote: > >> Surprisingly, I've found no way to access the HTTP response status code >> using mod_spidermonkey_curl. I'd love to see this feature added or discussed >> if it already exists and I'm missing it. >> >> --Stephen >> >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan processing of >>> 302 responses. This won't get you back into your same javascript, but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>> > >>> > I have considered writing JavaScript code to bridge two calls together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily response if >>> I use JavaScript? >>> > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/9da7f56e/attachment.html From samuelmukoti at gmail.com Wed Nov 25 11:40:00 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Wed, 25 Nov 2009 21:40:00 +0200 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: References: Message-ID: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> Hi all, I'm wanting to try out a my first large scale setup at the office, 200 extensions and 24 POTS incoming, also a T1 line once the telco guys are ready. I wanted assistance with choosing the most appropriate hardware. We already have about 150 analogue phones, and I was wondering what's best? A couple of grandstream FXS GXW4024? Also for my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma or digium card? The best voice quality is paramount. Lastly for T1 what cards are recommeded, I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, would that perform? Or do I need hardware transcoding? Thank you, Sam Twitter: twitter.com/samuelmukoti On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > Today's Topics: > > 1. Re: mod_conference kick to abort invitations (Michael Jerris) > 2. Re: Handling the 302 Moved Temporarily response from > JavaScript (Michael Jerris) > 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) > 4. Re: remote_media_ip variable not set (Michael Jerris) > 5. Re: How to find whether the destination extension supports > encryption (Michael Jerris) > 6. Re: Bypass_media and re_invite (srinivasula reddy) > 7. Re: Handling the 302 Moved Temporarily response from > JavaScript (Stephen Crosby) > 8. Re: Handling the 302 Moved Temporarily response from > JavaScript (Tihomir Culjaga) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 25 Nov 2009 12:44:46 -0500 > From: Michael Jerris > Subject: Re: [Freeswitch-users] mod_conference kick to abort > invitations > To: freeswitch-users at lists.freeswitch.org > Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> > Content-Type: text/plain; charset="windows-1252" > > Its a feature we don't have, patches welcome. > > Mike > > On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > >> Hi members, >> I?m controlling freeswitch with the conference module via xmlrpc. >> >> Is it desired that the kick command can only kick users that are >> connected to the conference? >> Is there no chance abort an invitation? >> The kick command has no effect until the person I invited with the >> dial command is connected. > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Wed, 25 Nov 2009 12:45:50 -0500 > From: Michael Jerris > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > response from JavaScript > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset=us-ascii > > In trunk there is a sofia profile setting to allow dialplan > processing of 302 responses. This won't get you back into your same > javascript, but you can probably do something clever from there. > > Mike > > On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> >> I have considered writing JavaScript code to bridge two calls >> together. However, I would like to perform custom handling of the >> 302 Moved Temporarily response. How do I handle the 302 Moved >> Temporarily response if I use JavaScript? >> > > > > ------------------------------ > > Message: 3 > Date: Wed, 25 Nov 2009 11:46:05 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via > proxy. > To: freeswitch-users at lists.freeswitch.org > Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> > Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes > > Yes an alias will be required for every domain you run on the profile > so it can find it. > > /b > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > >> Try an alias on the sip profile. >> >> Mike > > > > > ------------------------------ > > Message: 4 > Date: Wed, 25 Nov 2009 12:47:37 -0500 > From: Michael Jerris > Subject: Re: [Freeswitch-users] remote_media_ip variable not set > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset=us-ascii > > It's possible it does not. I just added some code to set it on auto- > adjust so it might be there sometimes now. You might need to add > some code in mod_sofia to add it other times. Maybe it makes sense > to move that var setting down to switch_rtp.c. Patches for this > would be welcome. > > Thanks > > Mike > > On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > >> Hi, >> >> In the case of proxy_media=true, does it gets set at all then? > > > > > ------------------------------ > > Message: 5 > Date: Wed, 25 Nov 2009 12:48:39 -0500 > From: Michael Jerris > Subject: Re: [Freeswitch-users] How to find whether the destination > extension supports encryption > To: freeswitch-users at lists.freeswitch.org > Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> > Content-Type: text/plain; charset=us-ascii > > You can send the call with secure enabled and if it supports it it > will use it. > > Mike > > On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > >> Hello, >> >> We have a mix of phones that support RTP encryption and those that >> do not. I have to support both types in the meanwhile, and would >> like to have encryption enabled on the relevant leg, even if the >> other leg does not support it (why? one of our ATAs either must >> have it unencrypted or have it encrypted, but cannot have both). >> >> How do I find whether the destination supports encryption? I do not >> want to manage an additional table in the database... >> > > > > ------------------------------ > > Message: 6 > Date: Wed, 25 Nov 2009 23:25:01 +0530 > From: srinivasula reddy > Subject: Re: [Freeswitch-users] Bypass_media and re_invite > To: freeswitch-users at lists.freeswitch.org > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > HI, > thanks for your reply, my requirement is i am doing failover stuff > with > freeswitch. i dont want cut the calls when freeswitch dies, when > failover > happens mean one freeswitch dies we are going to start the second > freeswitch, i dont want close call intiated by the first > freeswtich, they > are communicating with meida(bypass media). when one endpoing try to > end the > call at that time i want to close the call for the other end also. > > > srinivas > > On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris > wrote: > >> FreeSWITCH will kill the calls when you shut it down, if you >> intentionally >> kill the network without shutting down FreeSWITCH the only thing >> you can do >> is enable session timers or rtp timers in the soft phones to kill >> the call >> when FreeSWITCH dies or when the call is over. >> >> Mike >> >> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >> >>> Hi All, >>> >>> goodmorning to all, i have a scenario, two pjsua clients are >>> connected >> with Freeswitch and they are in call and bypass_media=true. i >> close the >> Freeswitch server, still they are in call, again i started the >> Freeswitch, >> and registerd these two endpoints, now how can i end the call >> (estabilished >> by the first Freeswitch)? if i call re_invite will it estabilish >> the call >> between two endpoints? >>> any idea? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > > > -- > Srinivasula Reddy K > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html > > ------------------------------ > > Message: 7 > Date: Wed, 25 Nov 2009 10:01:14 -0800 > From: Stephen Crosby > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > response from JavaScript > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Surprisingly, I've found no way to access the HTTP response status > code > using mod_spidermonkey_curl. I'd love to see this feature added or > discussed > if it already exists and I'm missing it. > > --Stephen > > On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris > wrote: > >> In trunk there is a sofia profile setting to allow dialplan >> processing of >> 302 responses. This won't get you back into your same javascript, >> but you >> can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. >> However, I would like to perform custom handling of the 302 Moved >> Temporarily response. How do I handle the 302 Moved Temporarily >> response if >> I use JavaScript? >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html > > ------------------------------ > > Message: 8 > Date: Wed, 25 Nov 2009 19:04:56 +0100 > From: Tihomir Culjaga > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > response from JavaScript > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > this is how i do it from the dialplan: > > > > > > expression="^(300030)(.*)|^\+(300030)(.*)"> > > > > > data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> > data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: > 1:32} : > ${caller_id_number})}"/> > > data="aPfx=${caller_id_number:0:6}"/> > data="aNum=${caller_id_number:6:16}"/> > data="IP_ADDR=${network_addr}:5060"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris > wrote: > >> In trunk there is a sofia profile setting to allow dialplan >> processing of >> 302 responses. This won't get you back into your same javascript, >> but you >> can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. >> However, I would like to perform custom handling of the 302 Moved >> Temporarily response. How do I handle the 302 Moved Temporarily >> response if >> I use JavaScript? >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > End of FreeSWITCH-users Digest, Vol 41, Issue 189 > ************************************************* From srinivas.ksvreddy at gmail.com Wed Nov 25 11:47:43 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Thu, 26 Nov 2009 01:17:43 +0530 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: can please tell me how can i exchange session state into sip library. Thanks srinivas On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: > For that you would need to fully exchange session state into the sip > library, something that is not available in that lib at this time. > > > On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: > > HI, > thanks for your reply, my requirement is i am doing failover stuff with > freeswitch. i dont want cut the calls when freeswitch dies, when failover > happens mean one freeswitch dies we are going to start the second > freeswitch, i dont want close call intiated by the first freeswtich, they > are communicating with meida(bypass media). when one endpoing try to end the > call at that time i want to close the call for the other end also. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/fda28014/attachment.html From chris.chen2004 at gmail.com Wed Nov 25 11:55:19 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Wed, 25 Nov 2009 14:55:19 -0500 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> Message-ID: <507898380911251155k29c52989v30d0e39bb18d4ac1@mail.gmail.com> One suggestion to you, please never consider the GXW4108 for any business use unless just in LAB. The GXW4108 will work when it is working,but I can tell you within one year you will be regretting your choice for use of GXW4108 if you put into production for business use. Chris On Wed, Nov 25, 2009 at 2:40 PM, Samuel Mukoti wrote: > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org > wrote: > > > Send FreeSWITCH-users mailing list submissions to > > freeswitch-users at lists.freeswitch.org > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > or, via email, send a message with subject or body 'help' to > > freeswitch-users-request at lists.freeswitch.org > > > > You can reach the person managing the list at > > freeswitch-users-owner at lists.freeswitch.org > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of FreeSWITCH-users digest..." > > > > > > Today's Topics: > > > > 1. Re: mod_conference kick to abort invitations (Michael Jerris) > > 2. Re: Handling the 302 Moved Temporarily response from > > JavaScript (Michael Jerris) > > 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) > > 4. Re: remote_media_ip variable not set (Michael Jerris) > > 5. Re: How to find whether the destination extension supports > > encryption (Michael Jerris) > > 6. Re: Bypass_media and re_invite (srinivasula reddy) > > 7. Re: Handling the 302 Moved Temporarily response from > > JavaScript (Stephen Crosby) > > 8. Re: Handling the 302 Moved Temporarily response from > > JavaScript (Tihomir Culjaga) > > > > > > ---------------------------------------------------------------------- > > > > Message: 1 > > Date: Wed, 25 Nov 2009 12:44:46 -0500 > > From: Michael Jerris > > Subject: Re: [Freeswitch-users] mod_conference kick to abort > > invitations > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> > > Content-Type: text/plain; charset="windows-1252" > > > > Its a feature we don't have, patches welcome. > > > > Mike > > > > On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > > > >> Hi members, > >> I?m controlling freeswitch with the conference module via xmlrpc. > >> > >> Is it desired that the kick command can only kick users that are > >> connected to the conference? > >> Is there no chance abort an invitation? > >> The kick command has no effect until the person I invited with the > >> dial command is connected. > > > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html > > > > ------------------------------ > > > > Message: 2 > > Date: Wed, 25 Nov 2009 12:45:50 -0500 > > From: Michael Jerris > > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > > response from JavaScript > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: > > Content-Type: text/plain; charset=us-ascii > > > > In trunk there is a sofia profile setting to allow dialplan > > processing of 302 responses. This won't get you back into your same > > javascript, but you can probably do something clever from there. > > > > Mike > > > > On Nov 24, 2009, at 5:04 PM, John Platts wrote: > > > >> > >> I have considered writing JavaScript code to bridge two calls > >> together. However, I would like to perform custom handling of the > >> 302 Moved Temporarily response. How do I handle the 302 Moved > >> Temporarily response if I use JavaScript? > >> > > > > > > > > ------------------------------ > > > > Message: 3 > > Date: Wed, 25 Nov 2009 11:46:05 -0600 > > From: Brian West > > Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via > > proxy. > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> > > Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes > > > > Yes an alias will be required for every domain you run on the profile > > so it can find it. > > > > /b > > > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > > >> Try an alias on the sip profile. > >> > >> Mike > > > > > > > > > > ------------------------------ > > > > Message: 4 > > Date: Wed, 25 Nov 2009 12:47:37 -0500 > > From: Michael Jerris > > Subject: Re: [Freeswitch-users] remote_media_ip variable not set > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: > > Content-Type: text/plain; charset=us-ascii > > > > It's possible it does not. I just added some code to set it on auto- > > adjust so it might be there sometimes now. You might need to add > > some code in mod_sofia to add it other times. Maybe it makes sense > > to move that var setting down to switch_rtp.c. Patches for this > > would be welcome. > > > > Thanks > > > > Mike > > > > On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > > > >> Hi, > >> > >> In the case of proxy_media=true, does it gets set at all then? > > > > > > > > > > ------------------------------ > > > > Message: 5 > > Date: Wed, 25 Nov 2009 12:48:39 -0500 > > From: Michael Jerris > > Subject: Re: [Freeswitch-users] How to find whether the destination > > extension supports encryption > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> > > Content-Type: text/plain; charset=us-ascii > > > > You can send the call with secure enabled and if it supports it it > > will use it. > > > > Mike > > > > On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > > > >> Hello, > >> > >> We have a mix of phones that support RTP encryption and those that > >> do not. I have to support both types in the meanwhile, and would > >> like to have encryption enabled on the relevant leg, even if the > >> other leg does not support it (why? one of our ATAs either must > >> have it unencrypted or have it encrypted, but cannot have both). > >> > >> How do I find whether the destination supports encryption? I do not > >> want to manage an additional table in the database... > >> > > > > > > > > ------------------------------ > > > > Message: 6 > > Date: Wed, 25 Nov 2009 23:25:01 +0530 > > From: srinivasula reddy > > Subject: Re: [Freeswitch-users] Bypass_media and re_invite > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: > > > > Content-Type: text/plain; charset="iso-8859-1" > > > > HI, > > thanks for your reply, my requirement is i am doing failover stuff > > with > > freeswitch. i dont want cut the calls when freeswitch dies, when > > failover > > happens mean one freeswitch dies we are going to start the second > > freeswitch, i dont want close call intiated by the first > > freeswtich, they > > are communicating with meida(bypass media). when one endpoing try to > > end the > > call at that time i want to close the call for the other end also. > > > > > > srinivas > > > > On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris > > wrote: > > > >> FreeSWITCH will kill the calls when you shut it down, if you > >> intentionally > >> kill the network without shutting down FreeSWITCH the only thing > >> you can do > >> is enable session timers or rtp timers in the soft phones to kill > >> the call > >> when FreeSWITCH dies or when the call is over. > >> > >> Mike > >> > >> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > >> > >>> Hi All, > >>> > >>> goodmorning to all, i have a scenario, two pjsua clients are > >>> connected > >> with Freeswitch and they are in call and bypass_media=true. i > >> close the > >> Freeswitch server, still they are in call, again i started the > >> Freeswitch, > >> and registerd these two endpoints, now how can i end the call > >> (estabilished > >> by the first Freeswitch)? if i call re_invite will it estabilish > >> the call > >> between two endpoints? > >>> any idea? > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Srinivasula Reddy K > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html > > > > ------------------------------ > > > > Message: 7 > > Date: Wed, 25 Nov 2009 10:01:14 -0800 > > From: Stephen Crosby > > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > > response from JavaScript > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: > > <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> > > Content-Type: text/plain; charset="utf-8" > > > > Surprisingly, I've found no way to access the HTTP response status > > code > > using mod_spidermonkey_curl. I'd love to see this feature added or > > discussed > > if it already exists and I'm missing it. > > > > --Stephen > > > > On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris > > wrote: > > > >> In trunk there is a sofia profile setting to allow dialplan > >> processing of > >> 302 responses. This won't get you back into your same javascript, > >> but you > >> can probably do something clever from there. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> > >>> > >>> I have considered writing JavaScript code to bridge two calls > >>> together. > >> However, I would like to perform custom handling of the 302 Moved > >> Temporarily response. How do I handle the 302 Moved Temporarily > >> response if > >> I use JavaScript? > >>> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html > > > > ------------------------------ > > > > Message: 8 > > Date: Wed, 25 Nov 2009 19:04:56 +0100 > > From: Tihomir Culjaga > > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > > response from JavaScript > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: > > <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> > > Content-Type: text/plain; charset="iso-8859-1" > > > > this is how i do it from the dialplan: > > > > > > > > > > > > > expression="^(300030)(.*)|^\+(300030)(.*)"> > > > > > > > > > > > data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> > > > data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: > > 1:32} : > > ${caller_id_number})}"/> > > > > > data="aPfx=${caller_id_number:0:6}"/> > > > data="aNum=${caller_id_number:6:16}"/> > > > data="IP_ADDR=${network_addr}:5060"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris > > wrote: > > > >> In trunk there is a sofia profile setting to allow dialplan > >> processing of > >> 302 responses. This won't get you back into your same javascript, > >> but you > >> can probably do something clever from there. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> > >>> > >>> I have considered writing JavaScript code to bridge two calls > >>> together. > >> However, I would like to perform custom handling of the 302 Moved > >> Temporarily response. How do I handle the 302 Moved Temporarily > >> response if > >> I use JavaScript? > >>> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html > > > > ------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > > > > End of FreeSWITCH-users Digest, Vol 41, Issue 189 > > ************************************************* > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/79bd1f8e/attachment-0001.html From brian at freeswitch.org Wed Nov 25 11:59:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 13:59:03 -0600 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <507898380911251155k29c52989v30d0e39bb18d4ac1@mail.gmail.com> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <507898380911251155k29c52989v30d0e39bb18d4ac1@mail.gmail.com> Message-ID: Or if you're dancing with the stars!!!!!! /b On Nov 25, 2009, at 1:55 PM, Chris Chen wrote: > One suggestion to you, please never consider the GXW4108 for any > business use unless just in LAB. The GXW4108 will work when it is > working,but I can tell you within one year you will be regretting > your choice for use of GXW4108 if you put into production for > business use. > > Chris From samuelmukoti at gmail.com Wed Nov 25 12:16:39 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Wed, 25 Nov 2009 22:16:39 +0200 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> Message-ID: Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews at sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org > wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> In trunk there is a sofia profile setting to allow dialplan >> processing of 302 responses. This won't get you back into your same >> javascript, but you can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. However, I would like to perform custom handling of the >>> 302 Moved Temporarily response. How do I handle the 302 Moved >>> Temporarily response if I use JavaScript? >>> >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Wed, 25 Nov 2009 11:46:05 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> proxy. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes >> >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >>> Try an alias on the sip profile. >>> >>> Mike >> >> >> >> >> ------------------------------ >> >> Message: 4 >> Date: Wed, 25 Nov 2009 12:47:37 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> It's possible it does not. I just added some code to set it on auto- >> adjust so it might be there sometimes now. You might need to add >> some code in mod_sofia to add it other times. Maybe it makes sense >> to move that var setting down to switch_rtp.c. Patches for this >> would be welcome. >> >> Thanks >> >> Mike >> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: >> >>> Hi, >>> >>> In the case of proxy_media=true, does it gets set at all then? >> >> >> >> >> ------------------------------ >> >> Message: 5 >> Date: Wed, 25 Nov 2009 12:48:39 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] How to find whether the destination >> extension supports encryption >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> >> Content-Type: text/plain; charset=us-ascii >> >> You can send the call with secure enabled and if it supports it it >> will use it. >> >> Mike >> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: >> >>> Hello, >>> >>> We have a mix of phones that support RTP encryption and those that >>> do not. I have to support both types in the meanwhile, and would >>> like to have encryption enabled on the relevant leg, even if the >>> other leg does not support it (why? one of our ATAs either must >>> have it unencrypted or have it encrypted, but cannot have both). >>> >>> How do I find whether the destination supports encryption? I do not >>> want to manage an additional table in the database... >>> >> >> >> >> ------------------------------ >> >> Message: 6 >> Date: Wed, 25 Nov 2009 23:25:01 +0530 >> From: srinivasula reddy >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> >> Content-Type: text/plain; charset="iso-8859-1" >> >> HI, >> thanks for your reply, my requirement is i am doing failover stuff >> with >> freeswitch. i dont want cut the calls when freeswitch dies, when >> failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the first >> freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to >> end the >> call at that time i want to close the call for the other end also. >> >> >> srinivas >> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris >> wrote: >> >>> FreeSWITCH will kill the calls when you shut it down, if you >>> intentionally >>> kill the network without shutting down FreeSWITCH the only thing >>> you can do >>> is enable session timers or rtp timers in the soft phones to kill >>> the call >>> when FreeSWITCH dies or when the call is over. >>> >>> Mike >>> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >>> >>>> Hi All, >>>> >>>> goodmorning to all, i have a scenario, two pjsua clients are >>>> connected >>> with Freeswitch and they are in call and bypass_media=true. i >>> close the >>> Freeswitch server, still they are in call, again i started the >>> Freeswitch, >>> and registerd these two endpoints, now how can i end the call >>> (estabilished >>> by the first Freeswitch)? if i call re_invite will it estabilish >>> the call >>> between two endpoints? >>>> any idea? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Srinivasula Reddy K >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html >> >> ------------------------------ >> >> Message: 7 >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> From: Stephen Crosby >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Surprisingly, I've found no way to access the HTTP response status >> code >> using mod_spidermonkey_curl. I'd love to see this feature added or >> discussed >> if it already exists and I'm missing it. >> >> --Stephen >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html >> >> ------------------------------ >> >> Message: 8 >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> From: Tihomir Culjaga >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> this is how i do it from the dialplan: >> >> >> >> >> >> > expression="^(300030)(.*)|^\+(300030)(.*)"> >> >> >> >> >> > data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> >> > data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: >> 1:32} : >> ${caller_id_number})}"/> >> >> > data="aPfx=${caller_id_number:0:6}"/> >> > data="aNum=${caller_id_number:6:16}"/> >> > data="IP_ADDR=${network_addr}:5060"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 >> ************************************************* > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From kristian.kielhofner at gmail.com Wed Nov 25 12:30:09 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 25 Nov 2009 15:30:09 -0500 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <507898380911251155k29c52989v30d0e39bb18d4ac1@mail.gmail.com> Message-ID: <2d9149cd0911251230h71ebc7b8n1203628bf97ed218@mail.gmail.com> That was a *GREAT* e-mail. On Wed, Nov 25, 2009 at 2:59 PM, Brian West wrote: > Or if you're dancing with the stars!!!!!! > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From mike at jerris.com Wed Nov 25 12:35:20 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 15:35:20 -0500 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: "something that is not available in that lib at this time." Mike On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote: > can please tell me how can i exchange session state into sip library. > > Thanks > srinivas > > On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: > For that you would need to fully exchange session state into the sip library, something that is not available in that lib at this time. > > > On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: > >> HI, >> thanks for your reply, my requirement is i am doing failover stuff with freeswitch. i dont want cut the calls when freeswitch dies, when failover happens mean one freeswitch dies we are going to start the second freeswitch, i dont want close call intiated by the first freeswtich, they are communicating with meida(bypass media). when one endpoing try to end the call at that time i want to close the call for the other end also. >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/c9abfc4c/attachment-0001.html From chris.chen2004 at gmail.com Wed Nov 25 12:40:25 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Wed, 25 Nov 2009 15:40:25 -0500 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> Message-ID: <507898380911251240q1a71b234x1458284bbdccc093@mail.gmail.com> You haven't really put it into production for more than one year. The issue with GXW4108 is that after some time, say a couple of months, either all FXO ports not working, or worse, some FXO ports not working, but after power recycling, they will come back to work for some time until on strike again at some time you have no control. This had been reported for a couple of years without improvement. Go google search you will find out, this has happened to many GXW4108 users. On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti wrote: > Thank you for those tips, > > I do have some small setups using gxw4108 they work or, except CID > doesn't seem to work. I will try the channel bank route - just don't > know too much about the setup options or how you'd purchase the > correct config, eg. For 150 FXS channel bank, can I get a single PCI > card for that? > > I may end up using the grandstream fxs gateways then use the T1 > channel bank from sangoma, > > Thank you all.. > > Lastly, I know asterisk now has an offical skype_ module, Is there > anything similar I could use? > > > On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: > > > Samuel - you could go with FXS gateways or channel banks. If you go > > the gateway route Grandstream or Audiocodes would work fine. > > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > > you could use a 24FXO channel bank cross connected with Rhino T1 > > cards, or individual FXO gateways but you may have a hard time > > finding 24 ports of FXO in a single GW. Best performing T1 cards in > > my experience (thousands of deployments) are Sangoma. Your server > > configuration looks fine. > > > > Cory J. Andrews > > Director New Market Initiatives > > > > Sayers Media Group > > VoIP Supply, LLC > > 454 Sonwil Drive > > Buffalo, NY 14225 > > 716-250-3402 OFFICE > > 716-630-1548 FAX > > 716-601-4474 MOBILE > > candrews at sayersmedia.com > > > > > > Have I exceeded your expectations? Please share your experience > > with my boss, Benjamin P. Sayers, CEO > > > > NOTICE: The information contained in this email and any document > > attached hereto is intended only for the named recipient(s). It is > > the property of the VoIP Supply, LLC and shall not be used, > > disclosed or reproduced without the express written consent of VoIP > > Supply, LLC. If you are not the intended recipient, nor the employee > > or agent responsible for delivering this message in confidence to > > the intended recipient(s), you are hereby notified that you have > > received this transmittal in error, and any review, dissemination, > > distribution or copying of this transmittal or its attachments is > > strictly prohibited. If you have received this transmittal and/or > > attachments in error, please notify me immediately by reply e-mail > > or telephone and then delete this message, including any > > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > > 14225 USA. > > > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Samuel Mukoti > > Sent: Wednesday, November 25, 2009 2:40 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Grandstream gateways > > > > Hi all, > > > > I'm wanting to try out a my first large scale setup at the office, 200 > > extensions and 24 POTS incoming, also a T1 line once the telco guys > > are ready. I wanted assistance with choosing the most appropriate > > hardware. We already have about 150 analogue phones, and I was > > wondering what's best? A couple of grandstream FXS GXW4024? Also for > > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > > or digium card? The best voice quality is paramount. Lastly for T1 > > what cards are recommeded, > > > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > > would that perform? Or do I need hardware transcoding? > > > > Thank you, > > > > Sam > > > > Twitter: twitter.com/samuelmukoti > > > > > > On 25 Nov,2009, at 8:05 PM, > freeswitch-users-request at lists.freeswitch.org > > wrote: > > > >> Send FreeSWITCH-users mailing list submissions to > >> freeswitch-users at lists.freeswitch.org > >> > >> To subscribe or unsubscribe via the World Wide Web, visit > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> or, via email, send a message with subject or body 'help' to > >> freeswitch-users-request at lists.freeswitch.org > >> > >> You can reach the person managing the list at > >> freeswitch-users-owner at lists.freeswitch.org > >> > >> When replying, please edit your Subject line so it is more specific > >> than "Re: Contents of FreeSWITCH-users digest..." > >> > >> > >> Today's Topics: > >> > >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) > >> 2. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Michael Jerris) > >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) > >> 4. Re: remote_media_ip variable not set (Michael Jerris) > >> 5. Re: How to find whether the destination extension supports > >> encryption (Michael Jerris) > >> 6. Re: Bypass_media and re_invite (srinivasula reddy) > >> 7. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Stephen Crosby) > >> 8. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Tihomir Culjaga) > >> > >> > >> --- > >> ------------------------------------------------------------------- > >> > >> Message: 1 > >> Date: Wed, 25 Nov 2009 12:44:46 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] mod_conference kick to abort > >> invitations > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> > >> Content-Type: text/plain; charset="windows-1252" > >> > >> Its a feature we don't have, patches welcome. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > >> > >>> Hi members, > >>> I?m controlling freeswitch with the conference module via xmlrpc. > >>> > >>> Is it desired that the kick command can only kick users that are > >>> connected to the conference? > >>> Is there no chance abort an invitation? > >>> The kick command has no effect until the person I invited with the > >>> dial command is connected. > >> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 2 > >> Date: Wed, 25 Nov 2009 12:45:50 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii > >> > >> In trunk there is a sofia profile setting to allow dialplan > >> processing of 302 responses. This won't get you back into your same > >> javascript, but you can probably do something clever from there. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> > >>> > >>> I have considered writing JavaScript code to bridge two calls > >>> together. However, I would like to perform custom handling of the > >>> 302 Moved Temporarily response. How do I handle the 302 Moved > >>> Temporarily response if I use JavaScript? > >>> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 3 > >> Date: Wed, 25 Nov 2009 11:46:05 -0600 > >> From: Brian West > >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via > >> proxy. > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> > >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes > >> > >> Yes an alias will be required for every domain you run on the profile > >> so it can find it. > >> > >> /b > >> > >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > >> > >>> Try an alias on the sip profile. > >>> > >>> Mike > >> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 4 > >> Date: Wed, 25 Nov 2009 12:47:37 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii > >> > >> It's possible it does not. I just added some code to set it on auto- > >> adjust so it might be there sometimes now. You might need to add > >> some code in mod_sofia to add it other times. Maybe it makes sense > >> to move that var setting down to switch_rtp.c. Patches for this > >> would be welcome. > >> > >> Thanks > >> > >> Mike > >> > >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > >> > >>> Hi, > >>> > >>> In the case of proxy_media=true, does it gets set at all then? > >> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 5 > >> Date: Wed, 25 Nov 2009 12:48:39 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] How to find whether the destination > >> extension supports encryption > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> > >> Content-Type: text/plain; charset=us-ascii > >> > >> You can send the call with secure enabled and if it supports it it > >> will use it. > >> > >> Mike > >> > >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > >> > >>> Hello, > >>> > >>> We have a mix of phones that support RTP encryption and those that > >>> do not. I have to support both types in the meanwhile, and would > >>> like to have encryption enabled on the relevant leg, even if the > >>> other leg does not support it (why? one of our ATAs either must > >>> have it unencrypted or have it encrypted, but cannot have both). > >>> > >>> How do I find whether the destination supports encryption? I do not > >>> want to manage an additional table in the database... > >>> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 6 > >> Date: Wed, 25 Nov 2009 23:25:01 +0530 > >> From: srinivasula reddy > >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> > >> Content-Type: text/plain; charset="iso-8859-1" > >> > >> HI, > >> thanks for your reply, my requirement is i am doing failover stuff > >> with > >> freeswitch. i dont want cut the calls when freeswitch dies, when > >> failover > >> happens mean one freeswitch dies we are going to start the second > >> freeswitch, i dont want close call intiated by the first > >> freeswtich, they > >> are communicating with meida(bypass media). when one endpoing try to > >> end the > >> call at that time i want to close the call for the other end also. > >> > >> > >> srinivas > >> > >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris > >> wrote: > >> > >>> FreeSWITCH will kill the calls when you shut it down, if you > >>> intentionally > >>> kill the network without shutting down FreeSWITCH the only thing > >>> you can do > >>> is enable session timers or rtp timers in the soft phones to kill > >>> the call > >>> when FreeSWITCH dies or when the call is over. > >>> > >>> Mike > >>> > >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > >>> > >>>> Hi All, > >>>> > >>>> goodmorning to all, i have a scenario, two pjsua clients are > >>>> connected > >>> with Freeswitch and they are in call and bypass_media=true. i > >>> close the > >>> Freeswitch server, still they are in call, again i started the > >>> Freeswitch, > >>> and registerd these two endpoints, now how can i end the call > >>> (estabilished > >>> by the first Freeswitch)? if i call re_invite will it estabilish > >>> the call > >>> between two endpoints? > >>>> any idea? > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Srinivasula Reddy K > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 7 > >> Date: Wed, 25 Nov 2009 10:01:14 -0800 > >> From: Stephen Crosby > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> > >> Content-Type: text/plain; charset="utf-8" > >> > >> Surprisingly, I've found no way to access the HTTP response status > >> code > >> using mod_spidermonkey_curl. I'd love to see this feature added or > >> discussed > >> if it already exists and I'm missing it. > >> > >> --Stephen > >> > >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris > >> wrote: > >> > >>> In trunk there is a sofia profile setting to allow dialplan > >>> processing of > >>> 302 responses. This won't get you back into your same javascript, > >>> but you > >>> can probably do something clever from there. > >>> > >>> Mike > >>> > >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >>> > >>>> > >>>> I have considered writing JavaScript code to bridge two calls > >>>> together. > >>> However, I would like to perform custom handling of the 302 Moved > >>> Temporarily response. How do I handle the 302 Moved Temporarily > >>> response if > >>> I use JavaScript? > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 8 > >> Date: Wed, 25 Nov 2009 19:04:56 +0100 > >> From: Tihomir Culjaga > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> > >> Content-Type: text/plain; charset="iso-8859-1" > >> > >> this is how i do it from the dialplan: > >> > >> > >> > >> > >> > >> >> expression="^(300030)(.*)|^\+(300030)(.*)"> > >> > >> > >> > >> > >> >> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> > >> >> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: > >> 1:32} : > >> ${caller_id_number})}"/> > >> > >> >> data="aPfx=${caller_id_number:0:6}"/> > >> >> data="aNum=${caller_id_number:6:16}"/> > >> >> data="IP_ADDR=${network_addr}:5060"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris > >> wrote: > >> > >>> In trunk there is a sofia profile setting to allow dialplan > >>> processing of > >>> 302 responses. This won't get you back into your same javascript, > >>> but you > >>> can probably do something clever from there. > >>> > >>> Mike > >>> > >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >>> > >>>> > >>>> I have considered writing JavaScript code to bridge two calls > >>>> together. > >>> However, I would like to perform custom handling of the 302 Moved > >>> Temporarily response. How do I handle the 302 Moved Temporarily > >>> response if > >>> I use JavaScript? > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html > >> > >> ------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > >> > >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 > >> ************************************************* > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/f6d17116/attachment-0001.html From brian at freeswitch.org Wed Nov 25 12:45:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 14:45:01 -0600 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <507898380911251240q1a71b234x1458284bbdccc093@mail.gmail.com> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <507898380911251240q1a71b234x1458284bbdccc093@mail.gmail.com> Message-ID: <5D71F7D7-7E93-499C-AFCA-61846CB3217F@freeswitch.org> Kill it, sunshine. /b On Nov 25, 2009, at 2:40 PM, Chris Chen wrote: > You haven't really put it into production for more than one year. > The issue with GXW4108 is that after some time, say a couple of > months, either all FXO ports not working, or worse, some FXO ports > not working, but after power recycling, they will come back to work > for some time until on strike again at some time you have no control. > > This had been reported for a couple of years without improvement. Go > google search you will find out, this has happened to many GXW4108 > users. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/9e386b99/attachment.html From srinivas.ksvreddy at gmail.com Wed Nov 25 12:58:33 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Thu, 26 Nov 2009 02:28:33 +0530 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: thanks for your reply mike, is there any api in freeswitch or any thing else to update lib programatically from pjsua. srinivas On Thu, Nov 26, 2009 at 2:05 AM, Michael Jerris wrote: > "something that is not available in that lib at this time." > > Mike > > On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote > > can please tell me how can i exchange session state into sip library. > > Thanks > srinivas > > On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: > >> For that you would need to fully exchange session state into the sip >> library, *something that is not available in that lib at this time.* >> >> >> On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: >> >> HI, >> thanks for your reply, my requirement is i am doing failover stuff with >> freeswitch. i dont want cut the calls when freeswitch dies, when failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the first freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to end the >> call at that time i want to close the call for the other end also. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/bdac13c8/attachment.html From testeador01 at gmail.com Wed Nov 25 13:00:14 2009 From: testeador01 at gmail.com (Milena) Date: Wed, 25 Nov 2009 16:00:14 -0500 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <507898380911251240q1a71b234x1458284bbdccc093@mail.gmail.com> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <507898380911251240q1a71b234x1458284bbdccc093@mail.gmail.com> Message-ID: Hello, Samuel: We also have some GXW4104 gateways, in small production/testing environments; our caller id works fine and none of them has failed in over a year of being used. The thing that i dislike about the GXW series is that it has no support for early media. Everyone: What FXO devices do you currently use / recommend? 2009/11/25 Chris Chen > You haven't really put it into production for more than one year. The issue > with GXW4108 is that after some time, say a couple of months, either all FXO > ports not working, or worse, some FXO ports not working, but after power > recycling, they will come back to work for some time until on strike again > at some time you have no control. > > This had been reported for a couple of years without improvement. Go google > search you will find out, this has happened to many GXW4108 users. > > > > On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti wrote: > >> Thank you for those tips, >> >> I do have some small setups using gxw4108 they work or, except CID >> doesn't seem to work. I will try the channel bank route - just don't >> know too much about the setup options or how you'd purchase the >> correct config, eg. For 150 FXS channel bank, can I get a single PCI >> card for that? >> >> I may end up using the grandstream fxs gateways then use the T1 >> channel bank from sangoma, >> >> Thank you all.. >> >> Lastly, I know asterisk now has an offical skype_ module, Is there >> anything similar I could use? >> >> >> On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: >> >> > Samuel - you could go with FXS gateways or channel banks. If you go >> > the gateway route Grandstream or Audiocodes would work fine. >> > Audiocodes are a bit more telco grade. If you have 25 POTS incoming >> > you could use a 24FXO channel bank cross connected with Rhino T1 >> > cards, or individual FXO gateways but you may have a hard time >> > finding 24 ports of FXO in a single GW. Best performing T1 cards in >> > my experience (thousands of deployments) are Sangoma. Your server >> > configuration looks fine. >> > >> > Cory J. Andrews >> > Director New Market Initiatives >> > >> > Sayers Media Group >> > VoIP Supply, LLC >> > 454 Sonwil Drive >> > Buffalo, NY 14225 >> > 716-250-3402 OFFICE >> > 716-630-1548 FAX >> > 716-601-4474 MOBILE >> > candrews at sayersmedia.com >> > >> > >> > Have I exceeded your expectations? Please share your experience >> > with my boss, Benjamin P. Sayers, CEO >> > >> > NOTICE: The information contained in this email and any document >> > attached hereto is intended only for the named recipient(s). It is >> > the property of the VoIP Supply, LLC and shall not be used, >> > disclosed or reproduced without the express written consent of VoIP >> > Supply, LLC. If you are not the intended recipient, nor the employee >> > or agent responsible for delivering this message in confidence to >> > the intended recipient(s), you are hereby notified that you have >> > received this transmittal in error, and any review, dissemination, >> > distribution or copying of this transmittal or its attachments is >> > strictly prohibited. If you have received this transmittal and/or >> > attachments in error, please notify me immediately by reply e-mail >> > or telephone and then delete this message, including any >> > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY >> > 14225 USA. >> >> > >> > >> > >> > -----Original Message----- >> > From: freeswitch-users-bounces at lists.freeswitch.org >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> > Samuel Mukoti >> > Sent: Wednesday, November 25, 2009 2:40 PM >> > To: freeswitch-users at lists.freeswitch.org >> > Subject: [Freeswitch-users] Grandstream gateways >> > >> > Hi all, >> > >> > I'm wanting to try out a my first large scale setup at the office, 200 >> > extensions and 24 POTS incoming, also a T1 line once the telco guys >> > are ready. I wanted assistance with choosing the most appropriate >> > hardware. We already have about 150 analogue phones, and I was >> > wondering what's best? A couple of grandstream FXS GXW4024? Also for >> > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma >> > or digium card? The best voice quality is paramount. Lastly for T1 >> > what cards are recommeded, >> > >> > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, >> > would that perform? Or do I need hardware transcoding? >> > >> > Thank you, >> > >> > Sam >> > >> > Twitter: twitter.com/samuelmukoti >> > >> > >> > On 25 Nov,2009, at 8:05 PM, >> freeswitch-users-request at lists.freeswitch.org >> > wrote: >> > >> >> Send FreeSWITCH-users mailing list submissions to >> >> freeswitch-users at lists.freeswitch.org >> >> >> >> To subscribe or unsubscribe via the World Wide Web, visit >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> or, via email, send a message with subject or body 'help' to >> >> freeswitch-users-request at lists.freeswitch.org >> >> >> >> You can reach the person managing the list at >> >> freeswitch-users-owner at lists.freeswitch.org >> >> >> >> When replying, please edit your Subject line so it is more specific >> >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> >> >> >> Today's Topics: >> >> >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> >> 2. Re: Handling the 302 Moved Temporarily response from >> >> JavaScript (Michael Jerris) >> >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> >> 5. Re: How to find whether the destination extension supports >> >> encryption (Michael Jerris) >> >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> >> 7. Re: Handling the 302 Moved Temporarily response from >> >> JavaScript (Stephen Crosby) >> >> 8. Re: Handling the 302 Moved Temporarily response from >> >> JavaScript (Tihomir Culjaga) >> >> >> >> >> >> --- >> >> ------------------------------------------------------------------- >> >> >> >> Message: 1 >> >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> >> From: Michael Jerris >> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> >> invitations >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> >> >> Content-Type: text/plain; charset="windows-1252" >> >> >> >> Its a feature we don't have, patches welcome. >> >> >> >> Mike >> >> >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >> >> >>> Hi members, >> >>> I?m controlling freeswitch with the conference module via xmlrpc. >> >>> >> >>> Is it desired that the kick command can only kick users that are >> >>> connected to the conference? >> >>> Is there no chance abort an invitation? >> >>> The kick command has no effect until the person I invited with the >> >>> dial command is connected. >> >> >> >> -------------- next part -------------- >> >> An HTML attachment was scrubbed... >> >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html >> >> >> >> ------------------------------ >> >> >> >> Message: 2 >> >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> >> From: Michael Jerris >> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> >> response from JavaScript >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: >> >> Content-Type: text/plain; charset=us-ascii >> >> >> >> In trunk there is a sofia profile setting to allow dialplan >> >> processing of 302 responses. This won't get you back into your same >> >> javascript, but you can probably do something clever from there. >> >> >> >> Mike >> >> >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >> >> >>> >> >>> I have considered writing JavaScript code to bridge two calls >> >>> together. However, I would like to perform custom handling of the >> >>> 302 Moved Temporarily response. How do I handle the 302 Moved >> >>> Temporarily response if I use JavaScript? >> >>> >> >> >> >> >> >> >> >> ------------------------------ >> >> >> >> Message: 3 >> >> Date: Wed, 25 Nov 2009 11:46:05 -0600 >> >> From: Brian West >> >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> >> proxy. >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> >> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes >> >> >> >> Yes an alias will be required for every domain you run on the profile >> >> so it can find it. >> >> >> >> /b >> >> >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >> >> >>> Try an alias on the sip profile. >> >>> >> >>> Mike >> >> >> >> >> >> >> >> >> >> ------------------------------ >> >> >> >> Message: 4 >> >> Date: Wed, 25 Nov 2009 12:47:37 -0500 >> >> From: Michael Jerris >> >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: >> >> Content-Type: text/plain; charset=us-ascii >> >> >> >> It's possible it does not. I just added some code to set it on auto- >> >> adjust so it might be there sometimes now. You might need to add >> >> some code in mod_sofia to add it other times. Maybe it makes sense >> >> to move that var setting down to switch_rtp.c. Patches for this >> >> would be welcome. >> >> >> >> Thanks >> >> >> >> Mike >> >> >> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: >> >> >> >>> Hi, >> >>> >> >>> In the case of proxy_media=true, does it gets set at all then? >> >> >> >> >> >> >> >> >> >> ------------------------------ >> >> >> >> Message: 5 >> >> Date: Wed, 25 Nov 2009 12:48:39 -0500 >> >> From: Michael Jerris >> >> Subject: Re: [Freeswitch-users] How to find whether the destination >> >> extension supports encryption >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> >> >> Content-Type: text/plain; charset=us-ascii >> >> >> >> You can send the call with secure enabled and if it supports it it >> >> will use it. >> >> >> >> Mike >> >> >> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: >> >> >> >>> Hello, >> >>> >> >>> We have a mix of phones that support RTP encryption and those that >> >>> do not. I have to support both types in the meanwhile, and would >> >>> like to have encryption enabled on the relevant leg, even if the >> >>> other leg does not support it (why? one of our ATAs either must >> >>> have it unencrypted or have it encrypted, but cannot have both). >> >>> >> >>> How do I find whether the destination supports encryption? I do not >> >>> want to manage an additional table in the database... >> >>> >> >> >> >> >> >> >> >> ------------------------------ >> >> >> >> Message: 6 >> >> Date: Wed, 25 Nov 2009 23:25:01 +0530 >> >> From: srinivasula reddy >> >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: >> >> >> >> Content-Type: text/plain; charset="iso-8859-1" >> >> >> >> HI, >> >> thanks for your reply, my requirement is i am doing failover stuff >> >> with >> >> freeswitch. i dont want cut the calls when freeswitch dies, when >> >> failover >> >> happens mean one freeswitch dies we are going to start the second >> >> freeswitch, i dont want close call intiated by the first >> >> freeswtich, they >> >> are communicating with meida(bypass media). when one endpoing try to >> >> end the >> >> call at that time i want to close the call for the other end also. >> >> >> >> >> >> srinivas >> >> >> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris >> >> wrote: >> >> >> >>> FreeSWITCH will kill the calls when you shut it down, if you >> >>> intentionally >> >>> kill the network without shutting down FreeSWITCH the only thing >> >>> you can do >> >>> is enable session timers or rtp timers in the soft phones to kill >> >>> the call >> >>> when FreeSWITCH dies or when the call is over. >> >>> >> >>> Mike >> >>> >> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >> >>> >> >>>> Hi All, >> >>>> >> >>>> goodmorning to all, i have a scenario, two pjsua clients are >> >>>> connected >> >>> with Freeswitch and they are in call and bypass_media=true. i >> >>> close the >> >>> Freeswitch server, still they are in call, again i started the >> >>> Freeswitch, >> >>> and registerd these two endpoints, now how can i end the call >> >>> (estabilished >> >>> by the first Freeswitch)? if i call re_invite will it estabilish >> >>> the call >> >>> between two endpoints? >> >>>> any idea? >> >>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>> users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> >> >> >> >> -- >> >> Srinivasula Reddy K >> >> -------------- next part -------------- >> >> An HTML attachment was scrubbed... >> >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html >> >> >> >> ------------------------------ >> >> >> >> Message: 7 >> >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> >> From: Stephen Crosby >> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> >> response from JavaScript >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: >> >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> >> >> Content-Type: text/plain; charset="utf-8" >> >> >> >> Surprisingly, I've found no way to access the HTTP response status >> >> code >> >> using mod_spidermonkey_curl. I'd love to see this feature added or >> >> discussed >> >> if it already exists and I'm missing it. >> >> >> >> --Stephen >> >> >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris >> >> wrote: >> >> >> >>> In trunk there is a sofia profile setting to allow dialplan >> >>> processing of >> >>> 302 responses. This won't get you back into your same javascript, >> >>> but you >> >>> can probably do something clever from there. >> >>> >> >>> Mike >> >>> >> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >> >>>> >> >>>> I have considered writing JavaScript code to bridge two calls >> >>>> together. >> >>> However, I would like to perform custom handling of the 302 Moved >> >>> Temporarily response. How do I handle the 302 Moved Temporarily >> >>> response if >> >>> I use JavaScript? >> >>>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>> users >> >>> http://www.freeswitch.org >> >>> >> >> -------------- next part -------------- >> >> An HTML attachment was scrubbed... >> >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html >> >> >> >> ------------------------------ >> >> >> >> Message: 8 >> >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> >> From: Tihomir Culjaga >> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> >> response from JavaScript >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: >> >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> >> >> Content-Type: text/plain; charset="iso-8859-1" >> >> >> >> this is how i do it from the dialplan: >> >> >> >> >> >> >> >> >> >> >> >> > >> expression="^(300030)(.*)|^\+(300030)(.*)"> >> >> >> >> >> >> >> >> >> >> > >> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> >> >> > >> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: >> >> 1:32} : >> >> ${caller_id_number})}"/> >> >> >> >> > >> data="aPfx=${caller_id_number:0:6}"/> >> >> > >> data="aNum=${caller_id_number:6:16}"/> >> >> > >> data="IP_ADDR=${network_addr}:5060"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris >> >> wrote: >> >> >> >>> In trunk there is a sofia profile setting to allow dialplan >> >>> processing of >> >>> 302 responses. This won't get you back into your same javascript, >> >>> but you >> >>> can probably do something clever from there. >> >>> >> >>> Mike >> >>> >> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >> >>>> >> >>>> I have considered writing JavaScript code to bridge two calls >> >>>> together. >> >>> However, I would like to perform custom handling of the 302 Moved >> >>> Temporarily response. How do I handle the 302 Moved Temporarily >> >>> response if >> >>> I use JavaScript? >> >>>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>> users >> >>> http://www.freeswitch.org >> >>> >> >> -------------- next part -------------- >> >> An HTML attachment was scrubbed... >> >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html >> >> >> >> ------------------------------ >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >> users >> >> http://www.freeswitch.org >> >> >> >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 >> >> ************************************************* >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > users >> > http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/368005c3/attachment-0001.html From mrene_lists at avgs.ca Wed Nov 25 13:01:27 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 25 Nov 2009 16:01:27 -0500 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: You can read all about the sip library at http://sofia-sip.sourceforge.net/refdocs/ Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 25-Nov-09, at 3:58 PM, srinivasula reddy wrote: > thanks for your reply mike, > is there any api in freeswitch or any thing else to update lib > programatically from pjsua. > > srinivas > > On Thu, Nov 26, 2009 at 2:05 AM, Michael Jerris > wrote: > "something that is not available in that lib at this time." > > Mike > > On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote > >> can please tell me how can i exchange session state into sip library. >> >> Thanks >> srinivas >> >> On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris >> wrote: >> For that you would need to fully exchange session state into the >> sip library, something that is not available in that lib at this >> time. >> >> >> On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: >> >>> HI, >>> thanks for your reply, my requirement is i am doing failover stuff >>> with freeswitch. i dont want cut the calls when freeswitch dies, >>> when failover happens mean one freeswitch dies we are going to >>> start the second freeswitch, i dont want close call intiated by >>> the first freeswtich, they are communicating with meida(bypass >>> media). when one endpoing try to end the call at that time i want >>> to close the call for the other end also. >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Srinivasula Reddy K >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ce968534/attachment.html From anthony.minessale at gmail.com Wed Nov 25 13:10:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 15:10:57 -0600 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: <191c3a030911251310h9f8bd1epf0d445c746e968a5@mail.gmail.com> I can spare you the pain and let you know outright that this sort of functionality will cost somewhere in the range of 125,000.00 to 150,000.00 to properly implement by assembling a team of consultants including members of the development team from both FreeSWITCH and Sofia-SIP and even if you have the money, finding the time to implement it would also be a factor as it's a few thousand man-hours of work. On Wed, Nov 25, 2009 at 3:01 PM, Mathieu Rene wrote: > You can read all about the sip library at > http://sofia-sip.sourceforge.net/refdocs/ > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 25-Nov-09, at 3:58 PM, srinivasula reddy wrote: > > thanks for your reply mike, > is there any api in freeswitch or any thing else to update lib > programatically from pjsua. > > srinivas > > On Thu, Nov 26, 2009 at 2:05 AM, Michael Jerris wrote: > >> "something that is not available in that lib at this time." >> >> Mike >> >> On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote >> >> can please tell me how can i exchange session state into sip library. >> >> Thanks >> srinivas >> >> On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: >> >>> For that you would need to fully exchange session state into the sip >>> library, *something that is not available in that lib at this time.* >>> >>> >>> On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: >>> >>> HI, >>> thanks for your reply, my requirement is i am doing failover stuff with >>> freeswitch. i dont want cut the calls when freeswitch dies, when failover >>> happens mean one freeswitch dies we are going to start the second >>> freeswitch, i dont want close call intiated by the first freeswtich, they >>> are communicating with meida(bypass media). when one endpoing try to end the >>> call at that time i want to close the call for the other end also. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Srinivasula Reddy K >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/93b12dba/attachment.html From anthony.minessale at gmail.com Wed Nov 25 13:19:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 15:19:56 -0600 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> Message-ID: <191c3a030911251319g60cdd5a3t33a82a560faf7a2b@mail.gmail.com> The processor power saved is negligible between PCMU and raw PCM and not worth the fuss. If you didn't decode the audio first you would not be able to mix the stream to produce a single file. So if we went to the trouble of making native media bugs to be able to do that you could barely use them so it would not be worth the 5k or more bounty to develop that functionality. On Sun, Nov 22, 2009 at 12:48 PM, Matthew Fong wrote: > I'm trying to conserve processor power by recording in native file format, > PCMU in my case. It works great with the following line > > session:execute("record", > "/tmp/my_recording."..session:getVariable("read_codec")); > > however it fails to work with > > session:execute("record_session", > "/tmp/my_recording."..session:getVariable("read_codec")); > or > record = api:execute("sched_api", '+1 none uuid_record > '..session:getVariable("uuid")..' start > /tmp/my_recording.'..session:getVariable("read_codec")); > > Why is it that it works with record, but not with record_session or > uuid_record? Is there something I'm over looking? In the latter two the > consul reports > > 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File > [/tmp/my_recording.PCMU] 8000hz > > as if it's recording, but /tmp/my_recording.PCMU never shows up. However if > I change it to .wav instead of .PCMU it works. Any ideas? > > --matt > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/a49b5d12/attachment-0001.html From lists at redbonez.net Wed Nov 25 14:18:34 2009 From: lists at redbonez.net (Adam Ford) Date: Wed, 25 Nov 2009 15:18:34 -0700 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> Message-ID: <01cb01ca6e1d$3c289540$b479bfc0$@net> Samuel, FreeSWITCH has a Skype module that uses Skype client instances to connect to the Skype network, you can read about it at http://wiki.freeswitch.org/wiki/Skypiax As far as an official Skype module for non-Asterisk PBX-es, it looks like it is in beta right now - http://www.skype.com/business/products/pbx-systems/sip/ -AF -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Samuel Mukoti Sent: Wednesday, November 25, 2009 1:17 PM Cc: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Grandstream gateways Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews at sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org > wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ 288d63a0/attachment-0001.html >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> In trunk there is a sofia profile setting to allow dialplan >> processing of 302 responses. This won't get you back into your same >> javascript, but you can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. However, I would like to perform custom handling of the >>> 302 Moved Temporarily response. How do I handle the 302 Moved >>> Temporarily response if I use JavaScript? >>> >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Wed, 25 Nov 2009 11:46:05 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> proxy. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes >> >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >>> Try an alias on the sip profile. >>> >>> Mike >> >> >> >> >> ------------------------------ >> >> Message: 4 >> Date: Wed, 25 Nov 2009 12:47:37 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> It's possible it does not. I just added some code to set it on auto- >> adjust so it might be there sometimes now. You might need to add >> some code in mod_sofia to add it other times. Maybe it makes sense >> to move that var setting down to switch_rtp.c. Patches for this >> would be welcome. >> >> Thanks >> >> Mike >> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: >> >>> Hi, >>> >>> In the case of proxy_media=true, does it gets set at all then? >> >> >> >> >> ------------------------------ >> >> Message: 5 >> Date: Wed, 25 Nov 2009 12:48:39 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] How to find whether the destination >> extension supports encryption >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> >> Content-Type: text/plain; charset=us-ascii >> >> You can send the call with secure enabled and if it supports it it >> will use it. >> >> Mike >> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: >> >>> Hello, >>> >>> We have a mix of phones that support RTP encryption and those that >>> do not. I have to support both types in the meanwhile, and would >>> like to have encryption enabled on the relevant leg, even if the >>> other leg does not support it (why? one of our ATAs either must >>> have it unencrypted or have it encrypted, but cannot have both). >>> >>> How do I find whether the destination supports encryption? I do not >>> want to manage an additional table in the database... >>> >> >> >> >> ------------------------------ >> >> Message: 6 >> Date: Wed, 25 Nov 2009 23:25:01 +0530 >> From: srinivasula reddy >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> >> Content-Type: text/plain; charset="iso-8859-1" >> >> HI, >> thanks for your reply, my requirement is i am doing failover stuff >> with >> freeswitch. i dont want cut the calls when freeswitch dies, when >> failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the first >> freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to >> end the >> call at that time i want to close the call for the other end also. >> >> >> srinivas >> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris >> wrote: >> >>> FreeSWITCH will kill the calls when you shut it down, if you >>> intentionally >>> kill the network without shutting down FreeSWITCH the only thing >>> you can do >>> is enable session timers or rtp timers in the soft phones to kill >>> the call >>> when FreeSWITCH dies or when the call is over. >>> >>> Mike >>> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >>> >>>> Hi All, >>>> >>>> goodmorning to all, i have a scenario, two pjsua clients are >>>> connected >>> with Freeswitch and they are in call and bypass_media=true. i >>> close the >>> Freeswitch server, still they are in call, again i started the >>> Freeswitch, >>> and registerd these two endpoints, now how can i end the call >>> (estabilished >>> by the first Freeswitch)? if i call re_invite will it estabilish >>> the call >>> between two endpoints? >>>> any idea? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Srinivasula Reddy K >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ ec246f47/attachment-0001.html >> >> ------------------------------ >> >> Message: 7 >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> From: Stephen Crosby >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Surprisingly, I've found no way to access the HTTP response status >> code >> using mod_spidermonkey_curl. I'd love to see this feature added or >> discussed >> if it already exists and I'm missing it. >> >> --Stephen >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ b8ea2be6/attachment-0001.html >> >> ------------------------------ >> >> Message: 8 >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> From: Tihomir Culjaga >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> this is how i do it from the dialplan: >> >> >> >> >> >> > expression="^(300030)(.*)|^\+(300030)(.*)"> >> >> >> >> >> > data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> >> > data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: >> 1:32} : >> ${caller_id_number})}"/> >> >> > data="aPfx=${caller_id_number:0:6}"/> >> > data="aNum=${caller_id_number:6:16}"/> >> > data="IP_ADDR=${network_addr}:5060"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ 638a2202/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 >> ************************************************* > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mike at jerris.com Wed Nov 25 14:38:18 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 17:38:18 -0500 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <01cb01ca6e1d$3c289540$b479bfc0$@net> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <01cb01ca6e1d$3c289540$b479bfc0$@net> Message-ID: <2ABF7CA4-5FBF-4E7D-8BE3-6D0C92717C92@jerris.com> On Nov 25, 2009, at 5:18 PM, Adam Ford wrote: > Samuel, > > FreeSWITCH has a Skype module that uses Skype client instances to connect to > the Skype network, you can read about it at > http://wiki.freeswitch.org/wiki/Skypiax > > As far as an official Skype module for non-Asterisk PBX-es, it looks like it > is in beta right now - > http://www.skype.com/business/products/pbx-systems/sip/ > > -AF If by in beta you mean they turned off all the servers the beta testers could talk to, then yes, it is indeed. Mike From anthony.minessale at gmail.com Wed Nov 25 14:40:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 16:40:45 -0600 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <01cb01ca6e1d$3c289540$b479bfc0$@net> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <01cb01ca6e1d$3c289540$b479bfc0$@net> Message-ID: <191c3a030911251440n46ec3ee8h92d1305b1542fc0@mail.gmail.com> Skype for SIP is just a SIP account you can get from skype that is somehow tied to skype, probably using the scary 2 year commerical endeavor to make skype work in asterisk. We should all thank Giovanni Maruzzelli for giving us a free solution. On Wed, Nov 25, 2009 at 4:18 PM, Adam Ford wrote: > Samuel, > > FreeSWITCH has a Skype module that uses Skype client instances to connect > to > the Skype network, you can read about it at > http://wiki.freeswitch.org/wiki/Skypiax > > As far as an official Skype module for non-Asterisk PBX-es, it looks like > it > is in beta right now - > http://www.skype.com/business/products/pbx-systems/sip/ > > -AF > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Samuel > Mukoti > Sent: Wednesday, November 25, 2009 1:17 PM > Cc: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Grandstream gateways > > Thank you for those tips, > > I do have some small setups using gxw4108 they work or, except CID > doesn't seem to work. I will try the channel bank route - just don't > know too much about the setup options or how you'd purchase the > correct config, eg. For 150 FXS channel bank, can I get a single PCI > card for that? > > I may end up using the grandstream fxs gateways then use the T1 > channel bank from sangoma, > > Thank you all.. > > Lastly, I know asterisk now has an offical skype_ module, Is there > anything similar I could use? > > > On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: > > > Samuel - you could go with FXS gateways or channel banks. If you go > > the gateway route Grandstream or Audiocodes would work fine. > > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > > you could use a 24FXO channel bank cross connected with Rhino T1 > > cards, or individual FXO gateways but you may have a hard time > > finding 24 ports of FXO in a single GW. Best performing T1 cards in > > my experience (thousands of deployments) are Sangoma. Your server > > configuration looks fine. > > > > Cory J. Andrews > > Director New Market Initiatives > > > > Sayers Media Group > > VoIP Supply, LLC > > 454 Sonwil Drive > > Buffalo, NY 14225 > > 716-250-3402 OFFICE > > 716-630-1548 FAX > > 716-601-4474 MOBILE > > candrews at sayersmedia.com > > > > > > Have I exceeded your expectations? Please share your experience > > with my boss, Benjamin P. Sayers, CEO > > > > NOTICE: The information contained in this email and any document > > attached hereto is intended only for the named recipient(s). It is > > the property of the VoIP Supply, LLC and shall not be used, > > disclosed or reproduced without the express written consent of VoIP > > Supply, LLC. If you are not the intended recipient, nor the employee > > or agent responsible for delivering this message in confidence to > > the intended recipient(s), you are hereby notified that you have > > received this transmittal in error, and any review, dissemination, > > distribution or copying of this transmittal or its attachments is > > strictly prohibited. If you have received this transmittal and/or > > attachments in error, please notify me immediately by reply e-mail > > or telephone and then delete this message, including any > > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > > 14225 USA. > > > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Samuel Mukoti > > Sent: Wednesday, November 25, 2009 2:40 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Grandstream gateways > > > > Hi all, > > > > I'm wanting to try out a my first large scale setup at the office, 200 > > extensions and 24 POTS incoming, also a T1 line once the telco guys > > are ready. I wanted assistance with choosing the most appropriate > > hardware. We already have about 150 analogue phones, and I was > > wondering what's best? A couple of grandstream FXS GXW4024? Also for > > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > > or digium card? The best voice quality is paramount. Lastly for T1 > > what cards are recommeded, > > > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > > would that perform? Or do I need hardware transcoding? > > > > Thank you, > > > > Sam > > > > Twitter: twitter.com/samuelmukoti > > > > > > On 25 Nov,2009, at 8:05 PM, > freeswitch-users-request at lists.freeswitch.org > > wrote: > > > >> Send FreeSWITCH-users mailing list submissions to > >> freeswitch-users at lists.freeswitch.org > >> > >> To subscribe or unsubscribe via the World Wide Web, visit > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> or, via email, send a message with subject or body 'help' to > >> freeswitch-users-request at lists.freeswitch.org > >> > >> You can reach the person managing the list at > >> freeswitch-users-owner at lists.freeswitch.org > >> > >> When replying, please edit your Subject line so it is more specific > >> than "Re: Contents of FreeSWITCH-users digest..." > >> > >> > >> Today's Topics: > >> > >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) > >> 2. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Michael Jerris) > >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) > >> 4. Re: remote_media_ip variable not set (Michael Jerris) > >> 5. Re: How to find whether the destination extension supports > >> encryption (Michael Jerris) > >> 6. Re: Bypass_media and re_invite (srinivasula reddy) > >> 7. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Stephen Crosby) > >> 8. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Tihomir Culjaga) > >> > >> > >> --- > >> ------------------------------------------------------------------- > >> > >> Message: 1 > >> Date: Wed, 25 Nov 2009 12:44:46 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] mod_conference kick to abort > >> invitations > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> > >> Content-Type: text/plain; charset="windows-1252" > >> > >> Its a feature we don't have, patches welcome. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > >> > >>> Hi members, > >>> I?m controlling freeswitch with the conference module via xmlrpc. > >>> > >>> Is it desired that the kick command can only kick users that are > >>> connected to the conference? > >>> Is there no chance abort an invitation? > >>> The kick command has no effect until the person I invited with the > >>> dial command is connected. > >> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ > 288d63a0/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 2 > >> Date: Wed, 25 Nov 2009 12:45:50 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii > >> > >> In trunk there is a sofia profile setting to allow dialplan > >> processing of 302 responses. This won't get you back into your same > >> javascript, but you can probably do something clever from there. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> > >>> > >>> I have considered writing JavaScript code to bridge two calls > >>> together. However, I would like to perform custom handling of the > >>> 302 Moved Temporarily response. How do I handle the 302 Moved > >>> Temporarily response if I use JavaScript? > >>> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 3 > >> Date: Wed, 25 Nov 2009 11:46:05 -0600 > >> From: Brian West > >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via > >> proxy. > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> > >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes > >> > >> Yes an alias will be required for every domain you run on the profile > >> so it can find it. > >> > >> /b > >> > >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > >> > >>> Try an alias on the sip profile. > >>> > >>> Mike > >> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 4 > >> Date: Wed, 25 Nov 2009 12:47:37 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii > >> > >> It's possible it does not. I just added some code to set it on auto- > >> adjust so it might be there sometimes now. You might need to add > >> some code in mod_sofia to add it other times. Maybe it makes sense > >> to move that var setting down to switch_rtp.c. Patches for this > >> would be welcome. > >> > >> Thanks > >> > >> Mike > >> > >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > >> > >>> Hi, > >>> > >>> In the case of proxy_media=true, does it gets set at all then? > >> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 5 > >> Date: Wed, 25 Nov 2009 12:48:39 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] How to find whether the destination > >> extension supports encryption > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> > >> Content-Type: text/plain; charset=us-ascii > >> > >> You can send the call with secure enabled and if it supports it it > >> will use it. > >> > >> Mike > >> > >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > >> > >>> Hello, > >>> > >>> We have a mix of phones that support RTP encryption and those that > >>> do not. I have to support both types in the meanwhile, and would > >>> like to have encryption enabled on the relevant leg, even if the > >>> other leg does not support it (why? one of our ATAs either must > >>> have it unencrypted or have it encrypted, but cannot have both). > >>> > >>> How do I find whether the destination supports encryption? I do not > >>> want to manage an additional table in the database... > >>> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 6 > >> Date: Wed, 25 Nov 2009 23:25:01 +0530 > >> From: srinivasula reddy > >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> > >> Content-Type: text/plain; charset="iso-8859-1" > >> > >> HI, > >> thanks for your reply, my requirement is i am doing failover stuff > >> with > >> freeswitch. i dont want cut the calls when freeswitch dies, when > >> failover > >> happens mean one freeswitch dies we are going to start the second > >> freeswitch, i dont want close call intiated by the first > >> freeswtich, they > >> are communicating with meida(bypass media). when one endpoing try to > >> end the > >> call at that time i want to close the call for the other end also. > >> > >> > >> srinivas > >> > >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris > >> wrote: > >> > >>> FreeSWITCH will kill the calls when you shut it down, if you > >>> intentionally > >>> kill the network without shutting down FreeSWITCH the only thing > >>> you can do > >>> is enable session timers or rtp timers in the soft phones to kill > >>> the call > >>> when FreeSWITCH dies or when the call is over. > >>> > >>> Mike > >>> > >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > >>> > >>>> Hi All, > >>>> > >>>> goodmorning to all, i have a scenario, two pjsua clients are > >>>> connected > >>> with Freeswitch and they are in call and bypass_media=true. i > >>> close the > >>> Freeswitch server, still they are in call, again i started the > >>> Freeswitch, > >>> and registerd these two endpoints, now how can i end the call > >>> (estabilished > >>> by the first Freeswitch)? if i call re_invite will it estabilish > >>> the call > >>> between two endpoints? > >>>> any idea? > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Srinivasula Reddy K > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ > ec246f47/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 7 > >> Date: Wed, 25 Nov 2009 10:01:14 -0800 > >> From: Stephen Crosby > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> > >> Content-Type: text/plain; charset="utf-8" > >> > >> Surprisingly, I've found no way to access the HTTP response status > >> code > >> using mod_spidermonkey_curl. I'd love to see this feature added or > >> discussed > >> if it already exists and I'm missing it. > >> > >> --Stephen > >> > >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris > >> wrote: > >> > >>> In trunk there is a sofia profile setting to allow dialplan > >>> processing of > >>> 302 responses. This won't get you back into your same javascript, > >>> but you > >>> can probably do something clever from there. > >>> > >>> Mike > >>> > >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >>> > >>>> > >>>> I have considered writing JavaScript code to bridge two calls > >>>> together. > >>> However, I would like to perform custom handling of the 302 Moved > >>> Temporarily response. How do I handle the 302 Moved Temporarily > >>> response if > >>> I use JavaScript? > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ > b8ea2be6/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 8 > >> Date: Wed, 25 Nov 2009 19:04:56 +0100 > >> From: Tihomir Culjaga > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> > >> Content-Type: text/plain; charset="iso-8859-1" > >> > >> this is how i do it from the dialplan: > >> > >> > >> > >> > >> > >> >> expression="^(300030)(.*)|^\+(300030)(.*)"> > >> > >> > >> > >> > >> >> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> > >> >> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: > >> 1:32} : > >> ${caller_id_number})}"/> > >> > >> >> data="aPfx=${caller_id_number:0:6}"/> > >> >> data="aNum=${caller_id_number:6:16}"/> > >> >> data="IP_ADDR=${network_addr}:5060"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris > >> wrote: > >> > >>> In trunk there is a sofia profile setting to allow dialplan > >>> processing of > >>> 302 responses. This won't get you back into your same javascript, > >>> but you > >>> can probably do something clever from there. > >>> > >>> Mike > >>> > >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >>> > >>>> > >>>> I have considered writing JavaScript code to bridge two calls > >>>> together. > >>> However, I would like to perform custom handling of the 302 Moved > >>> Temporarily response. How do I handle the 302 Moved Temporarily > >>> response if > >>> I use JavaScript? > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ > 638a2202/attachment.html > >> > >> ------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > >> > >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 > >> ************************************************* > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/7800dfaa/attachment-0001.html From anthony.minessale at gmail.com Wed Nov 25 14:58:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 16:58:17 -0600 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <2ABF7CA4-5FBF-4E7D-8BE3-6D0C92717C92@jerris.com> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <01cb01ca6e1d$3c289540$b479bfc0$@net> <2ABF7CA4-5FBF-4E7D-8BE3-6D0C92717C92@jerris.com> Message-ID: <191c3a030911251458t5475dfc9lb9665878de91b8c9@mail.gmail.com> exactly On Wed, Nov 25, 2009 at 4:38 PM, Michael Jerris wrote: > > On Nov 25, 2009, at 5:18 PM, Adam Ford wrote: > > > Samuel, > > > > FreeSWITCH has a Skype module that uses Skype client instances to connect > to > > the Skype network, you can read about it at > > http://wiki.freeswitch.org/wiki/Skypiax > > > > As far as an official Skype module for non-Asterisk PBX-es, it looks like > it > > is in beta right now - > > http://www.skype.com/business/products/pbx-systems/sip/ > > > > -AF > > If by in beta you mean they turned off all the servers the beta testers > could talk to, then yes, it is indeed. > > Mike > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/12c9df3f/attachment.html From anthony.minessale at gmail.com Wed Nov 25 15:10:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 17:10:04 -0600 Subject: [Freeswitch-users] remote_media_ip variable not set In-Reply-To: References: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> <2F929FDB-0E1B-49E0-A1E7-F4F1E2D548AD@avgs.ca> <27c25bc40911240756k7842c80kd75be2d3d93441b9@mail.gmail.com> Message-ID: <191c3a030911251510s6dad4526i7cc6e64925b8153a@mail.gmail.com> I added a patch to do it in more places On Wed, Nov 25, 2009 at 11:47 AM, Michael Jerris wrote: > It's possible it does not. I just added some code to set it on auto-adjust > so it might be there sometimes now. You might need to add some code in > mod_sofia to add it other times. Maybe it makes sense to move that var > setting down to switch_rtp.c. Patches for this would be welcome. > > Thanks > > Mike > > On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > > > Hi, > > > > In the case of proxy_media=true, does it gets set at all then? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/85b9b0aa/attachment.html From john_platts at hotmail.com Wed Nov 25 15:21:51 2009 From: john_platts at hotmail.com (John Platts) Date: Wed, 25 Nov 2009 17:21:51 -0600 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: , Message-ID: How do I turn on dialplan processing of 302 responses? I can solve my problem if I can process 302 responses in my dialplan. ---------------------------------------- > From: mike at jerris.com > Date: Wed, 25 Nov 2009 12:45:50 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript > > In trunk there is a sofia profile setting to allow dialplan processing of 302 responses. This won't get you back into your same javascript, but you can probably do something clever from there. > > Mike > > On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> >> I have considered writing JavaScript code to bridge two calls together. However, I would like to perform custom handling of the 302 Moved Temporarily response. How do I handle the 302 Moved Temporarily response if I use JavaScript? >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search?q=restaurants&form=MFESRP&publ=WLHMTAG&crea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 From jmesquita at freeswitch.org Wed Nov 25 17:02:54 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 25 Nov 2009 23:02:54 -0200 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: <8595daf70911250929w26eeb3aboae0f95042f35393b@mail.gmail.com> References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> <8595daf70911250929w26eeb3aboae0f95042f35393b@mail.gmail.com> Message-ID: These guys can on E1, not T1. They are not compatible with FS just yet, but we are working on it. Let me know off-list if you are interested. JM On Wed, Nov 25, 2009 at 3:29 PM, Imthiyaz Ahmed wrote: > I mean to tap tx and rx of a PRI line using sangoma tap and record > the call information and actual calls without distrubing the existing > line . freeswitch will work in passive mode like trunk side call > recorder. > > Thanks > Imthiyaz > > > On Wed, Nov 25, 2009 at 10:43 PM, Anthony Minessale > wrote: > > What do you mean by passive encoding? > > > > On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed > > wrote: > >> > >> hi > >> is it possibe to enable passive recording in sangoma tdm interface > >> in feeswich. pls advice > >> Best Regards > >> G.Imthiyaz Ahmed > >> > >> > >> > >> -- > >> Best Regards > >> G.Imthiyaz Ahmed > >> PeopleTech systems (P) ltd > >> http://peopletech.co.in > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Best Regards > G.Imthiyaz Ahmed > PeopleTech systems (P) ltd > http://peopletech.co.in > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/13422ae4/attachment.html From dujinfang at gmail.com Wed Nov 25 17:35:08 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 26 Nov 2009 09:35:08 +0800 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <191c3a030911251319g60cdd5a3t33a82a560faf7a2b@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <191c3a030911251319g60cdd5a3t33a82a560faf7a2b@mail.gmail.com> Message-ID: <23f91030911251735r3215a344h279a3f8589d5ff85@mail.gmail.com> Yeah, that's why I had to record to two files(read&write) and need to mix together by using sox. Do you only try to using PCMU to save CPU power matt? As Anthony said, the difference can be ignored. And you also need to take extra effort to make sure transcoding will not happen on a conversation. But it maybe useful for expensive codecs like g729, iLBC, speex etc for recording heavy scenarios. I'd like to take a look if there is a 5k bounty ;) 2009/11/26 Anthony Minessale > The processor power saved is negligible between PCMU and raw PCM and not > worth the fuss. > If you didn't decode the audio first you would not be able to mix the > stream to produce a single file. > So if we went to the trouble of making native media bugs to be able to do > that you could barely use them so it would not be worth the 5k or more > bounty to develop that functionality. > > > > On Sun, Nov 22, 2009 at 12:48 PM, Matthew Fong wrote: > >> I'm trying to conserve processor power by recording in native file format, >> PCMU in my case. It works great with the following line >> >> session:execute("record", >> "/tmp/my_recording."..session:getVariable("read_codec")); >> >> however it fails to work with >> >> session:execute("record_session", >> "/tmp/my_recording."..session:getVariable("read_codec")); >> or >> record = api:execute("sched_api", '+1 none uuid_record >> '..session:getVariable("uuid")..' start >> /tmp/my_recording.'..session:getVariable("read_codec")); >> >> Why is it that it works with record, but not with record_session or >> uuid_record? Is there something I'm over looking? In the latter two the >> consul reports >> >> 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File >> [/tmp/my_recording.PCMU] 8000hz >> >> as if it's recording, but /tmp/my_recording.PCMU never shows up. However >> if I change it to .wav instead of .PCMU it works. Any ideas? >> >> --matt >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/52f891bf/attachment-0001.html From josh at radianttiger.com Wed Nov 25 17:38:12 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 25 Nov 2009 17:38:12 -0800 Subject: [Freeswitch-users] ESL command completion Message-ID: Is there a way of determining if a call-command sent to a session via ESL has completed? Is there a return event which is always fired? Is there a identifier I can use to verify that the return event matches my command? Thanks, Josh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ccda899c/attachment.html From mike at jerris.com Wed Nov 25 18:08:59 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 21:08:59 -0500 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: , Message-ID: <5AB26AE2-2ABC-4D46-B61A-675B1390553F@jerris.com> from http://svn.freeswitch.org/svn/freeswitch/trunk/conf/sip_profiles/internal.xml It appears this never made the wiki, could someone please get it on there. Thanks Mike On Nov 25, 2009, at 6:21 PM, John Platts wrote: > > How do I turn on dialplan processing of 302 responses? I can solve my problem if I can process 302 responses in my dialplan. > > ---------------------------------------- >> From: mike at jerris.com >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript >> >> In trunk there is a sofia profile setting to allow dialplan processing of 302 responses. This won't get you back into your same javascript, but you can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls together. However, I would like to perform custom handling of the 302 Moved Temporarily response. How do I handle the 302 Moved Temporarily response if I use JavaScript? >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________ > Bing brings you maps, menus, and reviews organized in one place. > http://www.bing.com/search?q=restaurants&form=MFESRP&publ=WLHMTAG&crea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From josh at radianttiger.com Wed Nov 25 18:10:16 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 25 Nov 2009 18:10:16 -0800 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> Message-ID: Carlos, Do you have any documentation or scripts for your builds? I'm interested in having a working automated build and installer build process, and I'm curious if there's any work you've done that can make my job easier. :) Thanks, Josh On Wed, Nov 4, 2009 at 6:51 AM, Carlos Talbot wrote: > > I usually try to update the svn file at least once a month. I have a new > version ready that was compiled last night but am ironing out login issues > with the FS dudes for upload access. Also, the SVN snapshot now includes > binaries for 32 and 64 bit. It no longer includes flite though as the > install file was approaching 80MB in size. I will revisit this later if > others feel it important to include flite. > >> >> You mentioned FreePBX V3. I had been fumbling around trying to work out >> what >> this is and from what I've read, it seems to provide a GUI Front End for >> configuring FreeSwitch ? >> > Yes, it's still in development phase and as such not ready for production > use. > >> >> I am guessing that while it has been installed with FreeSwitch, I then >> need >> to run the FreePBX Installer to update the FreePBX/FreeSwitch >> configuration >> on my hardware ? >> >> >> When I start FreeSwitch, it does not automatically load the WAMPServer. >> >> Freeswitch and WAMPServer are independant of each other. WAMPServer is > bundled in this install for the purpose of FreePBX as MySQL, Apache and PHP > are all required components of FreePBX. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a >> web >> browser, I can see the WampServer logo and various tools such as phpinfo() >> and phpmyadmin. FreePBX is there under Your Projects. >> >> If you want to configure FreePBX you need to click on the FreePBX.url > shortcut that gets created on your desktop. > > >> When I opened this up the first time, it appeared to want to install >> FreePBX >> over FreeSwitch, I tried to abort this when it was going to overwrite some >> FreeSwitch conf files and I thought I'd better not go on until I had a >> better idea what was happening. I backed out of the FreePBX install and >> now >> I can't get the FreePBX or phpmyadmin pages up again (missing files) so it >> looks like I'm going to have to reinstall anyway. >> >> So, for next time,am I right in thinking that I should proceed with >> running >> the FreePBX install from the WAMPServer menu ? >> > > No, launch it from the shortcut as stated above. Unfortunately, at this > time there is very little user documentation on configuring FreePBX. Here is > the link to the developer's info: http://www.freepbx.org/v3 > > regards, > > Carlos > >> >> >> ----- Original Message ----- >> From: "Jeff Lenk" >> To: >> Sent: Tuesday, November 03, 2009 2:48 PM >> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >> >> >> > >> > Hi Dave, >> > >> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >> > >> > Just as you said freeswitch-1.0.4.exe is the tagged release and >> > freeswitch.exe is a newer svn snapshot. >> > >> > There should be no problems installing the new version allthough best to >> > just try and see! >> > >> > Not sure why the newest one is from October 7th. >> > >> > Jeff >> > >> > >> > Dave Stevenson wrote: >> >> >> >> Hi, >> >> >> >> I have read the Docs on the Wiki >> >> ( >> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> >> but am still not sure of what the different Windows install files are. >> >> Currently, the Windows Installer directory contains :- >> >> >> >> LATEST_SVN_15106 - 6 Bytes >> >> >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> >> >> freeswitch.exe - 32 Megabytes >> >> >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> >> September. The freeswitch.exe file is dated 7th October and think that >> it >> >> contains the minor updates since 3rd September ? >> >> >> >> Could someone who knows FreeSwitch under windows help me understand the >> >> two files please ? >> >> >> >> I chickened out of running the later exe in case it did something to >> the >> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >> with >> >> the old one already installed ? >> >> What will it actually do ? >> >> >> >> regards >> >> Dave >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > -- >> > View this message in context: >> > >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/f9ff8d51/attachment.html From mike at jerris.com Wed Nov 25 18:11:31 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 21:11:31 -0500 Subject: [Freeswitch-users] ESL command completion In-Reply-To: References: Message-ID: <34EF9B51-9B0A-45A9-A269-17E49D597BA1@jerris.com> There are execute_complete events. I can't recall everything that is in them but they should always be fired. Mike On Nov 25, 2009, at 8:38 PM, Josh Rivers wrote: > Is there a way of determining if a call-command sent to a session via ESL has completed? Is there a return event which is always fired? Is there a identifier I can use to verify that the return event matches my command? > > Thanks, > Josh > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From carlos.talbot at gmail.com Wed Nov 25 20:37:30 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 25 Nov 2009 22:37:30 -0600 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> Message-ID: <5800526b0911252037l5cf974cbie9542b40316ebf37@mail.gmail.com> I've just checked in the source files for the Inno Setup script I'm using to build the windows installer(svn 15681). That's about the extent of the documentation at this point. :) regards, Carlos On Wed, Nov 25, 2009 at 8:10 PM, Josh Rivers wrote: > Carlos, > > Do you have any documentation or scripts for your builds? I'm interested in > having a working automated build and installer build process, and I'm > curious if there's any work you've done that can make my job easier. :) > > Thanks, > Josh > > > On Wed, Nov 4, 2009 at 6:51 AM, Carlos Talbot wrote: > >> >> I usually try to update the svn file at least once a month. I have a new >> version ready that was compiled last night but am ironing out login issues >> with the FS dudes for upload access. Also, the SVN snapshot now includes >> binaries for 32 and 64 bit. It no longer includes flite though as the >> install file was approaching 80MB in size. I will revisit this later if >> others feel it important to include flite. >> >>> >>> You mentioned FreePBX V3. I had been fumbling around trying to work out >>> what >>> this is and from what I've read, it seems to provide a GUI Front End for >>> configuring FreeSwitch ? >>> >> Yes, it's still in development phase and as such not ready for production >> use. >> >>> >>> I am guessing that while it has been installed with FreeSwitch, I then >>> need >>> to run the FreePBX Installer to update the FreePBX/FreeSwitch >>> configuration >>> on my hardware ? >>> >>> >>> When I start FreeSwitch, it does not automatically load the WAMPServer. >>> >>> Freeswitch and WAMPServer are independant of each other. WAMPServer is >> bundled in this install for the purpose of FreePBX as MySQL, Apache and PHP >> are all required components of FreePBX. >> >> When I start WAMPServer manually, and open up localhost (127.0.0.1) in a >>> web >>> browser, I can see the WampServer logo and various tools such as >>> phpinfo() >>> and phpmyadmin. FreePBX is there under Your Projects. >>> >>> If you want to configure FreePBX you need to click on the FreePBX.url >> shortcut that gets created on your desktop. >> >> >>> When I opened this up the first time, it appeared to want to install >>> FreePBX >>> over FreeSwitch, I tried to abort this when it was going to overwrite >>> some >>> FreeSwitch conf files and I thought I'd better not go on until I had a >>> better idea what was happening. I backed out of the FreePBX install and >>> now >>> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >>> it >>> looks like I'm going to have to reinstall anyway. >>> >>> So, for next time,am I right in thinking that I should proceed with >>> running >>> the FreePBX install from the WAMPServer menu ? >>> >> >> No, launch it from the shortcut as stated above. Unfortunately, at this >> time there is very little user documentation on configuring FreePBX. Here is >> the link to the developer's info: http://www.freepbx.org/v3 >> >> regards, >> >> Carlos >> >>> >>> >>> ----- Original Message ----- >>> From: "Jeff Lenk" >>> To: >>> Sent: Tuesday, November 03, 2009 2:48 PM >>> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >>> >>> >>> > >>> > Hi Dave, >>> > >>> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >>> > >>> > Just as you said freeswitch-1.0.4.exe is the tagged release and >>> > freeswitch.exe is a newer svn snapshot. >>> > >>> > There should be no problems installing the new version allthough best >>> to >>> > just try and see! >>> > >>> > Not sure why the newest one is from October 7th. >>> > >>> > Jeff >>> > >>> > >>> > Dave Stevenson wrote: >>> >> >>> >> Hi, >>> >> >>> >> I have read the Docs on the Wiki >>> >> ( >>> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >>> >> but am still not sure of what the different Windows install files are. >>> >> Currently, the Windows Installer directory contains :- >>> >> >>> >> LATEST_SVN_15106 - 6 Bytes >>> >> >>> >> freeswitch-1.0.4.exe - 42 Megabytes >>> >> >>> >> freeswitch.exe - 32 Megabytes >>> >> >>> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >>> >> September. The freeswitch.exe file is dated 7th October and think that >>> it >>> >> contains the minor updates since 3rd September ? >>> >> >>> >> Could someone who knows FreeSwitch under windows help me understand >>> the >>> >> two files please ? >>> >> >>> >> I chickened out of running the later exe in case it did something to >>> the >>> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >>> with >>> >> the old one already installed ? >>> >> What will it actually do ? >>> >> >>> >> regards >>> >> Dave >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> > >>> > -- >>> > View this message in context: >>> > >>> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >>> > Sent from the freeswitch-users mailing list archive at Nabble.com. >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/fa70bb95/attachment-0001.html From edpimentl at gmail.com Wed Nov 25 20:49:10 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 25 Nov 2009 23:49:10 -0500 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> <8595daf70911250929w26eeb3aboae0f95042f35393b@mail.gmail.com> Message-ID: <9dc4a1670911252049j1ba1a2a8g3fdc8c884f843951@mail.gmail.com> Are you wanting to provide "Lawfull Interecept" functionanility for CALEA Compliance? http://www.netequalizer.com/caleafaq.php -E Gpro.ws edpimentl [SKype ] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/a0c1e462/attachment.html From JCasale at activenetwerx.com Wed Nov 25 21:42:28 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 26 Nov 2009 05:42:28 +0000 Subject: [Freeswitch-users] Faxing Advice Message-ID: I need to make faxing easy for some very computer illiterate folk. I am using an email service and going to use procmail to print anything incoming automatically but they cant get the hang of scanning to an email app, so I am going to buy a Linksys PAP2T as per the wiki. Since the setup will never receive inbound remote faxes, I just need to direct all fax's sent from the FXS port (that extension) to the email script in the wiki substituting the destination # as the alias portion of the email. So if I create a dialplan that catches the caller_id_number of the FXS port, does the $1 variable exist in the following scenario: as that's how our service requires fax's, the 10 digit # at their domain, fax.com. Is this a plausible setup? Lastly, I see in the interop list that Audiocodes Mediapack 114 is supported, but the 202 is not listed, is that simply because its new or is it known to not work? Given that its the same price as the Linksys, I would rather get it. Thanks! jlc From mctch at yahoo.com Wed Nov 25 23:33:06 2009 From: mctch at yahoo.com (Mark Crane) Date: Wed, 25 Nov 2009 23:33:06 -0800 (PST) Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: Message-ID: <221275.23339.qm@web56403.mail.re3.yahoo.com> "how about trying Fusionpbx.com? ( GUI)" -Ram I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was ready to release now but decided to do one more release candidate just to be sure. This should be the last release candidate before the release of version 1.0. The final release may be by the end of the week as long as no major issues are found. http://fusionpbx.com --- On Mon, 11/23/09, ram wrote: From: ram Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX To: freeswitch-users at lists.freeswitch.org Date: Monday, November 23, 2009, 10:54 PM On Mon, Nov 23, 2009 at 10:37 AM, Otis wrote: Thanks. I have to get a centos box I guess. Much appreciated Samuel 'Otis' ? how about trying Fusionpbx.com? ( GUI) ? Ram -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/c6afed1e/attachment.html From lakindia89 at gmail.com Thu Nov 26 01:27:10 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Thu, 26 Nov 2009 14:57:10 +0530 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> Message-ID: <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> Hi, Any help or suggestion regarding my previous post. Especially "I also noted that, if I don't receive any events, especially "SERVER_DISCONNECTED", then the connection is in established state, but once I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it correct??" Here is the program by which I confirmed the above! require ESL; use IO::Socket::INET; my $ip = "192.168.1.222"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; my $con; my $type = "user/"; for(;;) { # wait for any client to connect, a new client will get connected when a new call comes in the dialplan. my $new_sock = $sock->accept(); # Do fork and let the parent to wait for more clients. my $pid = fork(); if ($pid) { close($new_sock); next; } # Extract the host of the client. my $host = $new_sock->sockhost(); # file descriptor for the socket. my $fd = fileno($new_sock); print "Host name is $host\n"; # Create object for the ESL connection package to access the ESL functions. $con = new ESL::ESLconnection($fd); # Gets the info about this channel. my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s to %s\n", $uuid, $info->getHeader("caller-caller-id-number"), $info->getHeader("caller-destination-number"); # Answer the channel. $con->execute("answer"); # Set the event lock to tell the FS to execute the instructions in the given order. $con->setEventLock("true"); # Play a file & Get the personal number from the user. $con->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); $con->execute("hangup"); while($con->connected()) { my $e=$con->recvEvent(); my $ename=$e->getHeader("Event-Name"); print $e->serialize(); print "$ename\n"; print "Connection exists\n"; sleep(1); } print "Bye\n------------------------------------------------------------------\n"; close($new_sock); } I've not registered for any events. In the above program I'm receiving the SERVER_DISCONNECTED event. Output when receiving event: Host name is 192.168.1.222 Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 Event-Name: SERVER_DISCONNECTED SERVER_DISCONNECTED Connection exists Bye When I comment the recvEvent line, I got the following output. Host name is 192.168.1.222 Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 Connection exists Connection exists Connection exists Connection exists Connection exists On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy wrote: > I've tried the following program as per the suggestion that you've told. > But it seems, no success. Once the connection is closed, I created a new > connection and I send originate to originate a new call. But it is not > working. > > require ESL; > use IO::Socket::INET; > use Data::Dumper; > > my $ip = "192.168.1.222"; > my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', > Proto => 'tcp', Listen => 2, Reuse => 1 ); > die "Could not create socket: $!\n" unless $sock; > > my $make_call; > my $con; > my $type = "user/"; > > for(;;) { > my $new_sock = $sock->accept(); > my $pid = fork(); > if ($pid) { > close($new_sock); > next; > } > my $host = $new_sock->sockhost(); > my $fd = fileno($new_sock); > $con = new ESL::ESLconnection($fd); > my $info = $con->getInfo(); > my $uuid = $info->getHeader("unique-id"); > printf "Connected call %s, from %s to %s\n", $uuid, > $info->getHeader("caller-caller-id-number"), > $info->getHeader("caller-destination-number"); > > $con->filter("Unique-Id", $uuid); > $con->events("plain", "all"); > $con->execute("answer"); > $con->setEventLock("true"); > my $number=$con->execute("read","2 4 > /usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav > accnt_number 5000 #"); > while($con->connected()) > { > my $e=$con->recvEvent(); > my $ename=$e->getHeader("Event-Name"); > my $app=$e->getHeader("Application"); > if($ename eq "CHANNEL_EXECUTE_COMPLETE" and $app eq "read") > { > my $num=$e->getHeader("variable_accnt_number"); > print "$num\n"; > $con->execute("hangup"); > } > } > if(!$con->connected()) > { > print "Connection not exists\n"; > $con = new ESL::ESLconnection($fd); > $con->api("originate","user/1000 &park()"); > print "Hai\n"; > } > print > "Bye\n------------------------------------------------------------------\n"; > close($new_sock); > } > Output: > Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097 > 1000 > Connection not exists > Hai > Bye > ------------------------------------------------------------------ > The freeswitch log is in > http://pastebin.freeswitch.org/11258 > > I also noted that, if I don't receive any events, especially > "SERVER_DISCONNECTED", then the connection is in established state, but once > I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it > correct?? > > > > > > On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> or open a new outbound connection at the end of your script so you can >> send your originate command. >> Since the channel hanging up will close your existing connection since >> it's only an outbound single session socket. >> >> >> On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins wrote: >> >>> >>> >>> On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < >>> lakindia89 at gmail.com> wrote: >>> >>>> Hi, >>>> I'm using perl ESL to control the call in freeswitch. >>>> I'm having the following scenario, but not able to get it right. >>>> >>>> Dialplan: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> 1. User A calls to an extention (1000). >>>> 2. My ESL program will be running, and it answers the call. >>>> 3. Then the program will get a number from the user. >>>> 4. It will hangup the call. >>>> 5. The program has to call to the number that was given by the user. >>>> >>>> In the above scenario, I was able to do until the 4th step. After hangup >>>> the call, if I say originate it is not working. >>>> Any ideas on how to do this in ESL. >>>> >>>> >>> I want to make sure I understand what the script is supposed to be doing. >>> The caller will key in a phone number to your script and your script will >>> collect those digits. The script will then hangup on the caller and >>> originate a completely new call? Perhaps you could use sched_api to schedule >>> a new originate command for a few seconds into the future and then hangup? >>> -MC >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/10741c63/attachment-0001.html From mike at jerris.com Thu Nov 26 01:53:57 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Nov 2009 04:53:57 -0500 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> Message-ID: Your using outbound socket and you hangup the call, so it tells you it is done with the server disconnected message and drops the connection. This is all as expected. I guess I don't understand what you think is the problem. This code is doing exactly what I would expect it to do. Mike On Nov 26, 2009, at 4:27 AM, lakshmanan ganapathy wrote: > Hi, Any help or suggestion regarding my previous post. Especially > > "I also noted that, if I don't receive any events, especially > "SERVER_DISCONNECTED", then the connection is in established state, > but once I receive the "SERVER_DISCONNECTED" event, the connection > is closed. Is it correct??" > Here is the program by which I confirmed the above! > > require ESL; > use IO::Socket::INET; > > my $ip = "192.168.1.222"; > my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => > '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); > die "Could not create socket: $!\n" unless $sock; > my $con; > my $type = "user/"; > > for(;;) { > # wait for any client to connect, a new client will get > connected when a new call comes in the dialplan. > my $new_sock = $sock->accept(); > # Do fork and let the parent to wait for more clients. > my $pid = fork(); > if ($pid) { > close($new_sock); > next; > } > # Extract the host of the client. > my $host = $new_sock->sockhost(); > # file descriptor for the socket. > my $fd = fileno($new_sock); > print "Host name is $host\n"; > # Create object for the ESL connection package to access the > ESL functions. > $con = new ESL::ESLconnection($fd); > # Gets the info about this channel. > my $info = $con->getInfo(); > my $uuid = $info->getHeader("unique-id"); > printf "Connected call %s, from %s to %s\n", $uuid, $info- > >getHeader("caller-caller-id-number"), $info->getHeader("caller- > destination-number"); > > # Answer the channel. > $con->execute("answer"); > # Set the event lock to tell the FS to execute the > instructions in the given order. > $con->setEventLock("true"); > # Play a file & Get the personal number from the user. > $con->execute("playback","/usr/local/freeswitch/sounds/en/us/ > callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); > $con->execute("hangup"); > while($con->connected()) > { > my $e=$con->recvEvent(); > my $ename=$e->getHeader("Event-Name"); > print $e->serialize(); > print "$ename\n"; > print "Connection exists\n"; > sleep(1); > } > print "Bye > \n------------------------------------------------------------------ > \n"; > close($new_sock); > } > I've not registered for any events. > In the above program I'm receiving the SERVER_DISCONNECTED event. > Output when receiving event: > Host name is 192.168.1.222 > Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 > to 9097 > Event-Name: SERVER_DISCONNECTED > > SERVER_DISCONNECTED > Connection exists > Bye > > When I comment the recvEvent line, I got the following output. > > Host name is 192.168.1.222 > Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 > to 9097 > Connection exists > Connection exists > Connection exists > Connection exists > Connection exists > > > On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy > wrote: > I've tried the following program as per the suggestion that you've > told. But it seems, no success. Once the connection is closed, I > created a new connection and I send originate to originate a new > call. But it is not working. > > require ESL; > use IO::Socket::INET; > use Data::Dumper; > > my $ip = "192.168.1.222"; > my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => > '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); > die "Could not create socket: $!\n" unless $sock; > > my $make_call; > my $con; > my $type = "user/"; > > for(;;) { > my $new_sock = $sock->accept(); > my $pid = fork(); > if ($pid) { > close($new_sock); > next; > } > my $host = $new_sock->sockhost(); > my $fd = fileno($new_sock); > $con = new ESL::ESLconnection($fd); > my $info = $con->getInfo(); > my $uuid = $info->getHeader("unique-id"); > printf "Connected call %s, from %s to %s\n", $uuid, $info- > >getHeader("caller-caller-id-number"), $info->getHeader("caller- > destination-number"); > > $con->filter("Unique-Id", $uuid); > $con->events("plain", "all"); > $con->execute("answer"); > $con->setEventLock("true"); > my $number=$con->execute("read","2 4 /usr/local/freeswitch/ > sounds/en/us/callie/conference/8000/conf-pin.wav accnt_number 5000 > #"); > while($con->connected()) > { > my $e=$con->recvEvent(); > my $ename=$e->getHeader("Event-Name"); > my $app=$e->getHeader("Application"); > if($ename eq "CHANNEL_EXECUTE_COMPLETE" and $app eq > "read") > { > my $num=$e->getHeader > ("variable_accnt_number"); > print "$num\n"; > $con->execute("hangup"); > } > } > if(!$con->connected()) > { > print "Connection not exists\n"; > $con = new ESL::ESLconnection($fd); > $con->api("originate","user/1000 &park()"); > print "Hai\n"; > } > print "Bye > \n------------------------------------------------------------------ > \n"; > close($new_sock); > } > Output: > Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097 > 1000 > Connection not exists > Hai > Bye > ------------------------------------------------------------------ > The freeswitch log is in > http://pastebin.freeswitch.org/11258 > > I also noted that, if I don't receive any events, especially > "SERVER_DISCONNECTED", then the connection is in established state, > but once I receive the "SERVER_DISCONNECTED" event, the connection > is closed. Is it correct?? > > > > > > On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale > wrote: > or open a new outbound connection at the end of your script so you > can send your originate command. > Since the channel hanging up will close your existing connection > since it's only an outbound single session socket. > > > On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins > wrote: > > > On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy > wrote: > Hi, > I'm using perl ESL to control the call in freeswitch. > I'm having the following scenario, but not able to get it right. > > Dialplan: > > > > > > > > > 1. User A calls to an extention (1000). > 2. My ESL program will be running, and it answers the call. > 3. Then the program will get a number from the user. > 4. It will hangup the call. > 5. The program has to call to the number that was given by the user. > > In the above scenario, I was able to do until the 4th step. After > hangup the call, if I say originate it is not working. > Any ideas on how to do this in ESL. > > > I want to make sure I understand what the script is supposed to be > doing. The caller will key in a phone number to your script and your > script will collect those digits. The script will then hangup on the > caller and originate a completely new call? Perhaps you could use > sched_api to schedule a new originate command for a few seconds into > the future and then hangup? > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/2439d860/attachment.html From abeka at greatiam.com Thu Nov 26 02:53:23 2009 From: abeka at greatiam.com (Otis) Date: Thu, 26 Nov 2009 10:53:23 +0000 Subject: [Freeswitch-users] Requesting testing. In-Reply-To: References: Message-ID: <4B0E5E23.1060804@greatiam.com> Hi Checked out svn checkout y'day. I am in the UK. Installed . Installed on Fedora 11 i386 box. : bootstrap.sh configue --without-libcurl make make install On startup only errors: PMP I'm not behind a NAT so OK Stacksize registered as too high and advised to use the -waste switch. Other than the stack thing all quiet on the new front, Sir regards Michael Jerris wrote: > I have done quite a few changes to the build system and correcting build problems and other platform specific problems the last few days. Could everyone on the list please take a little time out of their day and do a clean fresh svn trunk checkout of FreeSWITCH and do a full build and report any errors you encounter (if not already reported) to http://jira.freeswitch.org. We have fixed things for many platforms including bsd, solaris, linux, and especially issues on OS X. Please try these out to make sure all works. > > Thanks > Mike > > > > From abeka at greatiam.com Thu Nov 26 02:59:00 2009 From: abeka at greatiam.com (Otis) Date: Thu, 26 Nov 2009 10:59:00 +0000 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: <4B0B2655.4010900@greatiam.com> References: <4B086689.6080804@greatiam.com> <4B097A89.2050400@greatiam.com> <4B0ABC4F.1010103@greatiam.com> <4B0B2655.4010900@greatiam.com> Message-ID: <4B0E5F74.10009@greatiam.com> This is resolved. I could someone to call my VOIPUSER number and call transferred to my designated extension. I could not get this to work from my network ie calling from one of my extensions and setting that the call be -rerouted to another extension. All OK now. Thanks folks Otis wrote: > Has anyone got any suggestion how I can set up a gateway to receive > incoming call on extension 1001 please. > Any generic conf file will do. my username with my gateway is s=say " > qwerty" and password "ytrewq" > > I have used the intruction from the link below without success. > > Thanks. > > > > > Otis wrote: >> Hello >> >> Could anyone point out what I have missed please ? >> At the moment I configured a gateway voipuser as described here >> : >> Any suggestion as to what path I can take will be highly welcome >> >> Thanks >> . >> >> >> >> >> Sam Abekah-Mensah wrote: >>>
Hi Michael >>> >>> Thanks >>> >>> I had set it to send incoming calls to extension 1001. This is in >>> the file abeka.xml in /usr/local/freeswitch/conf/dialplan/public >>> directory. >>> The contents are : >>> >>> >>> >>> >>> >>> >>> >>> >>> Is there >>> anything wrong with this please ? >>> >>> Thanks >>> >>> >>> >>> Michal Bielicki wrote: >>>> >>>> Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah: >>>> >>>>> >>>>> I need help as I cannot receive calls through VOIPUSER. This is a >>>>> learning setup Attached are my conf files. What is wrong with them >>>>> ? When I dial from a landline I get a continuous beep. >>>>> >>>>> Attached are my gateway and the conf file to transfer. Sopfia >>>>> Status is my screen message. I can see a FAIL and cannot make head >>>>> or tail of all that message. Hopefully anyone using voipuser or in >>>>> fact any of you clever folks can make sense of this. >>>>> >>>>> Thanks for your time. >>>>> >>>>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate >>>>> Buggy RFC2833 Mode! >>>>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec >>>>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>>> 2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec >>>>> Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] >>>>> 2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec >>>>> sofia/external/nobody at 213.166.5.133 PCMA/8000 20 ms 160 samples >>>>> 2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf >>>>> payload to 101 >>>>> 2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 >>>>> (sofia/external/nobody at 213.166.5.133) State Change CS_NEW -> CS_INIT >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_INIT >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>>>> (sofia/external/nobody at 213.166.5.133) State INIT >>>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 >>>>> sofia/external/nobody at 213.166.5.133 SOFIA INIT >>>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 >>>>> (sofia/external/nobody at 213.166.5.133) State Change CS_INIT -> >>>>> CS_ROUTING >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>>>> (sofia/external/nobody at 213.166.5.133) State INIT going to sleep >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_ROUTING >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 >>>>> (sofia/external/nobody at 213.166.5.133) State ROUTING >>>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 >>>>> sofia/external/nobody at 213.166.5.133 SOFIA ROUTING >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 >>>>> sofia/external/nobody at 213.166.5.133 Standard ROUTING >>>>> 2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 >>>>> Processing anonymous->abeka in context public >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->unloop] continue=false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (PASS) >>>>> [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->outside_call] continue=true >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Absolute Condition >>>>> [outside_call] >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Action >>>>> set(outside_call=true) >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->call_debug] continue=true >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->public_extensions] continue=false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/ >>>>> break=on-false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->public_did] continue=false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [public_did] destination_number(abeka) =~ /^(5551212)$/ >>>>> break=on-false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->sip at sip.voipuser.org] continue=false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [sip at sip.voipuser.org] destination_number(abeka) =~ /08715042951/ >>>>> break=on-false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->Inbound-abeka at sip.voipuser.org]] continue=false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [Inbound-abeka at sip.voipuser.org]] destination_number(abeka) =~ >>>>> /[08444846450]/ break=on-false >>>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 >>>>> (sofia/external/nobody at 213.166.5.133) State Change CS_ROUTING -> >>>>> CS_EXECUTE >>>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 >>>>> (sofia/external/nobody at 213.166.5.133) State ROUTING going to sleep >>>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 >>>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_EXECUTE >>>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 >>>>> (sofia/external/nobody at 213.166.5.133) State EXECUTE >>>>> 2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 >>>>> sofia/external/nobody at 213.166.5.133 SOFIA EXECUTE >>>>> 2009-11-21 22:07:15.706658 [DEBUG] switch_core_state_machine.c:151 >>>>> sofia/external/nobody at 213.166.5.133 Standard EXECUTE >>>>> EXECUTE sofia/external/nobody at 213.166.5.133 set(outside_call=true) >>>>> 2009-11-21 22:07:15.728613 [DEBUG] mod_dptools.c:748 >>>>> sofia/external/nobody at 213.166.5.133 SET [outside_call]=[true] >>>>> 2009-11-21 22:07:15.728613 [NOTICE] >>>>> switch_core_state_machine.c:179 Hangup >>>>> sofia/external/nobody at 213.166.5.133 [CS_EXECUTE] [NORMAL_CLEARING] >>>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_channel.c:1683 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [KILL] >>>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_session.c:932 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:491 >>>>> (sofia/external/nobody at 213.166.5.133) State EXECUTE going to sleep >>>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:398 >>>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_HANGUP >>>>> 2009-11-21 22:07:15.735830 [DEBUG] switch_core_state_machine.c:434 >>>>> (sofia/external/nobody at 213.166.5.133) State HANGUP >>>>> 2009-11-21 22:07:15.735830 [DEBUG] mod_sofia.c:338 Channel >>>>> sofia/external/nobody at 213.166.5.133 hanging up, cause: >>>>> NORMAL_CLEARING >>>>> 2009-11-21 22:07:15.737680 [DEBUG] mod_sofia.c:417 Responding to >>>>> INVITE with: 480 >>>>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:46 >>>>> sofia/external/nobody at 213.166.5.133 Standard HANGUP, cause: >>>>> NORMAL_CLEARING >>>>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:434 >>>>> (sofia/external/nobody at 213.166.5.133) State HANGUP going to sleep >>>>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_state_machine.c:476 >>>>> (sofia/external/nobody at 213.166.5.133) State Change CS_HANGUP -> >>>>> CS_REPORTING >>>>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_session.c:932 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:398 >>>>> (sofia/external/nobody at 213.166.5.133) Running State Change >>>>> CS_REPORTING >>>>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:612 >>>>> (sofia/external/nobody at 213.166.5.133) State REPORTING >>>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:53 >>>>> sofia/external/nobody at 213.166.5.133 Standard REPORTING, cause: >>>>> NORMAL_CLEARING >>>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:612 >>>>> (sofia/external/nobody at 213.166.5.133) State REPORTING going to sleep >>>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:411 >>>>> (sofia/external/nobody at 213.166.5.133) State Change CS_REPORTING -> >>>>> CS_DESTROY >>>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_session.c:1068 >>>>> Session 2 (sofia/external/nobody at 213.166.5.133) Locked, Waiting on >>>>> external entities >>>>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1086 >>>>> Session 2 (sofia/external/nobody at 213.166.5.133) Ended >>>>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1088 >>>>> Close Channel sofia/external/nobody at 213.166.5.133 [CS_DESTROY] >>>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>>>> (sofia/external/nobody at 213.166.5.133) State DESTROY >>>>> 2009-11-21 22:07:15.802636 [DEBUG] mod_sofia.c:255 >>>>> sofia/external/nobody at 213.166.5.133 SOFIA DESTROY >>>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:60 >>>>> sofia/external/nobody at 213.166.5.133 Standard DESTROY >>>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>>>> (sofia/external/nobody at 213.166.5.133) State DESTROY going to sleep >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> : >>>> >>>> >>>> you seem to have not specified an extension where the call should >>>> go to >>>> my voipuser.org setup looks like: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> I am also surprised that your setup works with a from-domain of >>>> sip.voipuser.org >>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >>>>> http://www.freeswitch.org >>>> >>>> *Michal Bielicki* >>>> HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa >>>> t. +48228753290 | f. +48228753291 michal.bielicki at halokwadrat.pl >>>> | w. >>>> www.halokwadrat.pl >>>> >>>> >>>> >>>> *Knowledge & Low Prices. Guaranteed!* >>>> >>> >>> >>> >>>
>> >> > > From abeka at greatiam.com Thu Nov 26 03:02:12 2009 From: abeka at greatiam.com (Otis) Date: Thu, 26 Nov 2009 11:02:12 +0000 Subject: [Freeswitch-users] Re-routing calls to PSTN Message-ID: <4B0E6034.6050802@greatiam.com> Hi folks Can I get FS to re-route incoming-calls to PSTN. If this has been raised before could someone direct me to URL or link please Thanks. From jonas.gauffin at gmail.com Thu Nov 26 03:03:58 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Thu, 26 Nov 2009 12:03:58 +0100 Subject: [Freeswitch-users] Problems with nat Message-ID: I got a freeswitch that is behind nat and got three profiles. External (all calls are going through a proxy): Internal (phones on the same lan as FS) Wan (phones that are not in the same LAN, connecting from internet) The problem is that phones registered on the internal profile gets RECOVERY_ON_TIMER_EXPIRE error after 40-60 seconds. Audio works fine in all profiles. Log from a call: http://pastebin.freeswitch.org/11303 I'm running freeswitch with the -nonat option. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/e9edb679/attachment.html From michaelt at voxcore.voxtelecom.co.za Thu Nov 26 04:08:03 2009 From: michaelt at voxcore.voxtelecom.co.za (Michael Toop) Date: Thu, 26 Nov 2009 14:08:03 +0200 Subject: [Freeswitch-users] B Leg Account Code on Fail Over dialing Message-ID: <330316f60911260408u408c8e2bq3e5cd311008d8a8@mail.gmail.com> Hi Everyone, How do I get the outbound sofia SIP route that the call took into a CDR when using fail over dialing with the 'bridge' application? ...I have tried numerous approaches with no luck, this last attempt pasted below did not work either: dialString = "{provider=providerB}sofia/gateway/sipB/%s|{provider=providerC}sofia/gateway/sipC/%s" % (numberToDial, numberToDial) session.execute("bridge",dialString) I am using mod_python and this line is in the Python module called in the dialplan. Thanks! Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/8d54eb4f/attachment.html From abeka at greatiam.com Thu Nov 26 04:11:04 2009 From: abeka at greatiam.com (Otis) Date: Thu, 26 Nov 2009 12:11:04 +0000 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <221275.23339.qm@web56403.mail.re3.yahoo.com> References: <221275.23339.qm@web56403.mail.re3.yahoo.com> Message-ID: <4B0E7058.1010106@greatiam.com> Thank you so much. Regards Mark Crane wrote: > "how about trying Fusionpbx.com ( GUI)" -Ram > > I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was > ready to release now but decided to do one more release candidate just > to be sure. This should be the last release candidate before the > release of version 1.0. > > The final release may be by the end of the week as long as no major > issues are found. > > http://fusionpbx.com > > > > > --- On *Mon, 11/23/09, ram //* wrote: > > > From: ram > Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > To: freeswitch-users at lists.freeswitch.org > Date: Monday, November 23, 2009, 10:54 PM > > > > On Mon, Nov 23, 2009 at 10:37 AM, Otis > wrote: > > Thanks. > > I have to get a centos box I guess. > > Much appreciated > > Samuel 'Otis' > > > > how about trying Fusionpbx.com ( GUI) > > Ram > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From abeka at greatiam.com Thu Nov 26 04:24:36 2009 From: abeka at greatiam.com (Otis) Date: Thu, 26 Nov 2009 12:24:36 +0000 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <221275.23339.qm@web56403.mail.re3.yahoo.com> References: <221275.23339.qm@web56403.mail.re3.yahoo.com> Message-ID: <4B0E7384.5010809@greatiam.com> Thanks. I will try it . I am on Fedora 11 Mark Crane wrote: > "how about trying Fusionpbx.com ( GUI)" -Ram > > I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was > ready to release now but decided to do one more release candidate just > to be sure. This should be the last release candidate before the > release of version 1.0. > > The final release may be by the end of the week as long as no major > issues are found. > > http://fusionpbx.com > > > > > --- On *Mon, 11/23/09, ram //* wrote: > > > From: ram > Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > To: freeswitch-users at lists.freeswitch.org > Date: Monday, November 23, 2009, 10:54 PM > > > > On Mon, Nov 23, 2009 at 10:37 AM, Otis > wrote: > > Thanks. > > I have to get a centos box I guess. > > Much appreciated > > Samuel 'Otis' > > > > how about trying Fusionpbx.com ( GUI) > > Ram > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ovvenkatesan at gmail.com Thu Nov 26 04:38:12 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Thu, 26 Nov 2009 18:08:12 +0530 Subject: [Freeswitch-users] How to run IVR application In-Reply-To: <87f2f3b90911241955v4e726111ked993c8dbb556f99@mail.gmail.com> References: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> <87f2f3b90911241955v4e726111ked993c8dbb556f99@mail.gmail.com> Message-ID: <47d63d920911260438j29b56ee5w587bd6315eb64c42@mail.gmail.com> Thank you very much MC . Its working :) . I started loving "FS" ;) On Wed, Nov 25, 2009 at 9:25 AM, Michael Collins wrote: > > > On Tue, Nov 24, 2009 at 6:03 PM, Lei Tang wrote: > >> you can do this in follow steps: >> 1.edit default.xml diaplan config file in your fs config >> directory(FS/conf/dialplan/default.xml), and section >> >> >> >> >> >> 2. edit your ivr script, your can refer to >> http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in >> lua. >> 3. connect your sip phone to fs, and dial 114, this will launch your ivr >> application >> >> > > You can also do IVRs with static XML. I recommend you try out the demo IVR > by dialing 5000. Now go look at the two main files that we used to build > that IVR: > > conf/autoload_configs/ivr.conf.xml (menu structure) > conf/lang/en/demo/demo-ivr.xml (phrase macros) > > it's overwhelming at first, however once you get the hang of it you'll > appreciate how powerful it is. The wiki and the sample XML config files have > lots of information so be sure to read as much as you can and try things. > You can't break anything. :) > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/953db637/attachment.html From rupa at rupa.com Thu Nov 26 05:50:42 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 26 Nov 2009 07:50:42 -0600 Subject: [Freeswitch-users] B Leg Account Code on Fail Over dialing In-Reply-To: <330316f60911260408u408c8e2bq3e5cd311008d8a8@mail.gmail.com> References: <330316f60911260408u408c8e2bq3e5cd311008d8a8@mail.gmail.com> Message-ID: You need to use brackets [] not braces {} for per-leg variables. On Thu, Nov 26, 2009 at 6:08 AM, Michael Toop < michaelt at voxcore.voxtelecom.co.za> wrote: > Hi Everyone, > > How do I get the outbound sofia SIP route that the call took into a CDR > when using fail over dialing with the 'bridge' application? > > ...I have tried numerous approaches with no luck, this last attempt pasted > below did not work either: > > dialString = > "{provider=providerB}sofia/gateway/sipB/%s|{provider=providerC}sofia/gateway/sipC/%s" > % (numberToDial, numberToDial) > session.execute("bridge",dialString) > > I am using mod_python and this line is in the Python module called in the > dialplan. > > Thanks! > > Michael > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/05dd1522/attachment.html From freeswitch at servercorps.com Thu Nov 26 07:17:29 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Thu, 26 Nov 2009 09:17:29 -0600 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <4B0E7384.5010809@greatiam.com> References: <221275.23339.qm@web56403.mail.re3.yahoo.com> <4B0E7384.5010809@greatiam.com> Message-ID: <92e7d2090911260717j11ffad78kdd11b1c87dfd87be@mail.gmail.com> Fedora and Centos installation instructions are very similar. You should be able to compile on Fedora without any problems that I'm aware of. Regards, Nik On Thu, Nov 26, 2009 at 06:24, Otis wrote: > Thanks. I will try it . I am on Fedora 11 > > > > Mark Crane wrote: > > "how about trying Fusionpbx.com ( GUI)" -Ram > > > > I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was > > ready to release now but decided to do one more release candidate just > > to be sure. This should be the last release candidate before the > > release of version 1.0. > > > > The final release may be by the end of the week as long as no major > > issues are found. > > > > http://fusionpbx.com > > > > > > > > > > --- On *Mon, 11/23/09, ram //* wrote: > > > > > > From: ram > > Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > > To: freeswitch-users at lists.freeswitch.org > > Date: Monday, November 23, 2009, 10:54 PM > > > > > > > > On Mon, Nov 23, 2009 at 10:37 AM, Otis > > wrote: > > > > Thanks. > > > > I have to get a centos box I guess. > > > > Much appreciated > > > > Samuel 'Otis' > > > > > > > > how about trying Fusionpbx.com ( GUI) > > > > Ram > > > > -----Inline Attachment Follows----- > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/1f1d2c65/attachment.html From brian at freeswitch.org Thu Nov 26 07:42:37 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Nov 2009 09:42:37 -0600 Subject: [Freeswitch-users] Problems with nat In-Reply-To: References: Message-ID: Are you doing this all on a linux box thats acting as your router too? If not you don't need two profiles... you also don't need to set the local-network-acl on ANY profile that isn't do anything with nat. /b On Nov 26, 2009, at 5:03 AM, Jonas Gauffin wrote: > I got a freeswitch that is behind nat and got three profiles. > > External (all calls are going through a proxy): > > > > > > > Internal (phones on the same lan as FS) > > > > > Wan (phones that are not in the same LAN, connecting from internet) > > > > > > > The problem is that phones registered on the internal profile gets > RECOVERY_ON_TIMER_EXPIRE error after 40-60 seconds. Audio works fine > in all profiles. > > Log from a call: > http://pastebin.freeswitch.org/11303 > > I'm running freeswitch with the -nonat option. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/e7541c6f/attachment.html From orien at tx.rr.com Thu Nov 26 09:38:19 2009 From: orien at tx.rr.com (Orien Love) Date: Thu, 26 Nov 2009 11:38:19 -0600 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. Message-ID: <4B0EBD0B.7000905@tx.rr.com> Is there any way to build a dial plan so that when an extension calls itself the call is automatically put to that users voice mail? Example, extension 1001 calling 1001 and is sent to voice mail (to receive messages). I know that there is a * code to get to voice mail, I cannot recall which one right now but my phones want to dial their extension to get to voice mail.I can modify the voice mail button but this works only for the first line registered at that phone. Any help is appreciated. Orien From jonas.gauffin at gmail.com Thu Nov 26 10:14:34 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Thu, 26 Nov 2009 19:14:34 +0100 Subject: [Freeswitch-users] Problems with nat In-Reply-To: References: Message-ID: It's a windowsserver which is behind a router. Which profile should local-network-acl be specified on? When I bridge calls to the outside world, should I use sofia/internal/@gateway or sofia/external/@gateway? On Thu, Nov 26, 2009 at 4:42 PM, Brian West wrote: > Are you doing this all on a linux box thats acting as your router too? If > not you don't need two profiles... you also don't need to set the > local-network-acl on ANY profile that isn't do anything with nat. > > /b > > On Nov 26, 2009, at 5:03 AM, Jonas Gauffin wrote: > > I got a freeswitch that is behind nat and got three profiles. > > External (all calls are going through a proxy): > > > > > > > Internal (phones on the same lan as FS) > > > > > Wan (phones that are not in the same LAN, connecting from internet) > > > > > > > The problem is that phones registered on the internal profile > gets RECOVERY_ON_TIMER_EXPIRE error after 40-60 seconds. Audio works fine in > all profiles. > > Log from a call: > http://pastebin.freeswitch.org/11303 > > I'm running freeswitch with the -nonat option. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/592524e3/attachment-0001.html From frank at impactfax.com Thu Nov 26 11:22:52 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 26 Nov 2009 14:22:52 -0500 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS Message-ID: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> "GREAT SCOTT!!! Cannot execute batched statements! If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS" I realize a bit off of list topic. But I do have mysql 3.51.18 and higher but for the life of me , I cannot seem to get the DSN config setup so that the odbc connector seems to tell FS that it can do multi statements. Anyone have any insight on how and where to set this flag? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/118b0a06/attachment.html From timuckun at gmail.com Thu Nov 26 12:22:44 2009 From: timuckun at gmail.com (Tim Uckun) Date: Fri, 27 Nov 2009 09:22:44 +1300 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: References: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> <855e4dcf0911221543o222bef63t1c3340b0a41d57c1@mail.gmail.com> <191c3a030911230838l103bc466p7582c1d05730f61a@mail.gmail.com> Message-ID: <855e4dcf0911261222x7c73593dn18e0c2997f09d633@mail.gmail.com> On Tue, Nov 24, 2009 at 5:48 AM, Eliot Gable wrote: > Or, you can use something like Smarty to cache your generated XML on > your web server and only invalidate those cached results when you > change something that will impact them. That sounds like a pretty sane way to go bout it. From mike at jerris.com Thu Nov 26 12:44:48 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Nov 2009 15:44:48 -0500 Subject: [Freeswitch-users] Problems with nat In-Reply-To: References: Message-ID: In this case you should not need 2 profiles either. On Nov 26, 2009, at 1:14 PM, Jonas Gauffin wrote: > It's a windowsserver which is behind a router. > > Which profile should local-network-acl be specified on? > > When I bridge calls to the outside world, should I use sofia/internal/@gateway or sofia/external/@gateway? > > > On Thu, Nov 26, 2009 at 4:42 PM, Brian West wrote: > Are you doing this all on a linux box thats acting as your router too? If not you don't need two profiles... you also don't need to set the local-network-acl on ANY profile that isn't do anything with nat. > > /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/afd5f554/attachment.html From mike at jerris.com Thu Nov 26 12:47:00 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Nov 2009 15:47:00 -0500 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. In-Reply-To: <4B0EBD0B.7000905@tx.rr.com> References: <4B0EBD0B.7000905@tx.rr.com> Message-ID: <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> Of course. Please read through the default configs and the getting started guide and xml dialplan information on the wiki. Mike On Nov 26, 2009, at 12:38 PM, Orien Love wrote: > Is there any way to build a dial plan so that when an extension calls > itself the call is automatically put to that users voice mail? > > Example, extension 1001 calling 1001 and is sent to voice mail (to > receive messages). > I know that there is a * code to get to voice mail, I cannot recall > which one right now but my phones want to dial their extension to get to > voice mail.I can modify the voice mail button but this works only for > the first line registered at that phone. From mike at jerris.com Thu Nov 26 12:53:21 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Nov 2009 15:53:21 -0500 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> References: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> Message-ID: <0655757B-61CA-490D-BDB0-873263555575@jerris.com> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445) On Nov 26, 2009, at 2:22 PM, Frank @ Impact wrote: > ?GREAT SCOTT!!! Cannot execute batched statements! > If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS? > > I realize a bit off of list topic? > > But I do have mysql 3.51.18 and higher but for the life of me , I cannot seem to get the DSN config setup so that the odbc connector seems to tell FS that it can do multi statements. > > Anyone have any insight on how and where to set this flag? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/ec26c59d/attachment.html From JCasale at activenetwerx.com Thu Nov 26 13:36:04 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 26 Nov 2009 21:36:04 +0000 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. In-Reply-To: <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> References: <4B0EBD0B.7000905@tx.rr.com> <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> Message-ID: >Of course. Please read through the default configs and the getting started guide and xml dialplan information on the wiki. > >Mike This is of interest to me as well, would that be something like this: Could anyone versed in xml and variables comment on this so it generically checked if the extension dialed was of your extension length, like ^(\d{3})$ then if it matched your caller_id_number go into the action so you could leave it as $1, not 100 in my case? That way you could only have one of these plans work for all extensions. From tculjaga at gmail.com Thu Nov 26 13:48:22 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 26 Nov 2009 22:48:22 +0100 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: <0655757B-61CA-490D-BDB0-873263555575@jerris.com> References: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> <0655757B-61CA-490D-BDB0-873263555575@jerris.com> Message-ID: <65d96fc80911261348p18c1d021of3b6500ff798f345@mail.gmail.com> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris wrote: > http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html > > MySQL Connector/ODBC now supports batched statements. In order to enable > cached statement support you must switch enable the batched > statement option (FLAG_MULTI_STATEMENTS, > 67108864, or Allow multiple statements > within a GUI configuration). Be aware that batched statements > create an increased chance of SQL injection attacks and you must > ensure that your application protects against this scenario. > (Bug#7445 ) > > > so, is this the right patch ? http://bugs.mysql.com/file.php?id=6994 T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/a512b1d8/attachment.html From eman at chabotel.com Thu Nov 26 14:26:15 2009 From: eman at chabotel.com (freeswitch list) Date: Thu, 26 Nov 2009 17:26:15 -0500 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. In-Reply-To: References: <4B0EBD0B.7000905@tx.rr.com> <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> Message-ID: <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> On Thu, Nov 26, 2009 at 4:36 PM, Joseph L. Casale wrote: > >Of course. Please read through the default configs and the getting > started guide and xml dialplan information on the wiki. > > > >Mike > > This is of interest to me as well, would that be something like this: > > > > > > > > > > Could anyone versed in xml and variables comment on this so it generically > checked > if the extension dialed was of your extension length, like ^(\d{3})$ then > if it matched > your caller_id_number go into the action so you could leave it as $1, not > 100 in my case? > > That way you could only have one of these plans work for all extensions. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/d00c2357/attachment.html From JCasale at activenetwerx.com Thu Nov 26 14:43:24 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 26 Nov 2009 22:43:24 +0000 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. In-Reply-To: <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> References: <4B0EBD0B.7000905@tx.rr.com> <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> Message-ID: >? Of course:) Thank you! jlc From Russell.Mosemann at cune.org Thu Nov 26 14:45:11 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Thu, 26 Nov 2009 16:45:11 -0600 Subject: [Freeswitch-users] dialplan rule to send the caller to voicemail when same extension is called. In-Reply-To: <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> References: <4B0EBD0B.7000905@tx.rr.com><2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> Message-ID: <33C8A289190544CA8CA66014B3153363@cune.pri> freeswitch list wrote: > I knew this day would come. After the accumulation of all of the knowledge from the list members, the list has finally achieved sentience and is now answering questions by itself. :-) -- Russell Mosemann From Prometheus001 at gmx.net Thu Nov 26 15:55:17 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 27 Nov 2009 00:55:17 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> Message-ID: <4B0F1565.6060909@gmx.net> I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded. Best regards Peter Brian West schrieb: > Yes an alias will be required for every domain you run on the profile > so it can find it. > > /b > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > >> Try an alias on the sip profile. >> >> Mike >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From rob4manhere at gmail.com Thu Nov 26 18:27:46 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 26 Nov 2009 20:27:46 -0600 Subject: [Freeswitch-users] dialplan rule to send the caller to voicemail when same extension is called. In-Reply-To: <33C8A289190544CA8CA66014B3153363@cune.pri> References: <4B0EBD0B.7000905@tx.rr.com><2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> <33C8A289190544CA8CA66014B3153363@cune.pri> Message-ID: <33089F3D-0DC0-481B-B1A7-E58B35A392A8@gmail.com> LOL thats funny. freeswitch, what is the meaning of life? On Nov 26, 2009, at 4:45 PM, Russell Mosemann wrote: > freeswitch list wrote: > >> > > I knew this day would come. After the accumulation of all of the > knowledge from the list members, the list has finally achieved > sentience and is now answering questions by itself. :-) > > -- > Russell Mosemann > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andrewkt at aktzero.com Thu Nov 26 20:08:21 2009 From: andrewkt at aktzero.com (Andrew Thompson) Date: Thu, 26 Nov 2009 23:08:21 -0500 Subject: [Freeswitch-users] Re-routing calls to PSTN In-Reply-To: <4B0E6034.6050802@greatiam.com> References: <4B0E6034.6050802@greatiam.com> Message-ID: <4B0F50B5.6030804@aktzero.com> On 11/26/2009 6:02 AM, Otis wrote: > Can I get FS to re-route incoming-calls to PSTN. If this has been > raised before could someone direct me to URL or link please Since I don't have a hard line, I do something like: -- Andrew Thompson From andrew at hijacked.us Thu Nov 26 20:13:26 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 26 Nov 2009 23:13:26 -0500 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <016701ca6dfc$1a8e9ae0$4fabd0a0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> <00e101ca6cab$c3525240$49f6f6c0$@net> <21CB5F92-98DE-4622-ADC5-013462A93BD2@freeswitch.org> <20091124064509.GA6360@hijacked.us> <016701ca6dfc$1a8e9ae0$4fabd0a0$@net> Message-ID: <20091127041326.GA2000@hijacked.us> On Wed, Nov 25, 2009 at 11:21:25AM -0700, Adam Ford wrote: > Awesome, thanks Andrew, I will have to keep an eye out for that patch. > Here's my patch in its (probably) final form. http://eagle.bsd.st/~andrew/mweek2.diff It includes a usage example that covers all but 2 of the US federal holidays (memorial day is a real toughie). I'm just waiting on Tony to green light it for commit. If the patch looks like a mess in your browser, blame the XML :) Andrew From eman at chabotel.com Thu Nov 26 20:26:26 2009 From: eman at chabotel.com (eman) Date: Thu, 26 Nov 2009 23:26:26 -0500 Subject: [Freeswitch-users] dialplan rule to send the caller to voicemail when same extension is called. In-Reply-To: <33089F3D-0DC0-481B-B1A7-E58B35A392A8@gmail.com> References: <4B0EBD0B.7000905@tx.rr.com> <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> <33C8A289190544CA8CA66014B3153363@cune.pri> <33089F3D-0DC0-481B-B1A7-E58B35A392A8@gmail.com> Message-ID: <164a9ab00911262026w65b65b1dw12846463af04170a@mail.gmail.com> Weird don't know how that got set to freeswitch list. On Thu, Nov 26, 2009 at 9:27 PM, Rob Forman wrote: > LOL thats funny. > > freeswitch, what is the meaning of life? > > > On Nov 26, 2009, at 4:45 PM, Russell Mosemann wrote: > > > freeswitch list wrote: > > > >> > > > > I knew this day would come. After the accumulation of all of the > > knowledge from the list members, the list has finally achieved > > sentience and is now answering questions by itself. :-) > > > > -- > > Russell Mosemann > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/28feed69/attachment-0001.html From orien at tx.rr.com Thu Nov 26 20:51:30 2009 From: orien at tx.rr.com (Orien Love) Date: Thu, 26 Nov 2009 22:51:30 -0600 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called.(Working) In-Reply-To: References: Message-ID: <4B0F5AD2.5040000@tx.rr.com> Thanks for all the help, here is what I put in the dialplan, I tested this and it is working for me. this was added just before the line Orien Love. Still learning, but getting there with help from all the great people on this list :) Subject: Re: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. From: freeswitch list Date: Thu, 26 Nov 2009 17:26:15 -0500 To: freeswitch-users at lists.freeswitch.org > > On Thu, Nov 26, 2009 at 4:36 PM, Joseph L. Casale > > wrote: > > >Of course. Please read through the default configs and the > getting started guide and xml dialplan information on the wiki. > > > >Mike > > This is of interest to me as well, would that be something like this: > > > > > > > > > > Could anyone versed in xml and variables comment on this so it > generically checked > if the extension dialed was of your extension length, like > ^(\d{3})$ then if it matched > your caller_id_number go into the action so you could leave it as > $1, not 100 in my case? > > That way you could only have one of these plans work for all > extensions. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ovvenkatesan at gmail.com Thu Nov 26 21:59:11 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Fri, 27 Nov 2009 11:29:11 +0530 Subject: [Freeswitch-users] How to run IVR application In-Reply-To: <47d63d920911260438j29b56ee5w587bd6315eb64c42@mail.gmail.com> References: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> <87f2f3b90911241955v4e726111ked993c8dbb556f99@mail.gmail.com> <47d63d920911260438j29b56ee5w587bd6315eb64c42@mail.gmail.com> Message-ID: <47d63d920911262159s4c61e51dsb8c74d9530bc2d80@mail.gmail.com> Hi MC, I have created won sample application yesterday, It was working fine. Today, I checked that my local ip has changed. so, I changed the domain(IP) name in sip-account settings in my x-lite configuration. After that x-lite is not able to register with FS. I am getting error like "Registration error 405 : Method not Allowed ". Could you please tell me ,why its happening ? Thanks in advance, Venkat. On Thu, Nov 26, 2009 at 6:08 PM, ovvenkat wrote: > Thank you very much MC . Its working :) . I started loving "FS" ;) > > On Wed, Nov 25, 2009 at 9:25 AM, Michael Collins wrote: > >> >> >> On Tue, Nov 24, 2009 at 6:03 PM, Lei Tang wrote: >> >>> you can do this in follow steps: >>> 1.edit default.xml diaplan config file in your fs config >>> directory(FS/conf/dialplan/default.xml), and section >>> >>> >>> >>> >>> >>> 2. edit your ivr script, your can refer to >>> http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in >>> lua. >>> 3. connect your sip phone to fs, and dial 114, this will launch your ivr >>> application >>> >>> >> >> You can also do IVRs with static XML. I recommend you try out the demo IVR >> by dialing 5000. Now go look at the two main files that we used to build >> that IVR: >> >> conf/autoload_configs/ivr.conf.xml (menu structure) >> conf/lang/en/demo/demo-ivr.xml (phrase macros) >> >> it's overwhelming at first, however once you get the hang of it you'll >> appreciate how powerful it is. The wiki and the sample XML config files have >> lots of information so be sure to read as much as you can and try things. >> You can't break anything. :) >> >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > If you have come to help me, you are wasting your time. > If you have come to because your liberation is bound up in mine, we can > work together. > > > Regards > Venkatesan OV. > -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/e7bbf853/attachment.html From jonas.gauffin at gmail.com Thu Nov 26 22:09:53 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Fri, 27 Nov 2009 07:09:53 +0100 Subject: [Freeswitch-users] Problems with nat In-Reply-To: References: Message-ID: Ok. I've been running this system since FS was a beta. It stopped working after a update. I'll switch to a single profile. What NAT settings should it have? I really want to get rid of the RECOVERY_ON_TIMER_EXPIRE error. On Thu, Nov 26, 2009 at 9:44 PM, Michael Jerris wrote: > In this case you should not need 2 profiles either. > > On Nov 26, 2009, at 1:14 PM, Jonas Gauffin wrote: > > It's a windowsserver which is behind a router. > > Which profile should local-network-acl be specified on? > > When I bridge calls to the outside world, should I use > sofia/internal/@gateway or > sofia/external/@gateway? > > > On Thu, Nov 26, 2009 at 4:42 PM, Brian West wrote: > >> Are you doing this all on a linux box thats acting as your router too? If >> not you don't need two profiles... you also don't need to set the >> local-network-acl on ANY profile that isn't do anything with nat. >> >> /b >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/b245108c/attachment.html From christian.loeschenkohl at xpirio.com Fri Nov 27 00:22:19 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Fri, 27 Nov 2009 09:22:19 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> <4B0B005E.4080202@xpirio.com> <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> Message-ID: <4B0F8C3B.4000800@xpirio.com> hello sorry, for my late reply my core debugging was at info not at debug, now it's changed and i have the log needed i'm sorry but pastebin doesn't work (it seems that my trace was to big) http://pastebin.freeswitch.org/11305 says "Query failure: Got a packet bigger than 'max_allowed_packet' bytes" i'll send the logfile personal to you, hope you don't dislike this br On 2009-11-24 01:48, Anthony Minessale wrote: > You forgot to set freeswitch to debug loglevel > > You need both of the following: > > console loglevel debug > sofia profile internal siptrace on > > > > > 2009/11/23 Christian L?schenkohl > > > sorry about wasting your time (wasn't my intent) > > the log is at http://pastebin.freeswitch.org/11240 > i called 5214448370068 (also other calls are in the log) > > they now have changed 180 to 183 on the sonus, but makes no > difference here > > br > > On 2009-11-23 22:07, Anthony Minessale wrote: > > do you have the ringback variable set on the channel? > > if so it will cause 180 to attempt to play inband ringback indication > > > > I have nothing left to say because I asked for the whole log with the > > siptrace enables not just 5 lines of it. > > If you still want help, give me the log to examine and I will > tell you > > what your problem is. > > > > > > > > 2009/11/23 Christian L?schenkohl > > > >> > > > > thany ou for your answer > > > > we use g729 on all our other connections in passthrough mode > and it > > also doesn't work with alaw. > > so i don't think it's related to this. > > > > br > > > > > > On 2009-11-23 20:48, Brian West wrote: > > > Well its also G729 so I suspect you don't have G729 > > > > > > /b > > > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > > > >> hi > > >> > > >> our freeswitch server has to talk to a sonus ip-switch > > >> when we want to setup a call we do get a "100 Trying" and then a > > >> "180 Ringing" > > >> within the "180 Ringing" we get a sdp with "a=sendonly" then our > > >> freeswitch > > >> quits with a CANCEL message. > > >> i simply don't get why our freeswitch aborts the session - i think > > >> it would work > > >> if no "a=sendonly" would be present in the sdp. > > >> > > >> my technical contact doesn't want to switch 180 to 183 on the > sonus > > >> side - this would > > >> also work (i think). in fact he says that 180 ringing is vaild, he > > >> isn't that wrong in > > >> this case. > > >> > > >> our freeswitch works in proxy mode, we do use trunk 15396 > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > > >> > > >> 92.63.208.36 - freeswitch > > >> 38.105.229.100 - sonus > > >> > > >> br > > >> > > >> -- > > >> Ing. Christian L?schenkohl > > >> Technische Leitung, Forschung& Entwicklung VoIP > > >> > > >> xpirio > > >> Telekommunikation& Service GmbH > > >> Lakeside B04 > > >> 9020 Klagenfurt > > >> Austria > > >> > > >> T +43 (0) 5 77 11 - 1000 > > >> F +43 (0) 5 77 11 - 1002 > > >> E christian.loeschenkohl at xpirio.com > > > > > > >> > > >> _______________________________________________ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > >> users > > >> http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung & Entwicklung VoIP > > > > xpirio > > Telekommunikation & Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From abeka at greatiam.com Fri Nov 27 00:33:06 2009 From: abeka at greatiam.com (Otis) Date: Fri, 27 Nov 2009 08:33:06 +0000 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <92e7d2090911260717j11ffad78kdd11b1c87dfd87be@mail.gmail.com> References: <221275.23339.qm@web56403.mail.re3.yahoo.com> <4B0E7384.5010809@greatiam.com> <92e7d2090911260717j11ffad78kdd11b1c87dfd87be@mail.gmail.com> Message-ID: <4B0F8EC2.2080609@greatiam.com> Yes. I ventured to use that and got some error in connecting to the mysql database. Will try with the default sqlite before getting adventurous again. Thanks Addison Martin wrote: > Fedora and Centos installation instructions are very similar. You > should be able to compile on Fedora without any problems that I'm > aware of. > > Regards, > > Nik > > > > On Thu, Nov 26, 2009 at 06:24, Otis > wrote: > > Thanks. I will try it . I am on Fedora 11 > > > > Mark Crane wrote: > > "how about trying Fusionpbx.com ( GUI)" -Ram > > > > I'll second that! I released FusionPBX 1.0 RC5 today. I thought > it was > > ready to release now but decided to do one more release > candidate just > > to be sure. This should be the last release candidate before the > > release of version 1.0. > > > > The final release may be by the end of the week as long as no major > > issues are found. > > > > http://fusionpbx.com > > > > > > > > > > --- On *Mon, 11/23/09, ram / >/* wrote: > > > > > > From: ram > > > Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > > To: freeswitch-users at lists.freeswitch.org > > > Date: Monday, November 23, 2009, 10:54 PM > > > > > > > > On Mon, Nov 23, 2009 at 10:37 AM, Otis > > >> wrote: > > > > Thanks. > > > > I have to get a centos box I guess. > > > > Much appreciated > > > > Samuel 'Otis' > > > > > > > > how about trying Fusionpbx.com ( GUI) > > > > Ram > > > > -----Inline Attachment Follows----- > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From leon at scarlet-internet.nl Fri Nov 27 00:36:38 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Fri, 27 Nov 2009 09:36:38 +0100 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: <65d96fc80911261348p18c1d021of3b6500ff798f345@mail.gmail.com> References: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> <0655757B-61CA-490D-BDB0-873263555575@jerris.com> <65d96fc80911261348p18c1d021of3b6500ff798f345@mail.gmail.com> Message-ID: <1FCBD543-07E9-4BCA-B650-D93F0F96D6C4@scarlet-internet.nl> There's a little info here on how to enable it with odbc: http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 regards, Leon On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: > > On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris > wrote: > http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- > news-3-51-18.html > > MySQL Connector/ODBC now supports batched statements. In order to > enable > cached statement support you must switch enable the batched > statement option (FLAG_MULTI_STATEMENTS, > 67108864, or Allow multiple statements > within a GUI configuration). Be aware that batched statements > create an increased chance of SQL injection attacks and you > must > ensure that your application protects against this scenario. > (Bug#7445) > > > so, is this the right patch ? > > http://bugs.mysql.com/file.php?id=6994 > > > T. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/3b2fdda4/attachment.html From abeka at greatiam.com Fri Nov 27 00:37:09 2009 From: abeka at greatiam.com (Otis) Date: Fri, 27 Nov 2009 08:37:09 +0000 Subject: [Freeswitch-users] Re-routing calls to PSTN In-Reply-To: <4B0F50B5.6030804@aktzero.com> References: <4B0E6034.6050802@greatiam.com> <4B0F50B5.6030804@aktzero.com> Message-ID: <4B0F8FB5.2030602@greatiam.com> Thank you very much . Please what are you calling a hard line ? Andrew Thompson wrote: >
On > 11/26/2009 6:02 AM, Otis wrote: >> Can I get FS to re-route incoming-calls to PSTN. If this has been >> raised before could someone direct me to URL or link please > > Since I don't have a hard line, I do something like: > > > > > data="sofia/gateway/YOURPROVIDER/18005551212"/> > > > > From zolotov at altron.ua Fri Nov 27 00:49:53 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Fri, 27 Nov 2009 10:49:53 +0200 Subject: [Freeswitch-users] Re-routing calls to PSTN In-Reply-To: <4B0F8FB5.2030602@greatiam.com> References: <4B0E6034.6050802@greatiam.com> <4B0F50B5.6030804@aktzero.com> <4B0F8FB5.2030602@greatiam.com> Message-ID: <4B0F92B1.2010601@altron.ua> Please try this: Otis ?????: > Thank you very much . Please what are you calling a hard line ? > > > > Andrew Thompson wrote: > >>
On >> 11/26/2009 6:02 AM, Otis wrote: >> >>> Can I get FS to re-route incoming-calls to PSTN. If this has been >>> raised before could someone direct me to URL or link please >>> >> Since I don't have a hard line, I do something like: >> >> >> >> >> > data="sofia/gateway/YOURPROVIDER/18005551212"/> >> >> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steve.kurzeja at gmail.com Fri Nov 27 02:00:49 2009 From: steve.kurzeja at gmail.com (Steve Kurzeja) Date: Fri, 27 Nov 2009 23:00:49 +1300 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: <191c3a030911251310h9f8bd1epf0d445c746e968a5@mail.gmail.com> References: <191c3a030911251310h9f8bd1epf0d445c746e968a5@mail.gmail.com> Message-ID: <5f7152000911270200o4cb73541ud05136bd02866447@mail.gmail.com> Is that USD ? :) On Thu, Nov 26, 2009 at 10:10 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I can spare you the pain and let you know outright that this sort of > functionality will cost somewhere in the range of 125,000.00 to 150,000.00 > to properly implement by assembling a team of consultants including members > of the development team from both FreeSWITCH and Sofia-SIP and even if you > have the money, finding the time to implement it would also be a factor as > it's a few thousand man-hours of work. > > > On Wed, Nov 25, 2009 at 3:01 PM, Mathieu Rene wrote: > >> You can read all about the sip library at >> http://sofia-sip.sourceforge.net/refdocs/ >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 25-Nov-09, at 3:58 PM, srinivasula reddy wrote: >> >> thanks for your reply mike, >> is there any api in freeswitch or any thing else to update lib >> programatically from pjsua. >> >> srinivas >> >> On Thu, Nov 26, 2009 at 2:05 AM, Michael Jerris wrote: >> >>> "something that is not available in that lib at this time." >>> >>> Mike >>> >>> On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote >>> >>> can please tell me how can i exchange session state into sip library. >>> >>> Thanks >>> srinivas >>> >>> On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: >>> >>>> For that you would need to fully exchange session state into the sip >>>> library, *something that is not available in that lib at this time.* >>>> >>>> >>>> On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: >>>> >>>> HI, >>>> thanks for your reply, my requirement is i am doing failover stuff with >>>> freeswitch. i dont want cut the calls when freeswitch dies, when failover >>>> happens mean one freeswitch dies we are going to start the second >>>> freeswitch, i dont want close call intiated by the first freeswtich, they >>>> are communicating with meida(bypass media). when one endpoing try to end the >>>> call at that time i want to close the call for the other end also. >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Srinivasula Reddy K >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Srinivasula Reddy K >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/f41f7967/attachment-0001.html From abeka at greatiam.com Fri Nov 27 03:15:08 2009 From: abeka at greatiam.com (Otis) Date: Fri, 27 Nov 2009 11:15:08 +0000 Subject: [Freeswitch-users] Connecting Multiple domains Message-ID: <4B0FB4BC.3090204@greatiam.com> Could someone please direct me to a link for connecting multiple say 2 domains each with their own FS server. Thanks From tculjaga at gmail.com Fri Nov 27 04:19:09 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 27 Nov 2009 13:19:09 +0100 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: <5f7152000911270200o4cb73541ud05136bd02866447@mail.gmail.com> References: <191c3a030911251310h9f8bd1epf0d445c746e968a5@mail.gmail.com> <5f7152000911270200o4cb73541ud05136bd02866447@mail.gmail.com> Message-ID: <65d96fc80911270419x5935d50cpcae48ca8dfa1ee92@mail.gmail.com> On Fri, Nov 27, 2009 at 11:00 AM, Steve Kurzeja wrote: > Is that USD ? :) > > i believe these are not Turkish liras :P -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/fddb6411/attachment.html From tculjaga at gmail.com Fri Nov 27 04:21:55 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 27 Nov 2009 13:21:55 +0100 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: <1FCBD543-07E9-4BCA-B650-D93F0F96D6C4@scarlet-internet.nl> References: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> <0655757B-61CA-490D-BDB0-873263555575@jerris.com> <65d96fc80911261348p18c1d021of3b6500ff798f345@mail.gmail.com> <1FCBD543-07E9-4BCA-B650-D93F0F96D6C4@scarlet-internet.nl> Message-ID: <65d96fc80911270421y5a2c6cb6n90d1f48fccb23a4c@mail.gmail.com> On Fri, Nov 27, 2009 at 9:36 AM, Leon de Rooij wrote: > There's a little info here on how to enable it with odbc: > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 > > regards, > > Leon > > well i was centanly blind when i asked this :P [maxpowersoft_odbc] Driver = MySQL SERVER = localhost PORT = 3306 DATABASE = myDatabase *OPTIONS = 67108864* Socket = /var/lib/mysql/mysql.sock Thanks. T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/c326afdc/attachment.html From frank at impactfax.com Fri Nov 27 06:41:44 2009 From: frank at impactfax.com (Frank @ Impact) Date: Fri, 27 Nov 2009 09:41:44 -0500 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: <1FCBD543-07E9-4BCA-B650-D93F0F96D6C4@scarlet-internet.nl> Message-ID: Thanks. But when I made these entries in /etc/odbc.ini and rebooted. [freeswitch] Driver = MySQL SERVER = 127.0.0.1 PORT = 4040 DATABASE = mydb OPTIONS = 67108864 .I still get FS complaining with this. Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched statements!#012If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS FreeSWITCH>version FreeSWITCH Version 1.0.trunk (15660) Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 EST 2009 x86_64 x86_64 x86_64 GNU/Linux >From /etc/odbcinst.ini DRIVER = /usr/lib64/libmyodbc5-5.1.5.so Setup = /usr/lib64/libodbcmyS.so Is this a FS issue ? or an issue with mysql odbc? Any insight would be great. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Leon de Rooij Sent: Friday, November 27, 2009 3:37 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS There's a little info here on how to enable it with odbc: http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 regards, Leon On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris wrote: http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445 ) so, is this the right patch ? http://bugs.mysql.com/file.php?id=6994 T. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/78f2b0c3/attachment-0001.html From leon at scarlet-internet.nl Fri Nov 27 07:19:03 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Fri, 27 Nov 2009 16:19:03 +0100 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: References: Message-ID: Are you using the myodbc 3.51.18 version or higher ? I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to upgrade from jaunty.. regards, Leon On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: > Thanks. But when I made these entries in /etc/odbc.ini and rebooted? > > [freeswitch] > Driver = MySQL > SERVER = 127.0.0.1 > PORT = 4040 > DATABASE = mydb > OPTIONS = 67108864 > > ?I still get FS complaining with this. > > Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 > [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched > statements!#012If you are using mysql, make sure you are using > MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS > > FreeSWITCH>version > FreeSWITCH Version 1.0.trunk (15660) > > Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 > EST 2009 x86_64 x86_64 x86_64 GNU/Linux > > From /etc/odbcinst.ini > DRIVER = /usr/lib64/libmyodbc5-5.1.5.so > Setup = /usr/lib64/libodbcmyS.so > > Is this a FS issue ? or an issue with mysql odbc? Any insight > would be great. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Leon de Rooij > Sent: Friday, November 27, 2009 3:37 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > > There's a little info here on how to enable it with odbc: > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 > > regards, > > Leon > > > On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: > > > > On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris > wrote: > http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- > news-3-51-18.html > > MySQL Connector/ODBC now supports batched statements. In order to > enable > cached statement support you must switch enable the batched > statement option (FLAG_MULTI_STATEMENTS, > 67108864, or Allow multiple statements > within a GUI configuration). Be aware that batched statements > create an increased chance of SQL injection attacks and you > must > ensure that your application protects against this scenario. > (Bug#7445) > > > so, is this the right patch ? > > http://bugs.mysql.com/file.php?id=6994 > > > T. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/9c86b324/attachment-0001.html From frank at impactfax.com Fri Nov 27 07:36:55 2009 From: frank at impactfax.com (Frank @ Impact) Date: Fri, 27 Nov 2009 10:36:55 -0500 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: Message-ID: Yes. I am using version 5.1 I am using Fedora 12. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Leon de Rooij Sent: Friday, November 27, 2009 10:19 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS Are you using the myodbc 3.51.18 version or higher ? I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to upgrade from jaunty.. regards, Leon On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: Thanks. But when I made these entries in /etc/odbc.ini and rebooted. [freeswitch] Driver = MySQL SERVER = 127.0.0.1 PORT = 4040 DATABASE = mydb OPTIONS = 67108864 .I still get FS complaining with this. Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched statements!#012If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS FreeSWITCH>version FreeSWITCH Version 1.0.trunk (15660) Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 EST 2009 x86_64 x86_64 x86_64 GNU/Linux >From /etc/odbcinst.ini DRIVER = /usr/lib64/libmyodbc5-5.1.5.so Setup = /usr/lib64/libodbcmyS.so Is this a FS issue ? or an issue with mysql odbc? Any insight would be great. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Leon de Rooij Sent: Friday, November 27, 2009 3:37 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS There's a little info here on how to enable it with odbc: http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 regards, Leon On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris wrote: http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445 ) so, is this the right patch ? http://bugs.mysql.com/file.php?id=6994 T. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/1bb30a5f/attachment-0001.html From andrewkt at aktzero.com Fri Nov 27 08:34:55 2009 From: andrewkt at aktzero.com (Andrew Thompson) Date: Fri, 27 Nov 2009 11:34:55 -0500 Subject: [Freeswitch-users] Re-routing calls to PSTN In-Reply-To: <4B0F8FB5.2030602@greatiam.com> References: <4B0E6034.6050802@greatiam.com> <4B0F50B5.6030804@aktzero.com> <4B0F8FB5.2030602@greatiam.com> Message-ID: <4B0FFFAF.6010700@aktzero.com> On 11/27/2009 3:37 AM, Otis wrote: > Thank you very much . Please what are you calling a hard line ? A real honest to goodness POTS line within 20 feet of and attached to my FS server. My calls come in and go out SIP, so If I was sending an inbound call back to the PSTN, I'd just route it back out through my SIP provider. -- Andrew Thompson From andrewkt at aktzero.com Fri Nov 27 08:44:51 2009 From: andrewkt at aktzero.com (Andrew Thompson) Date: Fri, 27 Nov 2009 11:44:51 -0500 Subject: [Freeswitch-users] FS missing in action (was: How to run IVR application) In-Reply-To: <47d63d920911262159s4c61e51dsb8c74d9530bc2d80@mail.gmail.com> References: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> <87f2f3b90911241955v4e726111ked993c8dbb556f99@mail.gmail.com> <47d63d920911260438j29b56ee5w587bd6315eb64c42@mail.gmail.com> <47d63d920911262159s4c61e51dsb8c74d9530bc2d80@mail.gmail.com> Message-ID: <4B100203.4090502@aktzero.com> On 11/27/2009 12:59 AM, ovvenkat wrote: > Hi MC, > > I have created won sample application yesterday, It was working fine. > Today, I checked that my local ip has changed. so, I changed the > domain(IP) name in sip-account settings in my x-lite configuration. > After that x-lite is not able to register with FS. I am getting error > like "Registration error 405 : Method not Allowed ". Could you please > tell me ,why its happening ? Wait, what? First, don't re-use an existing thread, messages have a tendancy to get ignored/lost that way. Did the IP of your FS change, or of your PC? I would expect "local ip" to mean your DHCP'ed address from your Internet connection. That should have no bearing on the IP of your FS. Go find your FS and make sure *IT* is still on the IP you expect it to be on. If your PC running x-lite is also your FS, you may have other issues with IP address changing that I don't know how to handle, as I've only used static IPs for FS. (Or, try connecting via localhost, 127.0.0.1 instead, until you're ready to really start using FS.) -- Andrew Thompson From jbarou at sqli.com Fri Nov 27 08:47:23 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Fri, 27 Nov 2009 17:47:23 +0100 Subject: [Freeswitch-users] Transfer Problem Message-ID: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> Hi everybody, I'm actually using the lastest version of Freeswitch, I have a problem. I have a trunk SIP with my PABX. There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) 2. one Alcatel IpTouch 4028 with number 987 (on PABX) 3. one Siemens Gigaset A580 IP with number 8401 (on Freeswitch) *The first test* is to say to the phone 2 to transfer all the call to number 8401. So when I dial 987 on the phone 1, all work perfectly, the phone 3 is ringing and it's work. I have that in the log : 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending early media 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP [sofia/internal/368 at 10.33.69.246] 10.33.169.92 port 23054 -> 10.33.69.246 port 32000 codec: 8 ms: 90 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 720 bytes per 90ms 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: v=0 o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 s=FreeSWITCH c=IN IP4 10.33.169.92 t=0 0 m=audio 23054 RTP/AVP 8 106 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:90 a=sendrecv 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer sofia/internal/368 at 10.33.69.246! 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ 368 at 10.33.69.246 skipping state [early][183] 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal sofia/internal/368 at 10.33.69.246 [BREAK] 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec Activation Success L16 at 8000hz 1 channel 90ms 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ 368 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port confirmed. 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read Buffer at 1440 bytes vs 81 *The Second Tes*t is to say to the phone 1 to transfer all the call to number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just one time and after it hangup. I have that in the log : 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending early media 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP [sofia/internal/987 at 10.33.69.246] 10.33.169.92 port 27732 -> 10.33.69.144 port 32000 codec: 8 ms: 90 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 720 bytes per 90ms 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: v=0 o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 s=FreeSWITCH c=IN IP4 10.33.169.92 t=0 0 m=audio 27732 RTP/AVP 8 106 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:90 a=sendrecv 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer sofia/internal/987 at 10.33.69.246! 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ 987 at 10.33.69.246 skipping state [early][183] 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal sofia/internal/987 at 10.33.69.246 [BREAK] 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec Activation Success L16 at 8000hz 1 channel 90ms 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ 987 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ 987 at 10.33.69.246 entering state [terminated][487] 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ 987 at 10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/987 at 10.33.69.246 [KILL] 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/987 at 10.33.69.246 [BREAK] 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 thread mismatch skipping state handler. 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [KILL] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change CS_HANGUP 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ sip:8401 at 10.33.170.231:5060 Overriding SIP cause 487 with 487 from the other leg 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ sip:8401 at 10.33.170.231:5060 hanging up, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to sofia/internal/sip:8401 at 10.33.170.231:5060 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:8401 at 10.33.170.231:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_HANGUP -> CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:8401 at 10.33.170.231:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_REPORTING -> CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 (sofia/internal/sip:8401 at 10.33.170.231:5060) Locked, Waiting on external entities 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 thread mismatch skipping state handler. 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 48 (sofia/internal/sip:8401 at 10.33.170.231:5060) Ended 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close Channel sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_DESTROY] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ sip:8401 at 10.33.170.231:5060 SOFIA DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:8401 at 10.33.170.231:5060 Standard DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY going to sleep 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. Cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/987 at 10.33.69.246) State EXECUTE going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/987 at 10.33.69.246) Running State Change CS_HANGUP 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/987 at 10.33.69.246) State HANGUP 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ 987 at 10.33.69.246 Overriding SIP cause 487 with 487 from the other leg 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ 987 at 10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 sofia/internal/987 at 10.33.69.246 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/987 at 10.33.69.246) State HANGUP going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/987 at 10.33.69.246) State Change CS_HANGUP -> CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/987 at 10.33.69.246 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/987 at 10.33.69.246) Running State Change CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/987 at 10.33.69.246) State REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 sofia/internal/987 at 10.33.69.246 Standard REPORTING, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/987 at 10.33.69.246) State REPORTING going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/987 at 10.33.69.246) State Change CS_REPORTING -> CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/987 at 10.33.69.246 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 (sofia/internal/987 at 10.33.69.246) Locked, Waiting on external entities 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 47 (sofia/internal/987 at 10.33.69.246) Ended 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close Channel sofia/internal/987 at 10.33.69.246 [CS_DESTROY] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/987 at 10.33.69.246) Running State Change CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/987 at 10.33.69.246) State DESTROY 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ 987 at 10.33.69.246 SOFIA DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 sofia/internal/987 at 10.33.69.246 Standard DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/987 at 10.33.69.246) State DESTROY going to sleep Finally when I tried to call the phone 3 with the phone 1 it's working, and not when I want to call the phone 3 with the phone 2, like just before, it's ringing just one time and hangup. Thanks you. Best Regards -- John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/316a1bfc/attachment-0001.html From samuelmukoti at gmail.com Fri Nov 27 08:45:51 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Fri, 27 Nov 2009 18:45:51 +0200 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: References: Message-ID: Hi, Any recommendations for apps that can I use ontop of freeswitch as a GUI manager, to manage extensions, queues, ivr, and dialplans? Thanks Sam On 27 Nov,2009, at 5:19 PM, freeswitch-users-request at lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > Today's Topics: > > 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 27 Nov 2009 16:19:03 +0100 > From: Leon de Rooij > Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="windows-1252" > > Are you using the myodbc 3.51.18 version or higher ? > > I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to > upgrade from jaunty.. > > regards, > > Leon > > > On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: > >> Thanks. But when I made these entries in /etc/odbc.ini and rebooted? >> >> [freeswitch] >> Driver = MySQL >> SERVER = 127.0.0.1 >> PORT = 4040 >> DATABASE = mydb >> OPTIONS = 67108864 >> >> ?I still get FS complaining with this. >> >> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 >> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched >> statements!#012If you are using mysql, make sure you are using >> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS >> >> FreeSWITCH>version >> FreeSWITCH Version 1.0.trunk (15660) >> >> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 >> EST 2009 x86_64 x86_64 x86_64 GNU/Linux >> >> From /etc/odbcinst.ini >> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so >> Setup = /usr/lib64/libodbcmyS.so >> >> Is this a FS issue ? or an issue with mysql odbc? Any insight >> would be great. >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Leon de Rooij >> Sent: Friday, November 27, 2009 3:37 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >> >> There's a little info here on how to enable it with odbc: >> >> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 >> >> regards, >> >> Leon >> >> >> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: >> >> >> >> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris >> wrote: >> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- >> news-3-51-18.html >> >> MySQL Connector/ODBC now supports batched statements. In order to >> enable >> cached statement support you must switch enable the batched >> statement option (FLAG_MULTI_STATEMENTS, >> 67108864, or Allow multiple statements >> within a GUI configuration). Be aware that batched statements >> create an increased chance of SQL injection attacks and you >> must >> ensure that your application protects against this scenario. >> (Bug#7445) >> >> >> so, is this the right patch ? >> >> http://bugs.mysql.com/file.php?id=6994 >> >> >> T. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/9c86b324/attachment.html > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > End of FreeSWITCH-users Digest, Vol 41, Issue 209 > ************************************************* From anthony.minessale at gmail.com Fri Nov 27 08:57:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 10:57:06 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <4B0F8C3B.4000800@xpirio.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> <4B0B005E.4080202@xpirio.com> <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> <4B0F8C3B.4000800@xpirio.com> Message-ID: <191c3a030911270857j697738f9n99e9e5bb1b71c38@mail.gmail.com> or you can put it at a url on your web site and just post a link 2009/11/27 Christian L?schenkohl > hello > > sorry, for my late reply > my core debugging was at info not at debug, now it's changed and i have the > log needed > > i'm sorry but pastebin doesn't work (it seems that my trace was to big) > http://pastebin.freeswitch.org/11305 says "Query failure: Got a packet > bigger than 'max_allowed_packet' bytes" > > i'll send the logfile personal to you, hope you don't dislike this > > br > > On 2009-11-24 01:48, Anthony Minessale wrote: > > You forgot to set freeswitch to debug loglevel > > > > You need both of the following: > > > > console loglevel debug > > sofia profile internal siptrace on > > > > > > > > > > 2009/11/23 Christian L?schenkohl > > > > > > sorry about wasting your time (wasn't my intent) > > > > the log is at http://pastebin.freeswitch.org/11240 > > i called 5214448370068 (also other calls are in the log) > > > > they now have changed 180 to 183 on the sonus, but makes no > > difference here > > > > br > > > > On 2009-11-23 22:07, Anthony Minessale wrote: > > > do you have the ringback variable set on the channel? > > > if so it will cause 180 to attempt to play inband ringback > indication > > > > > > I have nothing left to say because I asked for the whole log with > the > > > siptrace enables not just 5 lines of it. > > > If you still want help, give me the log to examine and I will > > tell you > > > what your problem is. > > > > > > > > > > > > 2009/11/23 Christian L?schenkohl > > > > > > > >> > > > > > > thany ou for your answer > > > > > > we use g729 on all our other connections in passthrough mode > > and it > > > also doesn't work with alaw. > > > so i don't think it's related to this. > > > > > > br > > > > > > > > > On 2009-11-23 20:48, Brian West wrote: > > > > Well its also G729 so I suspect you don't have G729 > > > > > > > > /b > > > > > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > > > > > >> hi > > > >> > > > >> our freeswitch server has to talk to a sonus ip-switch > > > >> when we want to setup a call we do get a "100 Trying" and then > a > > > >> "180 Ringing" > > > >> within the "180 Ringing" we get a sdp with "a=sendonly" then > our > > > >> freeswitch > > > >> quits with a CANCEL message. > > > >> i simply don't get why our freeswitch aborts the session - i > think > > > >> it would work > > > >> if no "a=sendonly" would be present in the sdp. > > > >> > > > >> my technical contact doesn't want to switch 180 to 183 on the > > sonus > > > >> side - this would > > > >> also work (i think). in fact he says that 180 ringing is vaild, > he > > > >> isn't that wrong in > > > >> this case. > > > >> > > > >> our freeswitch works in proxy mode, we do use trunk 15396 > > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > > > >> > > > >> 92.63.208.36 - freeswitch > > > >> 38.105.229.100 - sonus > > > >> > > > >> br > > > >> > > > >> -- > > > >> Ing. Christian L?schenkohl > > > >> Technische Leitung, Forschung& Entwicklung VoIP > > > >> > > > >> xpirio > > > >> Telekommunikation& Service GmbH > > > >> Lakeside B04 > > > >> 9020 Klagenfurt > > > >> Austria > > > >> > > > >> T +43 (0) 5 77 11 - 1000 > > > >> F +43 (0) 5 77 11 - 1002 > > > >> E christian.loeschenkohl at xpirio.com > > > > > > > > > > >> > > > >> _______________________________________________ > > > >> FreeSWITCH-users mailing list > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > > >> users > > > >> http://www.freeswitch.org > > > > > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > -- > > > Ing. Christian L?schenkohl > > > Technische Leitung, Forschung & Entwicklung VoIP > > > > > > xpirio > > > Telekommunikation & Service GmbH > > > Lakeside B04 > > > 9020 Klagenfurt > > > Austria > > > > > > T +43 (0) 5 77 11 - 1000 > > > F +43 (0) 5 77 11 - 1002 > > > E christian.loeschenkohl at xpirio.com > > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > > > > > > > > >> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > > > > > >> > > > IRC: irc.freenode.net > > #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > > > > > > > > >> > > > iax:guest at conference.freeswitch.org/888 > > > > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > > > > > >> > > > pstn:213-799-1400 > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung & Entwicklung VoIP > > > > xpirio > > Telekommunikation & Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/f8497a2d/attachment-0001.html From anthony.minessale at gmail.com Fri Nov 27 09:03:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 11:03:10 -0600 Subject: [Freeswitch-users] Transfer Problem In-Reply-To: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> References: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> Message-ID: <191c3a030911270903i341d1f83pa15f67443422cb67@mail.gmail.com> by latest do you mean SVN trunk? Can you issue the command "sofia profile internal siptrace on" before capturing your trace and post the results to http://pastebin.freeswitch.org or open a jira http://jira.freeswitch.orgon the issue and attach the log after you create the issue ticket, don't include it in the mailing list. On Fri, Nov 27, 2009 at 10:47 AM, Jonathan Barou wrote: > Hi everybody, > > I'm actually using the lastest version of Freeswitch, I have a problem. I > have a trunk SIP with my PABX. > > There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) > 2. one Alcatel IpTouch 4028 with number 987 > (on PABX) > 3. one Siemens Gigaset A580 IP with number > 8401 (on Freeswitch) > > > *The first test* is to say to the phone 2 to transfer all the call to > number 8401. So when I dial 987 on the phone 1, all work perfectly, the > phone 3 is ringing and it's work. I have that in the log : > > 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending early > media > > 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP > [sofia/internal/368 at 10.33.69.246] 10.33.169.92 port 23054 -> 10.33.69.246 > port 32000 codec: 8 ms: 90 > > 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer [soft] > 720 bytes per 90ms > > 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: > > v=0 > > o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 > > s=FreeSWITCH > > c=IN IP4 10.33.169.92 > > t=0 0 > > m=audio 23054 RTP/AVP 8 106 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:106 telephone-event/8000 > > a=fmtp:106 0-16 > > a=silenceSupp:off - - - - > > a=ptime:90 > > a=sendrecv > > > 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer > sofia/internal/368 at 10.33.69.246! > > 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] > > 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ > sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] > > 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ > 368 at 10.33.69.246 skipping state [early][183] > > 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal > sofia/internal/368 at 10.33.69.246 [BREAK] > > 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec > Activation Success L16 at 8000hz 1 channel 90ms > > 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play > Ringback Tone [%(2000,4000,440.0,480.0)] > > 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ > 368 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] > > 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port > confirmed. > > 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read > Buffer at 1440 bytes vs 81 > > > > *The Second Tes*t is to say to the phone 1 to transfer all the call to > number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just > one time and after it hangup. I have that in the log : > > > 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending > early media > > 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP > [sofia/internal/987 at 10.33.69.246] 10.33.169.92 port 27732 -> 10.33.69.144 > port 32000 codec: 8 ms: 90 > > 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer [soft] > 720 bytes per 90ms > > 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: > > v=0 > > o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 > > s=FreeSWITCH > > c=IN IP4 10.33.169.92 > > t=0 0 > > m=audio 27732 RTP/AVP 8 106 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:106 telephone-event/8000 > > a=fmtp:106 0-16 > > a=silenceSupp:off - - - - > > a=ptime:90 > > a=sendrecv > > > 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer > sofia/internal/987 at 10.33.69.246! > > 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] > > 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ > sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] > > 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ > 987 at 10.33.69.246 skipping state [early][183] > > 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal > sofia/internal/987 at 10.33.69.246 [BREAK] > > 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec > Activation Success L16 at 8000hz 1 channel 90ms > > 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play > Ringback Tone [%(2000,4000,440.0,480.0)] > > 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ > 987 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] > > 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ > 987 at 10.33.69.246 entering state [terminated][487] > > 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ > 987 at 10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] > > 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal > sofia/internal/987 at 10.33.69.246 [KILL] > > 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/987 at 10.33.69.246 [BREAK] > > 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 thread > mismatch skipping state handler. > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore original > codec. > > 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup > sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [KILL] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change > CS_HANGUP > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ > sip:8401 at 10.33.170.231:5060 Overriding SIP cause 487 with 487 from the > other leg > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ > sip:8401 at 10.33.170.231:5060 hanging up, cause: ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to > sofia/internal/sip:8401 at 10.33.170.231:5060 > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/sip:8401 at 10.33.170.231:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP going to sleep > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_HANGUP -> > CS_REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change > CS_REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/sip:8401 at 10.33.170.231:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING going to > sleep > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_REPORTING -> > CS_DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 > (sofia/internal/sip:8401 at 10.33.170.231:5060) Locked, Waiting on external > entities > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 thread > mismatch skipping state handler. > > 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate > Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > > 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 48 > (sofia/internal/sip:8401 at 10.33.170.231:5060) Ended > > 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close > Channel sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_DESTROY] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change > CS_DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ > sip:8401 at 10.33.170.231:5060 SOFIA DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/sip:8401 at 10.33.170.231:5060 Standard DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY going to sleep > > 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot create > outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > > 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. > Cause: ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/987 at 10.33.69.246) State EXECUTE going to sleep > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/987 at 10.33.69.246) Running State Change CS_HANGUP > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 > (sofia/internal/987 at 10.33.69.246) State HANGUP > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ > 987 at 10.33.69.246 Overriding SIP cause 487 with 487 from the other leg > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ > 987 at 10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/987 at 10.33.69.246 Standard HANGUP, cause: ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 > (sofia/internal/987 at 10.33.69.246) State HANGUP going to sleep > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/987 at 10.33.69.246) State Change CS_HANGUP -> CS_REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/987 at 10.33.69.246 [BREAK] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/987 at 10.33.69.246) Running State Change CS_REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 > (sofia/internal/987 at 10.33.69.246) State REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/987 at 10.33.69.246 Standard REPORTING, cause: > ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 > (sofia/internal/987 at 10.33.69.246) State REPORTING going to sleep > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/987 at 10.33.69.246) State Change CS_REPORTING -> CS_DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/987 at 10.33.69.246 [BREAK] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 > (sofia/internal/987 at 10.33.69.246) Locked, Waiting on external entities > > 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 47 > (sofia/internal/987 at 10.33.69.246) Ended > > 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close > Channel sofia/internal/987 at 10.33.69.246 [CS_DESTROY] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/987 at 10.33.69.246) Running State Change CS_DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/987 at 10.33.69.246) State DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ > 987 at 10.33.69.246 SOFIA DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/987 at 10.33.69.246 Standard DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/987 at 10.33.69.246) State DESTROY going to sleep > > Finally when I tried to call the phone 3 with the phone 1 it's working, and > not when I want to call the phone 3 with the phone 2, like just before, it's > ringing just one time and hangup. > > > Thanks you. > > > Best Regards > > -- > John > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/d11148c7/attachment-0001.html From abeka at greatiam.com Fri Nov 27 10:42:51 2009 From: abeka at greatiam.com (Otis) Date: Fri, 27 Nov 2009 18:42:51 +0000 Subject: [Freeswitch-users] Re-routing calls to PSTN In-Reply-To: <4B0FFFAF.6010700@aktzero.com> References: <4B0E6034.6050802@greatiam.com> <4B0F50B5.6030804@aktzero.com> <4B0F8FB5.2030602@greatiam.com> <4B0FFFAF.6010700@aktzero.com> Message-ID: <4B101DAB.4040500@greatiam.com> Ok. Thanks Andrew Thompson wrote: >
On > 11/27/2009 3:37 AM, Otis wrote: >> Thank you very much . Please what are you calling a hard line ? > A real honest to goodness POTS line within 20 feet of and attached to > my FS server. > > My calls come in and go out SIP, so If I was sending an inbound call > back to the PSTN, I'd just route it back out through my SIP provider. > From abeka at greatiam.com Fri Nov 27 10:48:35 2009 From: abeka at greatiam.com (Otis) Date: Fri, 27 Nov 2009 18:48:35 +0000 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: References: Message-ID: <4B101F03.5090802@greatiam.com> Hi I am no sure but read up on fusionpbx. I asked the same question and someone pointed me to that. check web site Regards Samuel Mukoti wrote: >
Hi, > > Any recommendations for apps that can I use ontop of freeswitch as a > GUI manager, to manage extensions, queues, ivr, and dialplans? > > Thanks > > Sam > > > On 27 Nov,2009, at 5:19 PM, > freeswitch-users-request at lists.freeswitch.org wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Fri, 27 Nov 2009 16:19:03 +0100 >> From: Leon de Rooij >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset="windows-1252" >> >> Are you using the myodbc 3.51.18 version or higher ? >> >> I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to >> upgrade from jaunty.. >> >> regards, >> >> Leon >> >> >> On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: >> >>> Thanks. But when I made these entries in /etc/odbc.ini and rebooted? >>> >>> [freeswitch] >>> Driver = MySQL >>> SERVER = 127.0.0.1 >>> PORT = 4040 >>> DATABASE = mydb >>> OPTIONS = 67108864 >>> >>> ?I still get FS complaining with this. >>> >>> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 >>> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched >>> statements!#012If you are using mysql, make sure you are using >>> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS >>> >>> FreeSWITCH>version >>> FreeSWITCH Version 1.0.trunk (15660) >>> >>> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 >>> EST 2009 x86_64 x86_64 x86_64 GNU/Linux >>> >>> From /etc/odbcinst.ini >>> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so >>> Setup = /usr/lib64/libodbcmyS.so >>> >>> Is this a FS issue ? or an issue with mysql odbc? Any insight >>> would be great. >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Leon de Rooij >>> Sent: Friday, November 27, 2009 3:37 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >>> >>> There's a little info here on how to enable it with odbc: >>> >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 >>> >>> regards, >>> >>> Leon >>> >>> >>> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: >>> >>> >>> >>> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris >>> wrote: >>> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- >>> news-3-51-18.html >>> >>> MySQL Connector/ODBC now supports batched statements. In order to >>> enable >>> cached statement support you must switch enable the batched >>> statement option (FLAG_MULTI_STATEMENTS, >>> 67108864, or Allow multiple statements >>> within a GUI configuration). Be aware that batched statements >>> create an increased chance of SQL injection attacks and you >>> must >>> ensure that your application protects against this scenario. >>> (Bug#7445) >>> >>> >>> so, is this the right patch ? >>> >>> http://bugs.mysql.com/file.php?id=6994 >>> >>> >>> T. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/9c86b324/attachment.html >> >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 209 >> ************************************************* > > >
> From anthony.minessale at gmail.com Fri Nov 27 11:03:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 13:03:48 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <191c3a030911270857j697738f9n99e9e5bb1b71c38@mail.gmail.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> <4B0B005E.4080202@xpirio.com> <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> <4B0F8C3B.4000800@xpirio.com> <191c3a030911270857j697738f9n99e9e5bb1b71c38@mail.gmail.com> Message-ID: <191c3a030911271103s462910acu96da189b4064e63e@mail.gmail.com> please update to latest trunk 15698 or greater and re-test. The 183 from the provider had a sendonly attr that tricked the proxy code into thinking it was a hold/unhold operation. On Fri, Nov 27, 2009 at 10:57 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > or you can put it at a url on your web site and just post a link > > > 2009/11/27 Christian L?schenkohl > > hello >> >> sorry, for my late reply >> my core debugging was at info not at debug, now it's changed and i have >> the log needed >> >> i'm sorry but pastebin doesn't work (it seems that my trace was to big) >> http://pastebin.freeswitch.org/11305 says "Query failure: Got a packet >> bigger than 'max_allowed_packet' bytes" >> >> i'll send the logfile personal to you, hope you don't dislike this >> >> br >> >> On 2009-11-24 01:48, Anthony Minessale wrote: >> > You forgot to set freeswitch to debug loglevel >> > >> > You need both of the following: >> > >> > console loglevel debug >> > sofia profile internal siptrace on >> > >> > >> > >> > >> > 2009/11/23 Christian L?schenkohl > > > >> > >> > sorry about wasting your time (wasn't my intent) >> > >> > the log is at http://pastebin.freeswitch.org/11240 >> > i called 5214448370068 (also other calls are in the log) >> > >> > they now have changed 180 to 183 on the sonus, but makes no >> > difference here >> > >> > br >> > >> > On 2009-11-23 22:07, Anthony Minessale wrote: >> > > do you have the ringback variable set on the channel? >> > > if so it will cause 180 to attempt to play inband ringback >> indication >> > > >> > > I have nothing left to say because I asked for the whole log with >> the >> > > siptrace enables not just 5 lines of it. >> > > If you still want help, give me the log to examine and I will >> > tell you >> > > what your problem is. >> > > >> > > >> > > >> > > 2009/11/23 Christian L?schenkohl >> > > > >> > > > > >> >> > > >> > > thany ou for your answer >> > > >> > > we use g729 on all our other connections in passthrough mode >> > and it >> > > also doesn't work with alaw. >> > > so i don't think it's related to this. >> > > >> > > br >> > > >> > > >> > > On 2009-11-23 20:48, Brian West wrote: >> > > > Well its also G729 so I suspect you don't have G729 >> > > > >> > > > /b >> > > > >> > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: >> > > > >> > > >> hi >> > > >> >> > > >> our freeswitch server has to talk to a sonus ip-switch >> > > >> when we want to setup a call we do get a "100 Trying" and then >> a >> > > >> "180 Ringing" >> > > >> within the "180 Ringing" we get a sdp with "a=sendonly" then >> our >> > > >> freeswitch >> > > >> quits with a CANCEL message. >> > > >> i simply don't get why our freeswitch aborts the session - i >> think >> > > >> it would work >> > > >> if no "a=sendonly" would be present in the sdp. >> > > >> >> > > >> my technical contact doesn't want to switch 180 to 183 on the >> > sonus >> > > >> side - this would >> > > >> also work (i think). in fact he says that 180 ringing is >> vaild, he >> > > >> isn't that wrong in >> > > >> this case. >> > > >> >> > > >> our freeswitch works in proxy mode, we do use trunk 15396 >> > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 >> > > >> >> > > >> 92.63.208.36 - freeswitch >> > > >> 38.105.229.100 - sonus >> > > >> >> > > >> br >> > > >> >> > > >> -- >> > > >> Ing. Christian L?schenkohl >> > > >> Technische Leitung, Forschung& Entwicklung VoIP >> > > >> >> > > >> xpirio >> > > >> Telekommunikation& Service GmbH >> > > >> Lakeside B04 >> > > >> 9020 Klagenfurt >> > > >> Austria >> > > >> >> > > >> T +43 (0) 5 77 11 - 1000 >> > > >> F +43 (0) 5 77 11 - 1002 >> > > >> E christian.loeschenkohl at xpirio.com >> > >> > > > > > >> > > >> >> > > >> _______________________________________________ >> > > >> FreeSWITCH-users mailing list >> > > >> FreeSWITCH-users at lists.freeswitch.org >> > >> > > > > > >> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > > >> users >> > > >> http://www.freeswitch.org >> > > > >> > > > >> > > > _______________________________________________ >> > > > FreeSWITCH-users mailing list >> > > > FreeSWITCH-users at lists.freeswitch.org >> > >> > > > > > >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > > >> > > >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > > http://www.freeswitch.org >> > > >> > > -- >> > > Ing. Christian L?schenkohl >> > > Technische Leitung, Forschung & Entwicklung VoIP >> > > >> > > xpirio >> > > Telekommunikation & Service GmbH >> > > Lakeside B04 >> > > 9020 Klagenfurt >> > > Austria >> > > >> > > T +43 (0) 5 77 11 - 1000 >> > > F +43 (0) 5 77 11 - 1002 >> > > E christian.loeschenkohl at xpirio.com >> > >> > > > > > >> > > >> > > _______________________________________________ >> > > FreeSWITCH-users mailing list >> > > FreeSWITCH-users at lists.freeswitch.org >> > >> > > > > > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > > >> > > >> > > >> > > >> > > -- >> > > Anthony Minessale II >> > > >> > > FreeSWITCH http://www.freeswitch.org/ >> > > ClueCon http://www.cluecon.com/ >> > > Twitter: http://twitter.com/FreeSWITCH_wire >> > > >> > > AIM: anthm >> > > MSN:anthony_minessale at hotmail.com >> > >> > >> > > >> > >> >> >> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > >> > >> > > >> > >> >> >> > > IRC: irc.freenode.net >> > #freeswitch >> > > >> > > FreeSWITCH Developer Conference >> > > sip:888 at conference.freeswitch.org >> > >> > >> > > >> > >> >> >> > > iax:guest at conference.freeswitch.org/888 >> > >> > > >> > > googletalk:conf+888 at conference.freeswitch.org >> > >> > >> > > >> > >> >> >> > > pstn:213-799-1400 >> > > >> > > >> > > >> > >> ------------------------------------------------------------------------ >> > > >> > > _______________________________________________ >> > > FreeSWITCH-users mailing list >> > > FreeSWITCH-users at lists.freeswitch.org >> > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > >> > -- >> > Ing. Christian L?schenkohl >> > Technische Leitung, Forschung & Entwicklung VoIP >> > >> > xpirio >> > Telekommunikation & Service GmbH >> > Lakeside B04 >> > 9020 Klagenfurt >> > Austria >> > >> > T +43 (0) 5 77 11 - 1000 >> > F +43 (0) 5 77 11 - 1002 >> > E christian.loeschenkohl at xpirio.com >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > >> > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > >> > >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> > googletalk:conf+888 at conference.freeswitch.org >> > >> > >> > pstn:213-799-1400 >> > >> > >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> -- >> Ing. Christian L?schenkohl >> Technische Leitung, Forschung & Entwicklung VoIP >> >> xpirio >> Telekommunikation & Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenkohl at xpirio.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/14f15f32/attachment-0001.html From anthony.minessale at gmail.com Fri Nov 27 11:19:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 13:19:31 -0600 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> Message-ID: <191c3a030911271119k3f38a343k8351b121275580b9@mail.gmail.com> I told you to make a new separate inbound connection back to the server from your script, do not use the same one thta was tethered to the call because its too late to use that one. Why do I have to answer you twice? On Thu, Nov 26, 2009 at 3:27 AM, lakshmanan ganapathy wrote: > Hi, Any help or suggestion regarding my previous post. Especially > > > "I also noted that, if I don't receive any events, especially > "SERVER_DISCONNECTED", then the connection is in established state, but once > I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it > correct??" > Here is the program by which I confirmed the above! > > > require ESL; > use IO::Socket::INET; > > my $ip = "192.168.1.222"; > my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', > Proto => 'tcp', Listen => 2, Reuse => 1 ); > die "Could not create socket: $!\n" unless $sock; > my $con; > my $type = "user/"; > > for(;;) { > # wait for any client to connect, a new client will get connected > when a new call comes in the dialplan. > > my $new_sock = $sock->accept(); > # Do fork and let the parent to wait for more clients. > > my $pid = fork(); > if ($pid) { > close($new_sock); > next; > } > # Extract the host of the client. > > my $host = $new_sock->sockhost(); > # file descriptor for the socket. > > my $fd = fileno($new_sock); > print "Host name is $host\n"; > # Create object for the ESL connection package to access the ESL > functions. > > $con = new ESL::ESLconnection($fd); > # Gets the info about this channel. > > my $info = $con->getInfo(); > my $uuid = $info->getHeader("unique-id"); > printf "Connected call %s, from %s to %s\n", $uuid, > $info->getHeader("caller-caller-id-number"), > $info->getHeader("caller-destination-number"); > > # Answer the channel. > $con->execute("answer"); > # Set the event lock to tell the FS to execute the instructions in > the given order. > $con->setEventLock("true"); > # Play a file & Get the personal number from the user. > > $con->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); > $con->execute("hangup"); > > while($con->connected()) > { > my $e=$con->recvEvent(); > my $ename=$e->getHeader("Event-Name"); > print $e->serialize(); > print "$ename\n"; > print "Connection exists\n"; > sleep(1); > > } > print > "Bye\n------------------------------------------------------------------\n"; > close($new_sock); > } > I've not registered for any events. > In the above program I'm receiving the SERVER_DISCONNECTED event. > Output when receiving event: > Host name is 192.168.1.222 > Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 > Event-Name: SERVER_DISCONNECTED > > SERVER_DISCONNECTED > Connection exists > Bye > > When I comment the recvEvent line, I got the following output. > > Host name is 192.168.1.222 > Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 > Connection exists > Connection exists > Connection exists > Connection exists > Connection exists > > > > On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> I've tried the following program as per the suggestion that you've told. >> But it seems, no success. Once the connection is closed, I created a new >> connection and I send originate to originate a new call. But it is not >> working. >> >> require ESL; >> use IO::Socket::INET; >> use Data::Dumper; >> >> my $ip = "192.168.1.222"; >> my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', >> Proto => 'tcp', Listen => 2, Reuse => 1 ); >> die "Could not create socket: $!\n" unless $sock; >> >> my $make_call; >> my $con; >> my $type = "user/"; >> >> for(;;) { >> my $new_sock = $sock->accept(); >> my $pid = fork(); >> if ($pid) { >> close($new_sock); >> next; >> } >> my $host = $new_sock->sockhost(); >> my $fd = fileno($new_sock); >> $con = new ESL::ESLconnection($fd); >> my $info = $con->getInfo(); >> my $uuid = $info->getHeader("unique-id"); >> printf "Connected call %s, from %s to %s\n", $uuid, >> $info->getHeader("caller-caller-id-number"), >> $info->getHeader("caller-destination-number"); >> >> $con->filter("Unique-Id", $uuid); >> $con->events("plain", "all"); >> $con->execute("answer"); >> $con->setEventLock("true"); >> my $number=$con->execute("read","2 4 >> /usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav >> accnt_number 5000 #"); >> while($con->connected()) >> { >> my $e=$con->recvEvent(); >> my $ename=$e->getHeader("Event-Name"); >> my $app=$e->getHeader("Application"); >> if($ename eq "CHANNEL_EXECUTE_COMPLETE" and $app eq >> "read") >> { >> my $num=$e->getHeader("variable_accnt_number"); >> print "$num\n"; >> $con->execute("hangup"); >> } >> } >> if(!$con->connected()) >> { >> print "Connection not exists\n"; >> $con = new ESL::ESLconnection($fd); >> $con->api("originate","user/1000 &park()"); >> print "Hai\n"; >> } >> print >> "Bye\n------------------------------------------------------------------\n"; >> close($new_sock); >> } >> Output: >> Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097 >> 1000 >> Connection not exists >> Hai >> Bye >> ------------------------------------------------------------------ >> The freeswitch log is in >> http://pastebin.freeswitch.org/11258 >> >> I also noted that, if I don't receive any events, especially >> "SERVER_DISCONNECTED", then the connection is in established state, but once >> I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it >> correct?? >> >> >> >> >> >> On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> or open a new outbound connection at the end of your script so you can >>> send your originate command. >>> Since the channel hanging up will close your existing connection since >>> it's only an outbound single session socket. >>> >>> >>> On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins wrote: >>> >>>> >>>> >>>> On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < >>>> lakindia89 at gmail.com> wrote: >>>> >>>>> Hi, >>>>> I'm using perl ESL to control the call in freeswitch. >>>>> I'm having the following scenario, but not able to get it right. >>>>> >>>>> Dialplan: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 1. User A calls to an extention (1000). >>>>> 2. My ESL program will be running, and it answers the call. >>>>> 3. Then the program will get a number from the user. >>>>> 4. It will hangup the call. >>>>> 5. The program has to call to the number that was given by the user. >>>>> >>>>> In the above scenario, I was able to do until the 4th step. After >>>>> hangup the call, if I say originate it is not working. >>>>> Any ideas on how to do this in ESL. >>>>> >>>>> >>>> I want to make sure I understand what the script is supposed to be >>>> doing. The caller will key in a phone number to your script and your script >>>> will collect those digits. The script will then hangup on the caller and >>>> originate a completely new call? Perhaps you could use sched_api to schedule >>>> a new originate command for a few seconds into the future and then hangup? >>>> -MC >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/360a2aad/attachment.html From eman at chabotel.com Fri Nov 27 11:28:41 2009 From: eman at chabotel.com (eman) Date: Fri, 27 Nov 2009 14:28:41 -0500 Subject: [Freeswitch-users] Connecting Multiple domains In-Reply-To: <4B0FB4BC.3090204@greatiam.com> References: <4B0FB4BC.3090204@greatiam.com> Message-ID: <164a9ab00911271128y76ac22a5q63233475f17c4a94@mail.gmail.com> check out http://wiki.freeswitch.org/wiki/Multi-tenant On Fri, Nov 27, 2009 at 6:15 AM, Otis wrote: > > Could someone please direct me to a link for connecting multiple say 2 > domains each with their own FS server. > > Thanks > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/e0e86311/attachment-0001.html From eman at chabotel.com Fri Nov 27 11:31:27 2009 From: eman at chabotel.com (eman) Date: Fri, 27 Nov 2009 14:31:27 -0500 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> Message-ID: <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> Check out the Linksys SPA2102 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > The only ATA mentioned on the WIKI that supports TLS/SRTP is the > Grandstream HandyTone 503. But, again according to the wiki, that > doesn't seem to behave to well with TLS ... > > On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: > > Mark Campbell-Smith wrote: > >> Does the SPA3102 support TLS or only SRTP? > > > > I don't know, but supporting only SRTP would be ridiculous, since the > keys > > would then be transmitted in the clear and therefore amenable to > interception. > > SRTP requires the SIP channel to be encrypted by TLS in order to be > secure. > > ZRTP, on the other hand, doesn't have this limitation: it works entirely > in > > RTP. > > > > I would be rather surprised were a hardware manufacturer to implement > SRTP > > without TLS for the SIP traffic. On the other hand, we've seen often in > this > > forum that some manufacturers are really clueless... > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/5e00b363/attachment.html From anthony.minessale at gmail.com Fri Nov 27 11:32:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 13:32:42 -0600 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <23f91030911251735r3215a344h279a3f8589d5ff85@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <191c3a030911251319g60cdd5a3t33a82a560faf7a2b@mail.gmail.com> <23f91030911251735r3215a344h279a3f8589d5ff85@mail.gmail.com> Message-ID: <191c3a030911271132w1dd8efe1x5fbda139d197fa99@mail.gmail.com> If you want to go to that much trouble consider http://www.orecx.com/web/ They snoop the RTP from an entirely different box and record your calls for you. On Wed, Nov 25, 2009 at 7:35 PM, Seven Du wrote: > Yeah, that's why I had to record to two files(read&write) and need to mix > together by using sox. Do you only try to using PCMU to save CPU power > matt? As Anthony said, the difference can be ignored. And you also need to > take extra effort to make sure transcoding will not happen on a > conversation. > > But it maybe useful for expensive codecs like g729, iLBC, speex etc for > recording heavy scenarios. I'd like to take a look if there is a 5k bounty > ;) > > 2009/11/26 Anthony Minessale > > The processor power saved is negligible between PCMU and raw PCM and not >> worth the fuss. >> If you didn't decode the audio first you would not be able to mix the >> stream to produce a single file. >> So if we went to the trouble of making native media bugs to be able to do >> that you could barely use them so it would not be worth the 5k or more >> bounty to develop that functionality. >> >> >> >> On Sun, Nov 22, 2009 at 12:48 PM, Matthew Fong wrote: >> >>> I'm trying to conserve processor power by recording in native file >>> format, PCMU in my case. It works great with the following line >>> >>> session:execute("record", >>> "/tmp/my_recording."..session:getVariable("read_codec")); >>> >>> however it fails to work with >>> >>> session:execute("record_session", >>> "/tmp/my_recording."..session:getVariable("read_codec")); >>> or >>> record = api:execute("sched_api", '+1 none uuid_record >>> '..session:getVariable("uuid")..' start >>> /tmp/my_recording.'..session:getVariable("read_codec")); >>> >>> Why is it that it works with record, but not with record_session or >>> uuid_record? Is there something I'm over looking? In the latter two the >>> consul reports >>> >>> 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File >>> [/tmp/my_recording.PCMU] 8000hz >>> >>> as if it's recording, but /tmp/my_recording.PCMU never shows up. However >>> if I change it to .wav instead of .PCMU it works. Any ideas? >>> >>> --matt >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/3ba0be37/attachment.html From anthony.minessale at gmail.com Fri Nov 27 13:11:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 15:11:48 -0600 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B0F1565.6060909@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> Message-ID: <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> Did you check the 2 replies that told you you need aliases in your sofia profile to translate the domain found in your message_waiting to the right profile? Both Brian and Mike answered you. On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX wrote: > I tried now with phones directly attached to the freeswitch (without an > OpenSIPS in between). I also added the alias. But the behaviour is as > before: > No notify message from freeswitch, neither after register nor after a > voicemail is recorded. > > Best regards > Peter > Brian West schrieb: > > Yes an alias will be required for every domain you run on the profile > > so it can find it. > > > > /b > > > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > > > > >> Try an alias on the sip profile. > >> > >> Mike > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/afe2bc87/attachment.html From mike at jerris.com Fri Nov 27 13:13:20 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Nov 2009 16:13:20 -0500 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B0F1565.6060909@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> Message-ID: <95C676DC-A619-4D00-B039-F59E3D74C059@jerris.com> Does the alias you added match the one that you saw in the event? The alias is 100% for sure the fix for this issue, please check again. Mike On Nov 26, 2009, at 6:55 PM, Peter P GMX wrote: > I tried now with phones directly attached to the freeswitch (without an > OpenSIPS in between). I also added the alias. But the behaviour is as > before: > No notify message from freeswitch, neither after register nor after a > voicemail is recorded. > > Best regards > Peter > Brian West schrieb: >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >> >>> Try an alias on the sip profile. >>> >>> Mike >>> From anthony.minessale at gmail.com Fri Nov 27 13:22:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 15:22:25 -0600 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> Message-ID: <191c3a030911271322tce11dcy991dc1d668179a76@mail.gmail.com> based on your example past sip1.mydomain.com is the domain in the packet and thus the profile should have an alias for this. Then the user must reside in your sip db with the user 200 and domain sip1.mydomain.com if you dont have this consider the force-register-domain and force-register-db-domain to normalize the host names. On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Did you check the 2 replies that told you you need aliases in your sofia > profile to translate the domain found in your message_waiting to the right > profile? Both Brian and Mike answered you. > > > > > > On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX wrote: > >> I tried now with phones directly attached to the freeswitch (without an >> OpenSIPS in between). I also added the alias. But the behaviour is as >> before: >> No notify message from freeswitch, neither after register nor after a >> voicemail is recorded. >> >> Best regards >> Peter >> Brian West schrieb: >> > Yes an alias will be required for every domain you run on the profile >> > so it can find it. >> > >> > /b >> > >> > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> > >> > >> >> Try an alias on the sip profile. >> >> >> >> Mike >> >> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/612afe7b/attachment.html From lists at redbonez.net Fri Nov 27 17:10:33 2009 From: lists at redbonez.net (Adam Ford) Date: Fri, 27 Nov 2009 18:10:33 -0700 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: <4B101F03.5090802@greatiam.com> References: <4B101F03.5090802@greatiam.com> Message-ID: <01f601ca6fc7$975e11a0$c61a34e0$@net> FusionPBX, FreePBX v3, and wikiPBX are the three that I have found in the past. However they all seem to be in the early stages of development, and not 100% stable. I can say this for sure about FreePBX and FusionPBX, but I have not actually tried wikiPBX. -AF -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Otis Sent: Friday, November 27, 2009 11:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch admin GUI Hi I am no sure but read up on fusionpbx. I asked the same question and someone pointed me to that. check web site Regards Samuel Mukoti wrote: >
Hi, > > Any recommendations for apps that can I use ontop of freeswitch as a > GUI manager, to manage extensions, queues, ivr, and dialplans? > > Thanks > > Sam > > > On 27 Nov,2009, at 5:19 PM, > freeswitch-users-request at lists.freeswitch.org wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Fri, 27 Nov 2009 16:19:03 +0100 >> From: Leon de Rooij >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset="windows-1252" >> >> Are you using the myodbc 3.51.18 version or higher ? >> >> I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to >> upgrade from jaunty.. >> >> regards, >> >> Leon >> >> >> On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: >> >>> Thanks. But when I made these entries in /etc/odbc.ini and rebooted? >>> >>> [freeswitch] >>> Driver = MySQL >>> SERVER = 127.0.0.1 >>> PORT = 4040 >>> DATABASE = mydb >>> OPTIONS = 67108864 >>> >>> ?I still get FS complaining with this. >>> >>> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 >>> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched >>> statements!#012If you are using mysql, make sure you are using >>> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS >>> >>> FreeSWITCH>version >>> FreeSWITCH Version 1.0.trunk (15660) >>> >>> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 >>> EST 2009 x86_64 x86_64 x86_64 GNU/Linux >>> >>> From /etc/odbcinst.ini >>> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so >>> Setup = /usr/lib64/libodbcmyS.so >>> >>> Is this a FS issue ? or an issue with mysql odbc? Any insight >>> would be great. >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Leon de Rooij >>> Sent: Friday, November 27, 2009 3:37 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >>> >>> There's a little info here on how to enable it with odbc: >>> >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 >>> >>> regards, >>> >>> Leon >>> >>> >>> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: >>> >>> >>> >>> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris >>> wrote: >>> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- >>> news-3-51-18.html >>> >>> MySQL Connector/ODBC now supports batched statements. In order to >>> enable >>> cached statement support you must switch enable the batched >>> statement option (FLAG_MULTI_STATEMENTS, >>> 67108864, or Allow multiple statements >>> within a GUI configuration). Be aware that batched statements >>> create an increased chance of SQL injection attacks and you >>> must >>> ensure that your application protects against this scenario. >>> (Bug#7445) >>> >>> >>> so, is this the right patch ? >>> >>> http://bugs.mysql.com/file.php?id=6994 >>> >>> >>> T. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/ 9c86b324/attachment.html >> >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 209 >> ************************************************* > > >
> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From tleyden at branchcut.com Fri Nov 27 18:50:42 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Fri, 27 Nov 2009 18:50:42 -0800 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: <01f601ca6fc7$975e11a0$c61a34e0$@net> References: <4B101F03.5090802@greatiam.com> <01f601ca6fc7$975e11a0$c61a34e0$@net> Message-ID: Well now might be a good time to give wikipbx a spin, because about 15 minutes ago we just released the second major release .. version 0.8 Here are the official release notes: --- WikiPBX has been converted from being based on Twisted.web2, a somewhat exotic webserver, to running as a mod_wsgi app within Apache2. Actually it can run under any webserver that support mod_wsgi. Additionally, the multi-tenancy has been changed from sip profile based multi-tenancy, which is not really "the normal way" to do this .. to the standard approach of domain based multi-tenancy. Along with that comes the ability to manage sip profiles from the GUI. A lot of security enhancements have been added, it is possible to force sip profile wide authorization, as well as per-extension dialplan security -- checking a flag in the dialplan will make it public (for anyone to access) or private (only registered users). The dialplan security is overlayed on top of the sip profile security .. and sip profile security takes precedence. (sip profile security is done by freeswitch whereas per-extension dialplan security is done by wikipbx) Gateways can now be per-tenant or shared among all tenants, and can be assigned to any sip profile on the system. Mod_voicemail now works with wikipbx, and is easy to configure. No GUI support yet in terms of visual voicemail. A really simple XML export / import has been added for existing users using version 0.5 to upgrade to version 0.8. Documentation has been completely re-written to reflect all changes in this release. What's still missing? The configuration XML that is served up to FreeSWITCH is really, really old. The only upshot is that they are served from static template files, so you can hack stuff in without having to look at any code. We are planning to fix this by the 1.0 release, slated for November of 2018. See http://wikipbx.subwiki.com/release-notes-0-8 for more details on this release and how to install it. On Fri, Nov 27, 2009 at 5:10 PM, Adam Ford wrote: > FusionPBX, FreePBX v3, and wikiPBX are the three that I have found in the > past. However they all seem to be in the early stages of development, and > not 100% stable. I can say this for sure about FreePBX and FusionPBX, but I > have not actually tried wikiPBX. > > -AF > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Otis > Sent: Friday, November 27, 2009 11:49 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Freeswitch admin GUI > > Hi > > I am no sure but read up on fusionpbx. I asked the same question and > someone pointed me to that. > check web site > > Regards > > > > Samuel Mukoti wrote: > >
Hi, > > > > Any recommendations for apps that can I use ontop of freeswitch as a > > GUI manager, to manage extensions, queues, ivr, and dialplans? > > > > Thanks > > > > Sam > > > > > > On 27 Nov,2009, at 5:19 PM, > > freeswitch-users-request at lists.freeswitch.org wrote: > > > >> Send FreeSWITCH-users mailing list submissions to > >> freeswitch-users at lists.freeswitch.org > >> > >> To subscribe or unsubscribe via the World Wide Web, visit > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> or, via email, send a message with subject or body 'help' to > >> freeswitch-users-request at lists.freeswitch.org > >> > >> You can reach the person managing the list at > >> freeswitch-users-owner at lists.freeswitch.org > >> > >> When replying, please edit your Subject line so it is more specific > >> than "Re: Contents of FreeSWITCH-users digest..." > >> > >> > >> Today's Topics: > >> > >> 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) > >> > >> > >> ---------------------------------------------------------------------- > >> > >> Message: 1 > >> Date: Fri, 27 Nov 2009 16:19:03 +0100 > >> From: Leon de Rooij > >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> Content-Type: text/plain; charset="windows-1252" > >> > >> Are you using the myodbc 3.51.18 version or higher ? > >> > >> I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to > >> upgrade from jaunty.. > >> > >> regards, > >> > >> Leon > >> > >> > >> On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: > >> > >>> Thanks. But when I made these entries in /etc/odbc.ini and rebooted? > >>> > >>> [freeswitch] > >>> Driver = MySQL > >>> SERVER = 127.0.0.1 > >>> PORT = 4040 > >>> DATABASE = mydb > >>> OPTIONS = 67108864 > >>> > >>> ?I still get FS complaining with this. > >>> > >>> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 > >>> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched > >>> statements!#012If you are using mysql, make sure you are using > >>> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS > >>> > >>> FreeSWITCH>version > >>> FreeSWITCH Version 1.0.trunk (15660) > >>> > >>> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 > >>> EST 2009 x86_64 x86_64 x86_64 GNU/Linux > >>> > >>> From /etc/odbcinst.ini > >>> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so > >>> Setup = /usr/lib64/libodbcmyS.so > >>> > >>> Is this a FS issue ? or an issue with mysql odbc? Any insight > >>> would be great. > >>> > >>> -----Original Message----- > >>> From: freeswitch-users-bounces at lists.freeswitch.org > >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org > >>> ] On Behalf Of Leon de Rooij > >>> Sent: Friday, November 27, 2009 3:37 AM > >>> To: freeswitch-users at lists.freeswitch.org > >>> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > >>> > >>> There's a little info here on how to enable it with odbc: > >>> > >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 > >>> > >>> regards, > >>> > >>> Leon > >>> > >>> > >>> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: > >>> > >>> > >>> > >>> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris > >>> wrote: > >>> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- > >>> news-3-51-18.html > >>> > >>> MySQL Connector/ODBC now supports batched statements. In order to > >>> enable > >>> cached statement support you must switch enable the batched > >>> statement option (FLAG_MULTI_STATEMENTS, > >>> 67108864, or Allow multiple statements > >>> within a GUI configuration). Be aware that batched statements > >>> create an increased chance of SQL injection attacks and you > >>> must > >>> ensure that your application protects against this scenario. > >>> (Bug#7445) > >>> > >>> > >>> so, is this the right patch ? > >>> > >>> http://bugs.mysql.com/file.php?id=6994 > >>> > >>> > >>> T. > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> > >>> http://www.freeswitch.org > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> > >>> http://www.freeswitch.org > >> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > >> > > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/ > 9c86b324/attachment.html > >> > >> > >> ------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> End of FreeSWITCH-users Digest, Vol 41, Issue 209 > >> ************************************************* > > > > > >
> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/d8d05d00/attachment-0001.html From juanbackson at gmail.com Fri Nov 27 21:48:21 2009 From: juanbackson at gmail.com (Juan Backson) Date: Sat, 28 Nov 2009 13:48:21 +0800 Subject: [Freeswitch-users] custom call counter Message-ID: <27c25bc40911272148o10bbcb9fo2566c5e9b64fa261@mail.gmail.com> Hi, Instead of using "show calls count" to obtain the current call count stat, I am writing some C code to increment a counter during on_answer_hook and decrement the counter during on_hangup_hook. It looks like my counter result is very closed to "show calls count" when the traffic is low, like 50 -60. But when traffic is high, like 1000 calls, my counter is showing 30% less. When all calls are finished, my counter becomes 0 again, and that proves that it does not increment/decrement more than it should. Is this normal? Does anyone have any idea why there is such discrepancy? thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/21dbd2b4/attachment.html From mike at jerris.com Fri Nov 27 23:01:47 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 28 Nov 2009 02:01:47 -0500 Subject: [Freeswitch-users] custom call counter In-Reply-To: <27c25bc40911272148o10bbcb9fo2566c5e9b64fa261@mail.gmail.com> References: <27c25bc40911272148o10bbcb9fo2566c5e9b64fa261@mail.gmail.com> Message-ID: <028E32A6-1B83-4B74-8ABD-6D4F317B7325@jerris.com> It depends on the timing of when your increment and decrement are vs when the sql calls to push the events into the tables that are used for show calls are. Also, the sql calls are batched and queued causing a little delay (less than a second). If your doing a lot of short lived calls there is sure to be timing discrepancy. I am sure there is even more discrepancy if you look at the output of status which shows the current number of sessions (those are individual call legs) as that information is a little more real time. Mike On Nov 28, 2009, at 12:48 AM, Juan Backson wrote: > Hi, > > Instead of using "show calls count" to obtain the current call count stat, I am writing some C code to increment a counter during on_answer_hook and decrement the counter during on_hangup_hook. > > It looks like my counter result is very closed to "show calls count" when the traffic is low, like 50 -60. But when traffic is high, like 1000 calls, my counter is showing 30% less. > > When all calls are finished, my counter becomes 0 again, and that proves that it does not increment/decrement more than it should. > > Is this normal? Does anyone have any idea why there is such discrepancy? > > thanks, > jb > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From talk2ram at gmail.com Sat Nov 28 00:51:51 2009 From: talk2ram at gmail.com (ram) Date: Sat, 28 Nov 2009 14:21:51 +0530 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: References: <4B101F03.5090802@greatiam.com> <01f601ca6fc7$975e11a0$c61a34e0$@net> Message-ID: On Sat, Nov 28, 2009 at 8:20 AM, Traun Leyden wrote: > > Well now might be a good time to give wikipbx a spin, because about 15 > minutes ago we just released the second major release .. version 0.8 > > Here are the official release notes: > > --- > > WikiPBX has been converted from being based on Twisted.web2, a somewhat > exotic webserver, to running as a mod_wsgi app within Apache2. Actually it > can run under any webserver that support mod_wsgi. > > Additionally, the multi-tenancy has been changed from sip profile based > multi-tenancy, which is not really "the normal way" to do this .. to the > standard approach of domain based multi-tenancy. Along with that comes the > ability to manage sip profiles from the GUI. > > A lot of security enhancements have been added, it is possible to force sip > profile wide authorization, as well as per-extension dialplan security -- > checking a flag in the dialplan will make it public (for anyone to access) > or private (only registered users). The dialplan security is overlayed on > top of the sip profile security .. and sip profile security takes > precedence. (sip profile security is done by freeswitch whereas > per-extension dialplan security is done by wikipbx) > > Gateways can now be per-tenant or shared among all tenants, and can be > assigned to any sip profile on the system. > > Mod_voicemail now works with wikipbx, and is easy to configure. No GUI > support yet in terms of visual voicemail. > > A really simple XML export / import has been added for existing users using > version 0.5 to upgrade to version 0.8. > > Documentation has been completely re-written to reflect all changes in this > release. > > What's still missing? The configuration XML that is served up to > FreeSWITCH is really, really old. The only upshot is that they are served > from static template files, so you can hack stuff in without having to look > at any code. We are planning to fix this by the 1.0 release, slated for > November of 2018. > > See http://wikipbx.subwiki.com/release-notes-0-8 for more details on this > release and how to install it. > Hi does wikipbx have any IRC channel Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/16091786/attachment.html From talk2ram at gmail.com Sat Nov 28 00:57:17 2009 From: talk2ram at gmail.com (ram) Date: Sat, 28 Nov 2009 14:27:17 +0530 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <4B0F8EC2.2080609@greatiam.com> References: <221275.23339.qm@web56403.mail.re3.yahoo.com> <4B0E7384.5010809@greatiam.com> <92e7d2090911260717j11ffad78kdd11b1c87dfd87be@mail.gmail.com> <4B0F8EC2.2080609@greatiam.com> Message-ID: On Fri, Nov 27, 2009 at 2:03 PM, Otis wrote: > Yes. I ventured to use that and got some error in connecting to the > mysql database. Will try with the default sqlite before getting > adventurous again. > > Hi download latest RC5 it has install wizard automatically create database ( sqllite/mysql/pgsql) Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/5e2c247e/attachment.html From grevenx at me.com Sat Nov 28 02:58:05 2009 From: grevenx at me.com (=?iso-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Sat, 28 Nov 2009 11:58:05 +0100 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: References: <4B101F03.5090802@greatiam.com> <01f601ca6fc7$975e11a0$c61a34e0$@net> Message-ID: On 28. nov. 2009, at 03.50, Traun Leyden wrote: > What's still missing? The configuration XML that is served up to FreeSWITCH is really, really old. The only upshot is that they are served from static template files, so you can hack stuff in without having to look at any code. We are planning to fix this by the 1.0 release, slated for November of 2018. Oh boy, I really DO hope FreeSWITCH will still be alive and well in 2018!! Best regards, Even Andr? From simon.woodhead at me.com Sat Nov 28 09:47:04 2009 From: simon.woodhead at me.com (Simon Woodhead) Date: Sat, 28 Nov 2009 17:47:04 +0000 Subject: [Freeswitch-users] Accessing custom SIP headers Message-ID: <86b72a770911280947m143f40aah640ff8e56ed08950@mail.gmail.com> Hi folks, I'm hoping someone can help me get at custom headers in the dial-plan. I've read about X- headers being accessible but need to get at some X_ headers passed through from a proxy. Reading the info app docs, the X shouldn't actually matter but no matter which way I try I always seem to get a null result. An example header in an INVITE is: X_ACCOUNTCODE: XXXXXX. I've tried the following dial-plan structures hoping one might work but none do: Any help would be much appreciated. Thanks, Simon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/6600985b/attachment.html From christian.loeschenkohl at xpirio.com Sat Nov 28 11:24:31 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Sat, 28 Nov 2009 20:24:31 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <191c3a030911271103s462910acu96da189b4064e63e@mail.gmail.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> <4B0B005E.4080202@xpirio.com> <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> <4B0F8C3B.4000800@xpirio.com> <191c3a030911270857j697738f9n99e9e5bb1b71c38@mail.gmail.com> <191c3a030911271103s462910acu96da189b4064e63e@mail.gmail.com> Message-ID: <4B1178EF.3060604@xpirio.com> works now thank you very much On 2009-11-27 20:03, Anthony Minessale wrote: > please update to latest trunk 15698 or greater and re-test. > The 183 from the provider had a sendonly attr that tricked the proxy > code into thinking it was a hold/unhold operation. > > > On Fri, Nov 27, 2009 at 10:57 AM, Anthony Minessale > > wrote: > > or you can put it at a url on your web site and just post a link > > > 2009/11/27 Christian L?schenkohl > > > hello > > sorry, for my late reply > my core debugging was at info not at debug, now it's changed and > i have the log needed > > i'm sorry but pastebin doesn't work (it seems that my trace was > to big) > http://pastebin.freeswitch.org/11305 says "Query failure: Got a > packet bigger than 'max_allowed_packet' bytes" > > i'll send the logfile personal to you, hope you don't dislike this > > br > > On 2009-11-24 01:48, Anthony Minessale wrote: > > You forgot to set freeswitch to debug loglevel > > > > You need both of the following: > > > > console loglevel debug > > sofia profile internal siptrace on > > > > > > > > > > 2009/11/23 Christian L?schenkohl > > > >> > > > > sorry about wasting your time (wasn't my intent) > > > > the log is at http://pastebin.freeswitch.org/11240 > > i called 5214448370068 (also other calls are in the log) > > > > they now have changed 180 to 183 on the sonus, but makes no > > difference here > > > > br > > > > On 2009-11-23 22:07, Anthony Minessale wrote: > > > do you have the ringback variable set on the channel? > > > if so it will cause 180 to attempt to play inband ringback > indication > > > > > > I have nothing left to say because I asked for the whole > log with the > > > siptrace enables not just 5 lines of it. > > > If you still want help, give me the log to examine and I will > > tell you > > > what your problem is. > > > > > > > > > > > > 2009/11/23 Christian L?schenkohl > > > > > > > > > > >>> > > > > > > thany ou for your answer > > > > > > we use g729 on all our other connections in passthrough > mode > > and it > > > also doesn't work with alaw. > > > so i don't think it's related to this. > > > > > > br > > > > > > > > > On 2009-11-23 20:48, Brian West wrote: > > > > Well its also G729 so I suspect you don't have G729 > > > > > > > > /b > > > > > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > > > > > >> hi > > > >> > > > >> our freeswitch server has to talk to a sonus ip-switch > > > >> when we want to setup a call we do get a "100 Trying" > and then a > > > >> "180 Ringing" > > > >> within the "180 Ringing" we get a sdp with "a=sendonly" > then our > > > >> freeswitch > > > >> quits with a CANCEL message. > > > >> i simply don't get why our freeswitch aborts the session > - i think > > > >> it would work > > > >> if no "a=sendonly" would be present in the sdp. > > > >> > > > >> my technical contact doesn't want to switch 180 to 183 > on the > > sonus > > > >> side - this would > > > >> also work (i think). in fact he says that 180 ringing is > vaild, he > > > >> isn't that wrong in > > > >> this case. > > > >> > > > >> our freeswitch works in proxy mode, we do use trunk 15396 > > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > > > >> > > > >> 92.63.208.36 - freeswitch > > > >> 38.105.229.100 - sonus > > > >> > > > >> br > > > >> > > > >> -- > > > >> Ing. Christian L?schenkohl > > > >> Technische Leitung, Forschung& Entwicklung VoIP > > > >> > > > >> xpirio > > > >> Telekommunikation& Service GmbH > > > >> Lakeside B04 > > > >> 9020 Klagenfurt > > > >> Austria > > > >> > > > >> T +43 (0) 5 77 11 - 1000 > > > >> F +43 (0) 5 77 11 - 1002 > > > >> E christian.loeschenkohl at xpirio.com > > > > > > > > > >> > > > >> > > > >> _______________________________________________ > > > >> FreeSWITCH-users mailing list > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > >> > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > > >> users > > > >> http://www.freeswitch.org > > > > > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > >> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > -- > > > Ing. Christian L?schenkohl > > > Technische Leitung, Forschung & Entwicklung VoIP > > > > > > xpirio > > > Telekommunikation & Service GmbH > > > Lakeside B04 > > > 9020 Klagenfurt > > > Austria > > > > > > T +43 (0) 5 77 11 - 1000 > > > F +43 (0) 5 77 11 - 1002 > > > E christian.loeschenkohl at xpirio.com > > > > > > > > > >> > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > > > > > > > >> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > > > > >> > > > IRC: irc.freenode.net > > > #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > > > > > > > >> > > > iax:guest at conference.freeswitch.org/888 > > > > > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > > > > >> > > > pstn:213-799-1400 > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung & Entwicklung VoIP > > > > xpirio > > Telekommunikation & Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From mctch at yahoo.com Sat Nov 28 13:42:35 2009 From: mctch at yahoo.com (Mark Crane) Date: Sat, 28 Nov 2009 13:42:35 -0800 (PST) Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: Message-ID: <150632.44098.qm@web56408.mail.re3.yahoo.com> During the install of FusionPBX if you try to connect to MySQL connection and use 'localhost' it will attempt to use a Unix Socket then throws an error. Instead use 127.0.0.1 then it will actually use TCP connection rather than the UnixSocket connection. This is not a bug in FusionPBX it seems to be just how PHP PDO MySQL handles the connection. Hope this helps. For the release version I will add a little wording suggesting 127.0.0.1 vs localhost for those that have a local MySQL install. Best Regards, Mark J Crane ? --- On Sat, 11/28/09, ram wrote: From: ram Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX To: freeswitch-users at lists.freeswitch.org Date: Saturday, November 28, 2009, 1:57 AM On Fri, Nov 27, 2009 at 2:03 PM, Otis wrote: Yes. I ventured to use that ?and got some error in connecting to the mysql database. Will try with the default sqlite before getting adventurous again. ? Hi ? download latest RC5 ? it has install wizard automatically create database ( sqllite/mysql/pgsql) ? Ram ? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/9d84a520/attachment.html From mctch at yahoo.com Sat Nov 28 13:51:41 2009 From: mctch at yahoo.com (Mark Crane) Date: Sat, 28 Nov 2009 13:51:41 -0800 (PST) Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: <01f601ca6fc7$975e11a0$c61a34e0$@net> Message-ID: <90909.63088.qm@web56405.mail.re3.yahoo.com> FusionPBX is very close to a release. FusionPBX is on the last release candidate 5 before a 1.0 release. Most of the work in the past couple weeks has been to make the install easier. ISO versions will be available in the future. I have multiple businesses already running live on FusionPBX. The project will advance faster the more it is used and the more feedback that is given. Mark J Crane http://www.fusionpbx.com --- On Fri, 11/27/09, Adam Ford wrote: From: Adam Ford Subject: Re: [Freeswitch-users] Freeswitch admin GUI To: freeswitch-users at lists.freeswitch.org Date: Friday, November 27, 2009, 6:10 PM FusionPBX, FreePBX v3, and wikiPBX are the three that I have found in the past. However they all seem to be in the early stages of development, and not 100% stable. I can say this for sure about FreePBX and FusionPBX, but I have not actually tried wikiPBX. -AF -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Otis Sent: Friday, November 27, 2009 11:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch admin GUI Hi I am no sure but read up on fusionpbx. I asked the same question and someone pointed me to that. check web site Regards Samuel Mukoti wrote: >
Hi, > > Any recommendations for apps that can I use ontop of freeswitch as a > GUI manager, to manage extensions, queues, ivr, and dialplans? > > Thanks > > Sam > > > On 27 Nov,2009, at 5:19 PM, > freeswitch-users-request at lists.freeswitch.org wrote: > >> Send FreeSWITCH-users mailing list submissions to >>? ? freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >>? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >>? ? freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >>? ? freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >>???1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Fri, 27 Nov 2009 16:19:03 +0100 >> From: Leon de Rooij >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset="windows-1252" >> >> Are you using the myodbc 3.51.18 version or higher ? >> >> I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to >> upgrade from jaunty.. >> >> regards, >> >> Leon >> >> >> On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: >> >>> Thanks.? But when I made these entries in /etc/odbc.ini and rebooted? >>> >>> [freeswitch] >>> Driver? ? ? ? ? = MySQL >>> SERVER? ? ? ? ? = 127.0.0.1 >>> PORT? ? ? ? ? ? = 4040 >>> DATABASE? ? ? ? = mydb >>> OPTIONS? ? ? ???= 67108864 >>> >>> ?I still get FS complaining with this. >>> >>> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 >>> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched >>> statements!#012If you are using mysql, make sure you are using >>> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS >>> >>> FreeSWITCH>version >>> FreeSWITCH Version 1.0.trunk (15660) >>> >>> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 >>> EST 2009 x86_64 x86_64 x86_64 GNU/Linux >>> >>> From /etc/odbcinst.ini >>> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so >>> Setup = /usr/lib64/libodbcmyS.so >>> >>> Is this a FS issue ?? or an issue with mysql odbc?? Any insight >>> would be great. >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Leon de Rooij >>> Sent: Friday, November 27, 2009 3:37 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >>> >>> There's a little info here on how to enable it with odbc: >>> >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 >>> >>> regards, >>> >>> Leon >>> >>> >>> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: >>> >>> >>> >>> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris >>> wrote: >>> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- >>> news-3-51-18.html >>> >>> MySQL Connector/ODBC now supports batched statements. In order to >>> enable >>>? ? ? ? cached statement support you must switch enable the batched >>>? ? ? ? statement option (FLAG_MULTI_STATEMENTS, >>>? ? ? ? 67108864, or Allow multiple statements >>>? ? ? ? within a GUI configuration). Be aware that batched statements >>>? ? ? ? create an increased chance of SQL injection attacks and you >>> must >>>? ? ? ? ensure that your application protects against this scenario. >>>? ? ???(Bug#7445) >>> >>> >>> so, is this the right patch ? >>> >>> http://bugs.mysql.com/file.php?id=6994 >>> >>> >>> T. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/ 9c86b324/attachment.html >> >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 209 >> ************************************************* > > >
> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/0178e815/attachment.html From john_platts at hotmail.com Sat Nov 28 21:34:24 2009 From: john_platts at hotmail.com (John Platts) Date: Sat, 28 Nov 2009 23:34:24 -0600 Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 Message-ID: I have updated my FreeSWITCH installation to revision 15700. I am experiencing call transfer problems whenever proxy media or bypass media is enabled. When proxy media and bypass media are both disabled, the call transfer does not fail and there are no audio issues. When proxy media mode is enabled, the call stays up after the transfer occurs, but there is no audio flowing on either end of the call. When bypass media mode is enabled, there is no audio flowing on either end of the call, and the call actually gets disconnected. I have collected detailed traces using the TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch command. I have attached a ZIP file named freeswitch-rev15700-traces-112809-2210.zip, which includes the following traces: - freeswitch-rev15700-trace-112809-2210-proxyandbypassoff.txt - A trace with both media proxying and media bypass disabled. The call is being transferred without any problems in this scenario. - freeswitch-rev15700-trace-112809-2210-proxyonandbypassoff.txt - A trace with media proxying enabled and media bypass disabled. Media proxying is enabled for the call legs in this scenario. The call stays up in this scenario, but there is no audio flowing after the transfer completed. In this scenario, FreeSWITCH does not shutdown cleanly, and there is a segmentation violation when FreeSWITCH is terminated. - freeswitch-rev15700-trace-112809-2210-proxyandbypasson.txt - A trace with both media proxying and media bypass enabled. Media bypass is enabled for the call legs in this scenario. The call actually gets dropped and there is no audio after the transfer is completed in this scenario. I have looked over the SIP traces of the failing scenarios. I have caught the following problems in the failing scenarios: - The o= line in SDP descriptors coming from the IP phone contains the private IP address, but the c= line in the SDP descriptors coming from the IP phone contains the public IP address. I have noticed a problem in re-INVITEs being sent from in proxy media and bypass media modes. The c= line in the re-invites contains the private IP address instead of the public IP address. The c= line was modified by a SIP ALG to contain a public IP address, but FreeSWITCH is actually not handling this correctly when calls are transferred. - The wrong codec is being negotiated in re-INVITE to the transferred number in the scenario when media proxying is enabled but media bypass is disabled. - In the scenario where media bypass is used, the re-INVITE actually appears to contain the correct details, and we are receiving the correct responses from our IP to IP gateway, but FreeSWITCH is not handling the media streams properly. Example of SDP descriptor coming from IP phone (with SDP descriptor modified by SIP ALG): v=0 o=- 123576 123576 IN IP4 192.168.1.4 s=- c=IN IP4 173.57.44.212 t=0 0 m=audio 16406 RTP/AVP 18 0 8 2 9 104 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:104 L16/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv Notice that the c= line has the correct public IP address and the m= line containing the correct port. Example of incorrect SDP descriptor being sent by FreeSWITCH in re-INVITES: v=0 o=- 121397 121398 IN IP4 192.168.1.4 s=- c=IN IP4 192.168.1.4 t=0 0 m=audio 16404 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=ptime:20 Note that the c= line contains the wrong IP address, but the m= line contains the correct RTP port. Example of wrong re-INVITE message being sent to the number that the call was being transferred to: INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj Max-Forwards: 69 From: "John Platts" ;tag=c61Drt38KF72m To: ;tag=2B1339E0-1A2C Call-ID: 1c095553-5741-122d-33a8-00185167f91d CSeq: 123615824 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 183 X-FS-Support: update_display Remote-Party-ID: "John Platts" ;party=calling;screen=yes;privacy=off v=0 o=- 123576 123577 IN IP4 192.168.1.4 s=- c=IN IP4 168.75.202.212 t=0 0 m=audio 30186 RTP/AVP 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 Here is the correct re-INVITE for the call that was unsuccessfully transferred (after the transfer was completed): INVITE sip:19729555871 at 168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKgaDHFKZrc06vD Max-Forwards: 16 From: ;tag=BX8mpZj5p6ggS To: ;tag=2B12D184-BEC Call-ID: 15A1F95-DBD611DE-8C95D9DF-3419A306 at 168.75.202.246 CSeq: 123615820 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 222 X-FS-Support: update_display v=0 o=- 121397 121399 IN IP4 192.168.1.4 s=- c=IN IP4 168.75.202.212 t=0 0 m=audio 26106 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 _________________________________________________________________ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch-rev15700-traces-112809-2210.zip Type: application/zip Size: 75260 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/dd970d52/attachment-0001.zip From john_platts at hotmail.com Sun Nov 29 04:51:57 2009 From: john_platts at hotmail.com (John Platts) Date: Sun, 29 Nov 2009 06:51:57 -0600 Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 In-Reply-To: References: Message-ID: To clarify the problem, the invite message is incorrect because comfort noise is being negotiated in the re-invite instead of G.711 or G.729: INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj Max-Forwards: 69 From: "John Platts" ;tag=c61Drt38KF72m To: ;tag=2B1339E0-1A2C Call-ID: 1c095553-5741-122d-33a8-00185167f91d CSeq: 123615824 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 183 X-FS-Support: update_display Remote-Party-ID: "John Platts" ;party=calling;screen=yes;privacy=off v=0 o=- 123576 123577 IN IP4 192.168.1.4 s=- c=IN IP4 168.75.202.212 t=0 0 m=audio 30186 RTP/AVP 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 How do I get it to negotiate G.711, G.729, or other codec instead of comfort noise? Our IP phones, our FXS gateways, and our IP to IP gateways expect G.711, G.729, iLBC (if supported by the endpoints), G.722 (if supported by the endpoints), or G.726 (if supported by the endpoints) be negotiated. ---------------------------------------- > From: john_platts at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Sat, 28 Nov 2009 23:34:24 -0600 > Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 > > > I have updated my FreeSWITCH installation to revision 15700. I am experiencing call transfer problems whenever proxy media or bypass media is enabled. When proxy media and bypass media are both disabled, the call transfer does not fail and there are no audio issues. When proxy media mode is enabled, the call stays up after the transfer occurs, but there is no audio flowing on either end of the call. When bypass media mode is enabled, there is no audio flowing on either end of the call, and the call actually gets disconnected. > > I have collected detailed traces using the TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch command. I have attached a ZIP file named freeswitch-rev15700-traces-112809-2210.zip, which includes the following traces: > - freeswitch-rev15700-trace-112809-2210-proxyandbypassoff.txt - A trace with both media proxying and media bypass disabled. The call is being transferred without any problems in this scenario. > - freeswitch-rev15700-trace-112809-2210-proxyonandbypassoff.txt - A trace with media proxying enabled and media bypass disabled. Media proxying is enabled for the call legs in this scenario. The call stays up in this scenario, but there is no audio flowing after the transfer completed. In this scenario, FreeSWITCH does not shutdown cleanly, and there is a segmentation violation when FreeSWITCH is terminated. > - freeswitch-rev15700-trace-112809-2210-proxyandbypasson.txt - A trace with both media proxying and media bypass enabled. Media bypass is enabled for the call legs in this scenario. The call actually gets dropped and there is no audio after the transfer is completed in this scenario. > > I have looked over the SIP traces of the failing scenarios. > > I have caught the following problems in the failing scenarios: > - The o= line in SDP descriptors coming from the IP phone contains the private IP address, but the c= line in the SDP descriptors coming from the IP phone contains the public IP address. I have noticed a problem in re-INVITEs being sent from in proxy media and bypass media modes. The c= line in the re-invites contains the private IP address instead of the public IP address. The c= line was modified by a SIP ALG to contain a public IP address, but FreeSWITCH is actually not handling this correctly when calls are transferred. > - The wrong codec is being negotiated in re-INVITE to the transferred number in the scenario when media proxying is enabled but media bypass is disabled. > - In the scenario where media bypass is used, the re-INVITE actually appears to contain the correct details, and we are receiving the correct responses from our IP to IP gateway, but FreeSWITCH is not handling the media streams properly. > > Example of SDP descriptor coming from IP phone (with SDP descriptor modified by SIP ALG): > v=0 > o=- 123576 123576 IN IP4 192.168.1.4 > s=- > c=IN IP4 173.57.44.212 > t=0 0 > m=audio 16406 RTP/AVP 18 0 8 2 9 104 101 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:104 L16/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > > Notice that the c= line has the correct public IP address and the m= line containing the correct port. > > Example of incorrect SDP descriptor being sent by FreeSWITCH in re-INVITES: > v=0 > o=- 121397 121398 IN IP4 192.168.1.4 > s=- > c=IN IP4 192.168.1.4 > t=0 0 > m=audio 16404 RTP/AVP 18 0 8 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendonly > a=ptime:20 > > Note that the c= line contains the wrong IP address, but the m= line contains the correct RTP port. > > Example of wrong re-INVITE message being sent to the number that the call was being transferred to: > INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj > Max-Forwards: 69 > From: "John Platts" ;tag=c61Drt38KF72m > To: ;tag=2B1339E0-1A2C > Call-ID: 1c095553-5741-122d-33a8-00185167f91d > CSeq: 123615824 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 183 > X-FS-Support: update_display > Remote-Party-ID: "John Platts" ;party=calling;screen=yes;privacy=off > > v=0 > o=- 123576 123577 IN IP4 192.168.1.4 > s=- > c=IN IP4 168.75.202.212 > t=0 0 > m=audio 30186 RTP/AVP 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > > Here is the correct re-INVITE for the call that was unsuccessfully transferred (after the transfer was completed): > INVITE sip:19729555871 at 168.75.202.246:5060 SIP/2.0 > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKgaDHFKZrc06vD > Max-Forwards: 16 > From: ;tag=BX8mpZj5p6ggS > To: ;tag=2B12D184-BEC > Call-ID: 15A1F95-DBD611DE-8C95D9DF-3419A306 at 168.75.202.246 > CSeq: 123615820 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 222 > X-FS-Support: update_display > > v=0 > o=- 121397 121399 IN IP4 192.168.1.4 > s=- > c=IN IP4 168.75.202.212 > t=0 0 > m=audio 26106 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > _________________________________________________________________ > Windows 7: I wanted simpler, now it's simpler. I'm a rock star. > http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/ From yehavi.bourvine at gmail.com Sun Nov 29 07:47:39 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 29 Nov 2009 17:47:39 +0200 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch Message-ID: Hello, I am trying to set a Polycom 501 phone to do conferencing via the conference room on Freeswitch rather than on the phone (as on the phone it is limited to 3 participants only). Anyone had success with it? I have on the Freeswitch an extension named Conf.* which activates the conference application (it works with other brands). On the Polycom I tried to define voIpProt.SIP.*conference*.address=sip:Conf0000 at freeswitch-server. The phone continues to create the conference locally and add the above Conf0000 to it, without REFERing the parties to it. The first phone which called is left on hold... Anyone managed to make this feature work? We use firmware 3.1.3. Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091129/7a593bce/attachment.html From abeka at greatiam.com Sun Nov 29 09:38:57 2009 From: abeka at greatiam.com (Otis) Date: Sun, 29 Nov 2009 17:38:57 +0000 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 41, Issue 219 In-Reply-To: References: Message-ID: <4B12B1B1.8060807@greatiam.com> Hello Mark Thank you so much. I will put the advise to work. Regards freeswitch-users-request at lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > ------------------------------------------------------------------------ > > Today's Topics: > > 1. Re: GUI for Freeswitch -- wikiPBX (Mark Crane) > 2. Re: Freeswitch admin GUI (Mark Crane) > 3. Call transfer fails in proxy media and bypass media modes in > FreeSWITCH revision 15700 (John Platts) > > > ------------------------------------------------------------------------ > > Subject: > Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > From: > Mark Crane > Date: > Sat, 28 Nov 2009 13:42:35 -0800 (PST) > To: > freeswitch-users at lists.freeswitch.org > > To: > freeswitch-users at lists.freeswitch.org > > > During the install of FusionPBX if you try to connect to MySQL > connection and use 'localhost' it will attempt to use a Unix Socket > then throws an error. > > Instead use 127.0.0.1 then it will actually use TCP connection rather > than the UnixSocket connection. > > This is not a bug in FusionPBX it seems to be just how PHP PDO MySQL > handles the connection. > > Hope this helps. For the release version I will add a little wording > suggesting 127.0.0.1 vs localhost for those that have a local MySQL > install. > > Best Regards, > > Mark J Crane > > > > > > --- On *Sat, 11/28/09, ram //* wrote: > > > From: ram > Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > To: freeswitch-users at lists.freeswitch.org > Date: Saturday, November 28, 2009, 1:57 AM > > > > On Fri, Nov 27, 2009 at 2:03 PM, Otis > wrote: > > Yes. I ventured to use that and got some error in connecting > to the > mysql database. Will try with the default sqlite before getting > adventurous again. > > > Hi > > download latest RC5 > > it has install wizard automatically create database ( > sqllite/mysql/pgsql) > > Ram > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > > Subject: > Re: [Freeswitch-users] Freeswitch admin GUI > From: > Mark Crane > Date: > Sat, 28 Nov 2009 13:51:41 -0800 (PST) > To: > freeswitch-users at lists.freeswitch.org > > To: > freeswitch-users at lists.freeswitch.org > > > FusionPBX is very close to a release. FusionPBX is on the last release > candidate 5 before a 1.0 release. Most of the work in the past couple > weeks has been to make the install easier. ISO versions will be > available in the future. > > I have multiple businesses already running live on FusionPBX. > > The project will advance faster the more it is used and the more > feedback that is given. > > Mark J Crane > http://www.fusionpbx.com > > > > --- On *Fri, 11/27/09, Adam Ford //* wrote: > > > From: Adam Ford > Subject: Re: [Freeswitch-users] Freeswitch admin GUI > To: freeswitch-users at lists.freeswitch.org > Date: Friday, November 27, 2009, 6:10 PM > > FusionPBX, FreePBX v3, and wikiPBX are the three that I have found > in the > past. However they all seem to be in the early stages of > development, and > not 100% stable. I can say this for sure about FreePBX and > FusionPBX, but I > have not actually tried wikiPBX. > > -AF > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On > Behalf Of Otis > Sent: Friday, November 27, 2009 11:49 AM > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Freeswitch admin GUI > > Hi > > I am no sure but read up on fusionpbx. I asked the same question and > someone pointed me to that. > check web site > > Regards > > > > Samuel Mukoti wrote: > >
Hi, > > > > Any recommendations for apps that can I use ontop of freeswitch > as a > > GUI manager, to manage extensions, queues, ivr, and dialplans? > > > > Thanks > > > > Sam > > > > > > On 27 Nov,2009, at 5:19 PM, > > freeswitch-users-request at lists.freeswitch.org > wrote: > > > >> Send FreeSWITCH-users mailing list submissions to > >> freeswitch-users at lists.freeswitch.org > > >> > >> To subscribe or unsubscribe via the World Wide Web, visit > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> or, via email, send a message with subject or body 'help' to > >> freeswitch-users-request at lists.freeswitch.org > > >> > >> You can reach the person managing the list at > >> freeswitch-users-owner at lists.freeswitch.org > > >> > >> When replying, please edit your Subject line so it is more specific > >> than "Re: Contents of FreeSWITCH-users digest..." > >> > >> > >> Today's Topics: > >> > >> 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) > >> > >> > >> > ---------------------------------------------------------------------- > >> > >> Message: 1 > >> Date: Fri, 27 Nov 2009 16:19:03 +0100 > >> From: Leon de Rooij > > >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > >> To: freeswitch-users at lists.freeswitch.org > > >> Message-ID: > > > >> Content-Type: text/plain; charset="windows-1252" > >> > >> Are you using the myodbc 3.51.18 version or higher ? > >> > >> I'm using 3.51.19 (ubuntu karmic) and it works properly. I also > had to > >> upgrade from jaunty.. > >> > >> regards, > >> > >> Leon > >> > >> > >> On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: > >> > >>> Thanks. But when I made these entries in /etc/odbc.ini and > rebooted? > >>> > >>> [freeswitch] > >>> Driver = MySQL > >>> SERVER = 127.0.0.1 > >>> PORT = 4040 > >>> DATABASE = mydb > >>> OPTIONS = 67108864 > >>> > >>> ?I still get FS complaining with this. > >>> > >>> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 > >>> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched > >>> statements!#012If you are using mysql, make sure you are using > >>> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS > >>> > >>> FreeSWITCH>version > >>> FreeSWITCH Version 1.0.trunk (15660) > >>> > >>> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 > >>> EST 2009 x86_64 x86_64 x86_64 GNU/Linux > >>> > >>> From /etc/odbcinst.ini > >>> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so > >>> Setup = /usr/lib64/libodbcmyS.so > >>> > >>> Is this a FS issue ? or an issue with mysql odbc? Any insight > >>> would be great. > >>> > >>> -----Original Message----- > >>> From: freeswitch-users-bounces at lists.freeswitch.org > > >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org > > >>> ] On Behalf Of Leon de Rooij > >>> Sent: Friday, November 27, 2009 3:37 AM > >>> To: freeswitch-users at lists.freeswitch.org > > >>> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > >>> > >>> There's a little info here on how to enable it with odbc: > >>> > >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 > >>> > >>> regards, > >>> > >>> Leon > >>> > >>> > >>> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: > >>> > >>> > >>> > >>> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris > > > >>> wrote: > >>> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- > >>> news-3-51-18.html > >>> > >>> MySQL Connector/ODBC now supports batched statements. In order to > >>> enable > >>> cached statement support you must switch enable the batched > >>> statement option (FLAG_MULTI_STATEMENTS, > >>> 67108864, or Allow multiple statements > >>> within a GUI configuration). Be aware that batched > statements > >>> create an increased chance of SQL injection attacks and you > >>> must > >>> ensure that your application protects against this > scenario. > >>> (Bug#7445) > >>> > >>> > >>> so, is this the right patch ? > >>> > >>> http://bugs.mysql.com/file.php?id=6994 > >>> > >>> > >>> T. > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> > >>> http://www.freeswitch.org > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> > >>> http://www.freeswitch.org > >> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > >> > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/ > 9c86b324/attachment.html > >> > >> > >> ------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> End of FreeSWITCH-users Digest, Vol 41, Issue 209 > >> ************************************************* > > > > > >
> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > > Subject: > [Freeswitch-users] Call transfer fails in proxy media and bypass media > modes in FreeSWITCH revision 15700 > From: > John Platts > Date: > Sat, 28 Nov 2009 23:34:24 -0600 > To: > > > To: > > > > I have updated my FreeSWITCH installation to revision 15700. I am experiencing call transfer problems whenever proxy media or bypass media is enabled. When proxy media and bypass media are both disabled, the call transfer does not fail and there are no audio issues. When proxy media mode is enabled, the call stays up after the transfer occurs, but there is no audio flowing on either end of the call. When bypass media mode is enabled, there is no audio flowing on either end of the call, and the call actually gets disconnected. > > I have collected detailed traces using the TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch command. I have attached a ZIP file named freeswitch-rev15700-traces-112809-2210.zip, which includes the following traces: > - freeswitch-rev15700-trace-112809-2210-proxyandbypassoff.txt - A trace with both media proxying and media bypass disabled. The call is being transferred without any problems in this scenario. > - freeswitch-rev15700-trace-112809-2210-proxyonandbypassoff.txt - A trace with media proxying enabled and media bypass disabled. Media proxying is enabled for the call legs in this scenario. The call stays up in this scenario, but there is no audio flowing after the transfer completed. In this scenario, FreeSWITCH does not shutdown cleanly, and there is a segmentation violation when FreeSWITCH is terminated. > - freeswitch-rev15700-trace-112809-2210-proxyandbypasson.txt - A trace with both media proxying and media bypass enabled. Media bypass is enabled for the call legs in this scenario. The call actually gets dropped and there is no audio after the transfer is completed in this scenario. > > I have looked over the SIP traces of the failing scenarios. > > I have caught the following problems in the failing scenarios: > - The o= line in SDP descriptors coming from the IP phone contains the private IP address, but the c= line in the SDP descriptors coming from the IP phone contains the public IP address. I have noticed a problem in re-INVITEs being sent from in proxy media and bypass media modes. The c= line in the re-invites contains the private IP address instead of the public IP address. The c= line was modified by a SIP ALG to contain a public IP address, but FreeSWITCH is actually not handling this correctly when calls are transferred. > - The wrong codec is being negotiated in re-INVITE to the transferred number in the scenario when media proxying is enabled but media bypass is disabled. > - In the scenario where media bypass is used, the re-INVITE actually appears to contain the correct details, and we are receiving the correct responses from our IP to IP gateway, but FreeSWITCH is not handling the media streams properly. > > Example of SDP descriptor coming from IP phone (with SDP descriptor modified by SIP ALG): > v=0 > o=- 123576 123576 IN IP4 192.168.1.4 > s=- > c=IN IP4 173.57.44.212 > t=0 0 > m=audio 16406 RTP/AVP 18 0 8 2 9 104 101 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:104 L16/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > > Notice that the c= line has the correct public IP address and the m= line containing the correct port. > > Example of incorrect SDP descriptor being sent by FreeSWITCH in re-INVITES: > v=0 > o=- 121397 121398 IN IP4 192.168.1.4 > s=- > c=IN IP4 192.168.1.4 > t=0 0 > m=audio 16404 RTP/AVP 18 0 8 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendonly > a=ptime:20 > > Note that the c= line contains the wrong IP address, but the m= line contains the correct RTP port. > > Example of wrong re-INVITE message being sent to the number that the call was being transferred to: > INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj > Max-Forwards: 69 > From: "John Platts" ;tag=c61Drt38KF72m > To: ;tag=2B1339E0-1A2C > Call-ID: 1c095553-5741-122d-33a8-00185167f91d > CSeq: 123615824 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 183 > X-FS-Support: update_display > Remote-Party-ID: "John Platts" ;party=calling;screen=yes;privacy=off > > v=0 > o=- 123576 123577 IN IP4 192.168.1.4 > s=- > c=IN IP4 168.75.202.212 > t=0 0 > m=audio 30186 RTP/AVP 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > > Here is the correct re-INVITE for the call that was unsuccessfully transferred (after the transfer was completed): > INVITE sip:19729555871 at 168.75.202.246:5060 SIP/2.0 > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKgaDHFKZrc06vD > Max-Forwards: 16 > From: ;tag=BX8mpZj5p6ggS > To: ;tag=2B12D184-BEC > Call-ID: 15A1F95-DBD611DE-8C95D9DF-3419A306 at 168.75.202.246 > CSeq: 123615820 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 222 > X-FS-Support: update_display > > v=0 > o=- 121397 121399 IN IP4 192.168.1.4 > s=- > c=IN IP4 168.75.202.212 > t=0 0 > m=audio 26106 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > _________________________________________________________________ > Windows 7: I wanted simpler, now it's simpler. I'm a rock star. > http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ujjval at simplesignal.com Sun Nov 29 11:17:37 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Sun, 29 Nov 2009 11:17:37 -0800 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch In-Reply-To: References: Message-ID: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> Polycom Firmware matrix (Look at the polycom website) does not allow firmware higher than 2.3.2 (I think) to be loaded on the old 501 phones...So first confirm you are on a supported firmware release... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi Bourvine Sent: Sunday, November 29, 2009 8:48 AM To: freeswitch-users Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch Hello, I am trying to set a Polycom 501 phone to do conferencing via the conference room on Freeswitch rather than on the phone (as on the phone it is limited to 3 participants only). Anyone had success with it? I have on the Freeswitch an extension named Conf.* which activates the conference application (it works with other brands). On the Polycom I tried to define voIpProt.SIP.conference.address=sip:Conf0000 at freeswitch-server. The phone continues to create the conference locally and add the above Conf0000 to it, without REFERing the parties to it. The first phone which called is left on hold... Anyone managed to make this feature work? We use firmware 3.1.3. Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091129/cf802f04/attachment.html From errotan at gmail.com Sun Nov 29 10:06:51 2009 From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=) Date: Sun, 29 Nov 2009 19:06:51 +0100 Subject: [Freeswitch-users] CDR records Message-ID: <200911291906.51520.errotan@gmail.com> Hi Guys! I'm using the latest svn (15711) with the default xml config. Only modified cdr_csv.conf.xml the line to Here is what i do: 1. 1000 calls 1001 (1001 answers the call) 2. 1001 do blind transfer to 1002 (using *1) 3. 1001 hangs up 4. 1002 answers the call 5. 1002 and 1000 hangs up 3 cdr records are generated (simplified): from,to,start,duration "1000" "1001" "2009-11-29 15:21:53" "53" "50" "1000" "1002" "2009-11-29 15:21:53" "79" "76" "1000" "1002" "2009-11-29 15:22:46" "26" "23" As you can see the second cdr is incorrect because 1000 doesn't speak with 1002 for 76 second. Is this a normal ? Is it possible to make only 2 record ? Thank you for any answer. From jbarou at sqli.com Mon Nov 30 00:33:13 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Mon, 30 Nov 2009 09:33:13 +0100 Subject: [Freeswitch-users] Transfer Problem In-Reply-To: <191c3a030911270903i341d1f83pa15f67443422cb67@mail.gmail.com> References: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> <191c3a030911270903i341d1f83pa15f67443422cb67@mail.gmail.com> Message-ID: <8048ff7f0911300033u45c7aa5cwca16581ef9a22c2b@mail.gmail.com> My version is FreeSWITCH Version 1.0.trunk (15691M) http://jira.freeswitch.org/browse/FSBUILD-213 Thanks you. 2009/11/27 Anthony Minessale > by latest do you mean SVN trunk? > > Can you issue the command "sofia profile internal siptrace on" before > capturing your trace and post the results > to http://pastebin.freeswitch.org or open a jira > http://jira.freeswitch.org on the issue and attach the log after you > create the issue ticket, don't include it in the mailing list. > > > On Fri, Nov 27, 2009 at 10:47 AM, Jonathan Barou wrote: > >> Hi everybody, >> >> I'm actually using the lastest version of Freeswitch, I have a problem. I >> have a trunk SIP with my PABX. >> >> There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) >> 2. one Alcatel IpTouch 4028 with number 987 >> (on PABX) >> 3. one Siemens Gigaset A580 IP with number >> 8401 (on Freeswitch) >> >> >> *The first test* is to say to the phone 2 to transfer all the call to >> number 8401. So when I dial 987 on the phone 1, all work perfectly, the >> phone 3 is ringing and it's work. I have that in the log : >> >> 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending >> early media >> >> 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP >> [sofia/internal/368 at 10.33.69.246] 10.33.169.92 port 23054 -> 10.33.69.246 >> port 32000 codec: 8 ms: 90 >> >> 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer [soft] >> 720 bytes per 90ms >> >> 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: >> >> v=0 >> >> o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 >> >> s=FreeSWITCH >> >> c=IN IP4 10.33.169.92 >> >> t=0 0 >> >> m=audio 23054 RTP/AVP 8 106 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:106 telephone-event/8000 >> >> a=fmtp:106 0-16 >> >> a=silenceSupp:off - - - - >> >> a=ptime:90 >> >> a=sendrecv >> >> >> 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer >> sofia/internal/368 at 10.33.69.246! >> >> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >> >> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ >> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >> >> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ >> 368 at 10.33.69.246 skipping state [early][183] >> >> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal >> sofia/internal/368 at 10.33.69.246 [BREAK] >> >> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >> Activation Success L16 at 8000hz 1 channel 90ms >> >> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play >> Ringback Tone [%(2000,4000,440.0,480.0)] >> >> 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ >> 368 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >> >> 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port >> confirmed. >> >> 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read >> Buffer at 1440 bytes vs 81 >> >> >> >> *The Second Tes*t is to say to the phone 1 to transfer all the call to >> number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just >> one time and after it hangup. I have that in the log : >> >> >> 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending >> early media >> >> 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP >> [sofia/internal/987 at 10.33.69.246] 10.33.169.92 port 27732 -> 10.33.69.144 >> port 32000 codec: 8 ms: 90 >> >> 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer [soft] >> 720 bytes per 90ms >> >> 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: >> >> v=0 >> >> o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 >> >> s=FreeSWITCH >> >> c=IN IP4 10.33.169.92 >> >> t=0 0 >> >> m=audio 27732 RTP/AVP 8 106 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:106 telephone-event/8000 >> >> a=fmtp:106 0-16 >> >> a=silenceSupp:off - - - - >> >> a=ptime:90 >> >> a=sendrecv >> >> >> 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer >> sofia/internal/987 at 10.33.69.246! >> >> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >> >> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ >> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >> >> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ >> 987 at 10.33.69.246 skipping state [early][183] >> >> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal >> sofia/internal/987 at 10.33.69.246 [BREAK] >> >> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >> Activation Success L16 at 8000hz 1 channel 90ms >> >> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play >> Ringback Tone [%(2000,4000,440.0,480.0)] >> >> 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ >> 987 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >> >> 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ >> 987 at 10.33.69.246 entering state [terminated][487] >> >> 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ >> 987 at 10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] >> >> 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal >> sofia/internal/987 at 10.33.69.246 [KILL] >> >> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/987 at 10.33.69.246 [BREAK] >> >> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 thread >> mismatch skipping state handler. >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore >> original codec. >> >> 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup >> sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_CONSUME_MEDIA] >> [ORIGINATOR_CANCEL] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [KILL] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >> CS_HANGUP >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >> sip:8401 at 10.33.170.231:5060 Overriding SIP cause 487 with 487 from the >> other leg >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ >> sip:8401 at 10.33.170.231:5060 hanging up, cause: ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to >> sofia/internal/sip:8401 at 10.33.170.231:5060 >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard HANGUP, cause: >> ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP going to sleep >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_HANGUP -> >> CS_REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >> CS_REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard REPORTING, cause: >> ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING going to >> sleep >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_REPORTING -> >> CS_DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) Locked, Waiting on external >> entities >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 thread >> mismatch skipping state handler. >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate >> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] >> >> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 48 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) Ended >> >> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >> Channel sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_DESTROY] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >> CS_DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >> sip:8401 at 10.33.170.231:5060 SOFIA DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY going to sleep >> >> 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot create >> outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate >> Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >> >> 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. >> Cause: ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/987 at 10.33.69.246) State EXECUTE going to sleep >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/987 at 10.33.69.246) Running State Change CS_HANGUP >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >> (sofia/internal/987 at 10.33.69.246) State HANGUP >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >> 987 at 10.33.69.246 Overriding SIP cause 487 with 487 from the other leg >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ >> 987 at 10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >> sofia/internal/987 at 10.33.69.246 Standard HANGUP, cause: ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >> (sofia/internal/987 at 10.33.69.246) State HANGUP going to sleep >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/987 at 10.33.69.246) State Change CS_HANGUP -> CS_REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/987 at 10.33.69.246 [BREAK] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/987 at 10.33.69.246) Running State Change CS_REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >> (sofia/internal/987 at 10.33.69.246) State REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >> sofia/internal/987 at 10.33.69.246 Standard REPORTING, cause: >> ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >> (sofia/internal/987 at 10.33.69.246) State REPORTING going to sleep >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >> (sofia/internal/987 at 10.33.69.246) State Change CS_REPORTING -> CS_DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/987 at 10.33.69.246 [BREAK] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 >> (sofia/internal/987 at 10.33.69.246) Locked, Waiting on external entities >> >> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 47 >> (sofia/internal/987 at 10.33.69.246) Ended >> >> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >> Channel sofia/internal/987 at 10.33.69.246 [CS_DESTROY] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >> (sofia/internal/987 at 10.33.69.246) Running State Change CS_DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >> (sofia/internal/987 at 10.33.69.246) State DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >> 987 at 10.33.69.246 SOFIA DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >> sofia/internal/987 at 10.33.69.246 Standard DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >> (sofia/internal/987 at 10.33.69.246) State DESTROY going to sleep >> >> Finally when I tried to call the phone 3 with the phone 1 it's working, >> and not when I want to call the phone 3 with the phone 2, like just before, >> it's ringing just one time and hangup. >> >> >> Thanks you. >> >> >> Best Regards >> >> -- >> John >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jonathan BAROU Groupe SQLI - CRCI 0472405368 jbarou at sqli.com 1, place Verrazzano 69258 LYON CEDEX 09 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/79f5ecb6/attachment-0001.html From lakindia89 at gmail.com Mon Nov 30 02:03:59 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 30 Nov 2009 15:33:59 +0530 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <191c3a030911271119k3f38a343k8351b121275580b9@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> <191c3a030911271119k3f38a343k8351b121275580b9@mail.gmail.com> Message-ID: <7d79b3930911300203n5879c24fte50dbcada4aa2309@mail.gmail.com> In the previous reply you told me to use new "OUTBOUND" connection. But in this post you mention "INBOUND" connection. That confusion only made me to ask the question once again. Pardon me if I made any mistake. Making a new inbound connection does the task. Thanks for that. On Sat, Nov 28, 2009 at 12:49 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I told you to make a new separate inbound connection back to the server > from your script, do not use the same one thta was tethered to the call > because its too late to use that one. > > Why do I have to answer you twice? > > > > On Thu, Nov 26, 2009 at 3:27 AM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> Hi, Any help or suggestion regarding my previous post. Especially >> >> >> "I also noted that, if I don't receive any events, especially >> "SERVER_DISCONNECTED", then the connection is in established state, but once >> I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it >> correct??" >> Here is the program by which I confirmed the above! >> >> >> require ESL; >> use IO::Socket::INET; >> >> my $ip = "192.168.1.222"; >> my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', >> Proto => 'tcp', Listen => 2, Reuse => 1 ); >> die "Could not create socket: $!\n" unless $sock; >> my $con; >> my $type = "user/"; >> >> for(;;) { >> # wait for any client to connect, a new client will get connected >> when a new call comes in the dialplan. >> >> my $new_sock = $sock->accept(); >> # Do fork and let the parent to wait for more clients. >> >> my $pid = fork(); >> if ($pid) { >> close($new_sock); >> next; >> } >> # Extract the host of the client. >> >> my $host = $new_sock->sockhost(); >> # file descriptor for the socket. >> >> my $fd = fileno($new_sock); >> print "Host name is $host\n"; >> # Create object for the ESL connection package to access the ESL >> functions. >> >> $con = new ESL::ESLconnection($fd); >> # Gets the info about this channel. >> >> my $info = $con->getInfo(); >> my $uuid = $info->getHeader("unique-id"); >> printf "Connected call %s, from %s to %s\n", $uuid, >> $info->getHeader("caller-caller-id-number"), >> $info->getHeader("caller-destination-number"); >> >> # Answer the channel. >> $con->execute("answer"); >> # Set the event lock to tell the FS to execute the instructions in >> the given order. >> $con->setEventLock("true"); >> # Play a file & Get the personal number from the user. >> >> $con->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); >> $con->execute("hangup"); >> >> while($con->connected()) >> { >> my $e=$con->recvEvent(); >> my $ename=$e->getHeader("Event-Name"); >> print $e->serialize(); >> print "$ename\n"; >> print "Connection exists\n"; >> sleep(1); >> >> } >> print >> "Bye\n------------------------------------------------------------------\n"; >> close($new_sock); >> } >> I've not registered for any events. >> In the above program I'm receiving the SERVER_DISCONNECTED event. >> Output when receiving event: >> Host name is 192.168.1.222 >> Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 >> Event-Name: SERVER_DISCONNECTED >> >> SERVER_DISCONNECTED >> Connection exists >> Bye >> >> When I comment the recvEvent line, I got the following output. >> >> Host name is 192.168.1.222 >> Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 >> Connection exists >> Connection exists >> Connection exists >> Connection exists >> Connection exists >> >> >> >> On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy < >> lakindia89 at gmail.com> wrote: >> >>> I've tried the following program as per the suggestion that you've told. >>> But it seems, no success. Once the connection is closed, I created a new >>> connection and I send originate to originate a new call. But it is not >>> working. >>> >>> require ESL; >>> use IO::Socket::INET; >>> use Data::Dumper; >>> >>> my $ip = "192.168.1.222"; >>> my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => >>> '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); >>> die "Could not create socket: $!\n" unless $sock; >>> >>> my $make_call; >>> my $con; >>> my $type = "user/"; >>> >>> for(;;) { >>> my $new_sock = $sock->accept(); >>> my $pid = fork(); >>> if ($pid) { >>> close($new_sock); >>> next; >>> } >>> my $host = $new_sock->sockhost(); >>> my $fd = fileno($new_sock); >>> $con = new ESL::ESLconnection($fd); >>> my $info = $con->getInfo(); >>> my $uuid = $info->getHeader("unique-id"); >>> printf "Connected call %s, from %s to %s\n", $uuid, >>> $info->getHeader("caller-caller-id-number"), >>> $info->getHeader("caller-destination-number"); >>> >>> $con->filter("Unique-Id", $uuid); >>> $con->events("plain", "all"); >>> $con->execute("answer"); >>> $con->setEventLock("true"); >>> my $number=$con->execute("read","2 4 >>> /usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav >>> accnt_number 5000 #"); >>> while($con->connected()) >>> { >>> my $e=$con->recvEvent(); >>> my $ename=$e->getHeader("Event-Name"); >>> my $app=$e->getHeader("Application"); >>> if($ename eq "CHANNEL_EXECUTE_COMPLETE" and $app eq >>> "read") >>> { >>> my $num=$e->getHeader("variable_accnt_number"); >>> print "$num\n"; >>> $con->execute("hangup"); >>> } >>> } >>> if(!$con->connected()) >>> { >>> print "Connection not exists\n"; >>> $con = new ESL::ESLconnection($fd); >>> $con->api("originate","user/1000 &park()"); >>> print "Hai\n"; >>> } >>> print >>> "Bye\n------------------------------------------------------------------\n"; >>> close($new_sock); >>> } >>> Output: >>> Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097 >>> 1000 >>> Connection not exists >>> Hai >>> Bye >>> ------------------------------------------------------------------ >>> The freeswitch log is in >>> http://pastebin.freeswitch.org/11258 >>> >>> I also noted that, if I don't receive any events, especially >>> "SERVER_DISCONNECTED", then the connection is in established state, but once >>> I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it >>> correct?? >>> >>> >>> >>> >>> >>> On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> or open a new outbound connection at the end of your script so you can >>>> send your originate command. >>>> Since the channel hanging up will close your existing connection since >>>> it's only an outbound single session socket. >>>> >>>> >>>> On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins wrote: >>>> >>>>> >>>>> >>>>> On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < >>>>> lakindia89 at gmail.com> wrote: >>>>> >>>>>> Hi, >>>>>> I'm using perl ESL to control the call in freeswitch. >>>>>> I'm having the following scenario, but not able to get it right. >>>>>> >>>>>> Dialplan: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 1. User A calls to an extention (1000). >>>>>> 2. My ESL program will be running, and it answers the call. >>>>>> 3. Then the program will get a number from the user. >>>>>> 4. It will hangup the call. >>>>>> 5. The program has to call to the number that was given by the user. >>>>>> >>>>>> In the above scenario, I was able to do until the 4th step. After >>>>>> hangup the call, if I say originate it is not working. >>>>>> Any ideas on how to do this in ESL. >>>>>> >>>>>> >>>>> I want to make sure I understand what the script is supposed to be >>>>> doing. The caller will key in a phone number to your script and your script >>>>> will collect those digits. The script will then hangup on the caller and >>>>> originate a completely new call? Perhaps you could use sched_api to schedule >>>>> a new originate command for a few seconds into the future and then hangup? >>>>> -MC >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/5e3a1382/attachment-0001.html From neilp at cs.stanford.edu Mon Nov 30 01:49:43 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Mon, 30 Nov 2009 15:19:43 +0530 Subject: [Freeswitch-users] errors installing wanpipe drivers Message-ID: Hi All, I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: > cd wanpipe-3.5.6.5/ > make openzap ... make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make -C api/libstelephony clean make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make[1]: *** No rule to make target `clean'. Stop. make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make: *** [all_lib] Error 2 The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/debbdc85/attachment.html From devel at thom.fr.eu.org Mon Nov 30 02:41:02 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Mon, 30 Nov 2009 11:41:02 +0100 Subject: [Freeswitch-users] errors installing wanpipe drivers In-Reply-To: References: Message-ID: I did manage to build these drivers, but maybe you're not doing it the right way. Sangoma document state that the drivers should be built by using their ./Setup script that does all that is required. I did use ./Setup install which builds the kernel modules, the wanrouter utilities and install all the required stuff. Then you can go back to freeswitch and build the mod_openzap/libopenzap. Fran?ois On Mon, 30 Nov 2009 15:19:43 +0530, Neil Patel wrote: Hi All, I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: > cd wanpipe-3.5.6.5/ > make openzap ... make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make -C api/libstelephony clean make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make[1]: *** No rule to make target `clean'. Stop. make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make: *** [all_lib] Error 2 The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/20f6c2b1/attachment.html From mike at jerris.com Mon Nov 30 03:23:26 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 30 Nov 2009 06:23:26 -0500 Subject: [Freeswitch-users] errors installing wanpipe drivers In-Reply-To: References: Message-ID: make openzap is the correct way to build when using with openzap/freeswitch. If you are having issues with this you should check with sangoma support as to why that build of the drivers is not supporting it properly and what version you should be using. Mike On Nov 30, 2009, at 5:41 AM, Fran?ois Legal wrote: > I did manage to build these drivers, but maybe you're not doing it the right way. Sangoma document state that the drivers should be built by using their ./Setup script that does all that is required. > > I did use ./Setup install which builds the kernel modules, the wanrouter utilities and install all the required stuff. > > Then you can go back to freeswitch and build the mod_openzap/libopenzap. > > > Fran?ois > > > On Mon, 30 Nov 2009 15:19:43 +0530, Neil Patel wrote: > > Hi All, > > I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: > > > cd wanpipe-3.5.6.5/ > > make openzap > ... > make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' > make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' > make -C api/libstelephony clean > make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' > make[1]: *** No rule to make target `clean'. Stop. > make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' > make: *** [all_lib] Error 2 > > The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. > > Thanks, > Neil > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/f80061e3/attachment.html From michaelt at voxcore.voxtelecom.co.za Mon Nov 30 04:12:18 2009 From: michaelt at voxcore.voxtelecom.co.za (Michael Toop) Date: Mon, 30 Nov 2009 14:12:18 +0200 Subject: [Freeswitch-users] DTMF Digits Lost when Under Load In-Reply-To: <191c3a030911160725k38ebcda8ta8c38c36eb80e627@mail.gmail.com> References: <330316f60911152307w2800f2e1r87c77d6dcd70be65@mail.gmail.com> <191c3a030911160725k38ebcda8ta8c38c36eb80e627@mail.gmail.com> Message-ID: <330316f60911300412k52e5bbd6h4236c696f9a45524@mail.gmail.com> Hi All, Thought I would share my solution to this DTMF problem: it turns out my ISP was capping my bandwidth & dropping packets to keep the connection & 1Mbps, so the experienced DTMF loss was actually packets being discarded. On my way to this discovery I tested Freeswitch & DTMF quite thoroughly & never actually found any problems even at hundreds of concurrent calls. Here is how I tested, who knows this might be useful to someone: - I used SIPp to generate calls & a Python script to log the received DTMF digits - SIPp command line: - sipp -sf dtmfSenario.xml -d 10000 -s 451 -l 96 -mp 5606 -i xxx.xxx.xxx.xxx - dtmfSenario.xml below - Dialplan: - - Python: - import sys from freeswitch import * def get_number(session,invalid,num=20): digits = session.getDigits(num, "", 15000) consoleLog("info","Got '%s' digits from user.\n" % digits) if digits == '': # Invalid call if invalid == 3: consoleLog("info","Three invalid attempts!!\n") session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav") session.hangup() sys.exit(0) else: session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav") get_number(session,invalid + 1) else: consoleLog("info","Got a valid number: %s, proceeding...\n" % digits) return digits def handler(session, args): session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav") numberToDial = get_number(session,2,num=10) consoleLog('info','Got 10 DTMF digits. Writing "1" to file...\n') fo = open('/tmp/dtmfData.csv','a') fo.write('"1"\n') fo.close() # Do some stuff & wait for SIPP to hangup session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav") session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav") return - DTMF senario file: - # cat dtmfSenario.xml ;tag=[call_number] To: sut Call-ID: [call_id] CSeq: 1 INVITE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[local_ip_type] [local_ip] t=0 0 m=audio [auto_media_port] RTP/AVP 18 100 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ]]> ;tag=[call_number] To: sut [peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> ;tag=[call_number] To: sut [peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> Cheers, Michael On Mon, Nov 16, 2009 at 5:25 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > That's a pretty small problem description to be so sure about something. > It would probably be better to capture some evidence of the exact problem > you are having since we are using computers and we need to see the computers > in action doing something specifically incorrect to diagnose any sort of > problem. Take the time to describe the origin and destination of your > calls, the call flow, the hardware in use on both ends of the call, detailed > console logs on debug level, (maybe even uncomment the 2833 debug ifded in > switch_rtp.c) and gather something to go on besides "I seem to be losing > dtmf) maybe a packect capture of the networking interface on both ends of > these calls. > > Also problems should be reported to http://jira.freeswitch.org not this > mailing list. > Save us a step if you report a jira and provide all the info above or we > will just have to ask for it again. > > > On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop < > michaelt at voxcore.voxtelecom.co.za> wrote: > >> Hi All, >> >> I have an issue that when my call volumes on my FS IVR box > 30 calls >> DTMF digits are lost (using RFC2833). It is definitely load related as it >> all works perfectly under 30 calls. >> >> Any pointers or a solution to the problem? >> >> Thanks, >> >> Michael >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/227f9d35/attachment-0001.html From woodydickson at gmail.com Mon Nov 30 06:49:32 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Mon, 30 Nov 2009 22:49:32 +0800 Subject: [Freeswitch-users] park on hook Message-ID: Hi, Is there anyway to detect when a channel is park in a way that is similar to hangup-hook or answer-hook? I would like to detect that inside a custom mod, without using the event mechanism? woody -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/9c6fba42/attachment.html From devel at thom.fr.eu.org Mon Nov 30 07:36:48 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Mon, 30 Nov 2009 16:36:48 +0100 Subject: [Freeswitch-users] =?utf-8?q?CLIP_on_FXS_channels_with_mod=5Fopen?= =?utf-8?q?zap?= Message-ID: <567eba90a27903f327c037bcb6062b1a@thom.fr.eu.org> Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. Fran?ois -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/d258e10c/attachment.html From moises.silva at gmail.com Mon Nov 30 07:39:17 2009 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 30 Nov 2009 10:39:17 -0500 Subject: [Freeswitch-users] errors installing wanpipe drivers In-Reply-To: References: Message-ID: On Mon, Nov 30, 2009 at 4:49 AM, Neil Patel wrote: > Hi All, > > I am currently installing a Sangoma A102 card to work with FS using wanpipe > drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related > modules to compile: > > > cd wanpipe-3.5.6.5/ > > make openzap > ... > make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' > make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' > make -C api/libstelephony clean > make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' > make[1]: *** No rule to make target `clean'. Stop. > make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' > make: *** [all_lib] Error 2 > > The libstelephony directory has no Makefile in it. Why is it missing? Is > there a version of wanpipe drivers that will work? I have been unsuccessful > with 3.4.4 and 3.5.6 in similar fashion. > > Hi Neil, Most likely the creation of the Makefile failed (since you mention you can't see a Makefile). Please be sure to have installed the pre-requisites listed at http://wiki.sangoma.com/Requirements Particularly in this case, libtool, autoconf and automake packages. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/6c28008f/attachment.html From anthony.minessale at gmail.com Mon Nov 30 08:48:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Nov 2009 10:48:26 -0600 Subject: [Freeswitch-users] CLIP on FXS channels with mod_openzap In-Reply-To: <567eba90a27903f327c037bcb6062b1a@thom.fr.eu.org> References: <567eba90a27903f327c037bcb6062b1a@thom.fr.eu.org> Message-ID: <191c3a030911300848m6990a978jff57a4b74dd2192d@mail.gmail.com> can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: > Hello, > > > > I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems > on the FXS ports. > > When I ring on FXS ports, the connected phone does not display > callerid/callerid-name. > > I tried turning the stuff of in openzap.conf.xml () but it did not help. > > > > As a side note, turning this on on the FXO ports drops the callerid > information on incoming calls. > > > > Running freeswitch 1.0.4 on linux 2.6.27. > > > > Fran?ois > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/fb01126e/attachment.html From helmut.kuper at ewetel.de Mon Nov 30 08:52:17 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 30 Nov 2009 17:52:17 +0100 Subject: [Freeswitch-users] Sangoma RTP TAP Message-ID: <4B13F841.1080905@ewetel.de> Hello, has anyone of you tried the RTP TAP function of sangoma`s wanpipe driver? It is described here: http://wiki.sangoma.com/wanpipe-voice-rtp-tap On my side wanrouter log says that RTP TAB is configured and enabled, but I can't detect any udp packets received by the remote server (which is described by RTP_TAP_IP, RTP_TAP_MAC and RTP_TAP_PORT). I've latest driver and double checked the wanpipe.conf config. I tried to send some udp packets from wanpiping server to remote server, where the packets were shown up via tcpdump. So there is no FW problem involved. Each try to do some kind of printf debugging in wanpipe-driver doesn't succeed. Any ideas? From kristian.kielhofner at gmail.com Mon Nov 30 08:54:42 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 30 Nov 2009 11:54:42 -0500 Subject: [Freeswitch-users] Accessing custom SIP headers In-Reply-To: <86b72a770911280947m143f40aah640ff8e56ed08950@mail.gmail.com> References: <86b72a770911280947m143f40aah640ff8e56ed08950@mail.gmail.com> Message-ID: <2d9149cd0911300854j251f1481s6ad9405f0b2effb5@mail.gmail.com> The correct way to pass non-standard headers is X- not X_ . On Sat, Nov 28, 2009 at 12:47 PM, Simon Woodhead wrote: > Hi folks, > I'm hoping someone can help me get at custom headers in the dial-plan. I've > read about X- headers being accessible but need to get at some X_ headers > passed through from a proxy. Reading the info app docs, the X shouldn't > actually matter but no matter which way I try I always seem to get a null > result. > An example header in an INVITE is: > X_ACCOUNTCODE: XXXXXX. > I've tried the following dial-plan structures hoping one might work but none > do: > data="accountcodea=${variable_sip_h_X_ACCOUNTCODE}" /> > /> > > > Any help would be much appreciated. > Thanks, > Simon > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From afritzlists at fritztech.com Mon Nov 30 08:54:52 2009 From: afritzlists at fritztech.com (Andrew Fritz) Date: Mon, 30 Nov 2009 10:54:52 -0600 Subject: [Freeswitch-users] Polycom Phones and Domains Message-ID: <4B13F8DC.9050108@fritztech.com> I'm attempting to configure several varieties of polycom (SoundPoint IP 550, SoundPoint IP 601) phones to connect to a freeswitch instance using a domain other than default (i.e. the ip address). Everything works wonderfully as long as the domain is named exactly the same thing as the server host provided to the phone (whether that is the server's ip address, or a hostname resolved via dns). As long as those match everything is fine. What I'm trying to sort out is, is it possible to convince the phone to use something other than the server's hostname/ip as the second part of the user name (i.e. user at host)? Or should I just resign myself to making the domain name some host name that can resolve via DNS? I've tried including @somedomain in the authid field of the line of the phone, but freeswitch reports that the user someuser at somedomain@serverip couldn't be authenticated. It appears that the phone always appends the server name... Does anyone know of a way that I configure a polycom to connect as user1 at d1 and another to connect as user2 at d2 where d1 and d2 are NOT in DNS or are polycoms just going to always use the server host as the @ part of the username? Andrew From anthony.minessale at gmail.com Mon Nov 30 09:22:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Nov 2009 11:22:36 -0600 Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 In-Reply-To: References: Message-ID: <191c3a030911300922y794a23bawf66e203ffc76af89@mail.gmail.com> I don't quite understand what you are talking about? So you have bypass_media=true and you attempt to make an attended xfer as soon as you complete the transfer according to your trace FS does re-invites to convert the call to be exchanging media with FS. The o= lines you don't like are being set by the anonymous device in your callflow and should not impact anything at all. Are you saying something that used to work suddenly has caused you problems or is this the first time you are trying this because we have tested this scenario many times. Are you getting packet captures also and checking where the media is going after those re-invites? if you are intentionally using an ALG you might try without it because 100% of ALG we have ever seen have been badly broken when working with something like FS. On Sun, Nov 29, 2009 at 6:51 AM, John Platts wrote: > > To clarify the problem, the invite message is incorrect because comfort > noise is being negotiated in the re-invite instead of G.711 or G.729: > INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj > Max-Forwards: 69 > From: "John Platts" > >;tag=c61Drt38KF72m > To: > >;tag=2B1339E0-1A2C > Call-ID: 1c095553-5741-122d-33a8-00185167f91d > CSeq: 123615824 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 183 > X-FS-Support: update_display > Remote-Party-ID: "John Platts" > >;party=calling;screen=yes;privacy=off > > v=0 > o=- 123576 123577 IN IP4 192.168.1.4 > s=- > c=IN IP4 168.75.202.212 > t=0 0 > m=audio 30186 RTP/AVP 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > > How do I get it to negotiate G.711, G.729, or other codec instead of > comfort noise? Our IP phones, our FXS gateways, and our IP to IP gateways > expect G.711, G.729, iLBC (if supported by the endpoints), G.722 (if > supported by the endpoints), or G.726 (if supported by the endpoints) be > negotiated. > > ---------------------------------------- > > From: john_platts at hotmail.com > > To: freeswitch-users at lists.freeswitch.org > > Date: Sat, 28 Nov 2009 23:34:24 -0600 > > Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass > media modes in FreeSWITCH revision 15700 > > > > > > I have updated my FreeSWITCH installation to revision 15700. I am > experiencing call transfer problems whenever proxy media or bypass media is > enabled. When proxy media and bypass media are both disabled, the call > transfer does not fail and there are no audio issues. When proxy media mode > is enabled, the call stays up after the transfer occurs, but there is no > audio flowing on either end of the call. When bypass media mode is enabled, > there is no audio flowing on either end of the call, and the call actually > gets disconnected. > > > > I have collected detailed traces using the TPORT_LOG=1 > /usr/local/freeswitch/bin/freeswitch command. I have attached a ZIP file > named freeswitch-rev15700-traces-112809-2210.zip, which includes the > following traces: > > - freeswitch-rev15700-trace-112809-2210-proxyandbypassoff.txt - A trace > with both media proxying and media bypass disabled. The call is being > transferred without any problems in this scenario. > > - freeswitch-rev15700-trace-112809-2210-proxyonandbypassoff.txt - A trace > with media proxying enabled and media bypass disabled. Media proxying is > enabled for the call legs in this scenario. The call stays up in this > scenario, but there is no audio flowing after the transfer completed. In > this scenario, FreeSWITCH does not shutdown cleanly, and there is a > segmentation violation when FreeSWITCH is terminated. > > - freeswitch-rev15700-trace-112809-2210-proxyandbypasson.txt - A trace > with both media proxying and media bypass enabled. Media bypass is enabled > for the call legs in this scenario. The call actually gets dropped and there > is no audio after the transfer is completed in this scenario. > > > > I have looked over the SIP traces of the failing scenarios. > > > > I have caught the following problems in the failing scenarios: > > - The o= line in SDP descriptors coming from the IP phone contains the > private IP address, but the c= line in the SDP descriptors coming from the > IP phone contains the public IP address. I have noticed a problem in > re-INVITEs being sent from in proxy media and bypass media modes. The c= > line in the re-invites contains the private IP address instead of the public > IP address. The c= line was modified by a SIP ALG to contain a public IP > address, but FreeSWITCH is actually not handling this correctly when calls > are transferred. > > - The wrong codec is being negotiated in re-INVITE to the transferred > number in the scenario when media proxying is enabled but media bypass is > disabled. > > - In the scenario where media bypass is used, the re-INVITE actually > appears to contain the correct details, and we are receiving the correct > responses from our IP to IP gateway, but FreeSWITCH is not handling the > media streams properly. > > > > Example of SDP descriptor coming from IP phone (with SDP descriptor > modified by SIP ALG): > > v=0 > > o=- 123576 123576 IN IP4 192.168.1.4 > > s=- > > c=IN IP4 173.57.44.212 > > t=0 0 > > m=audio 16406 RTP/AVP 18 0 8 2 9 104 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:2 G726-32/8000 > > a=rtpmap:9 G722/8000 > > a=rtpmap:104 L16/16000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > a=sendrecv > > > > Notice that the c= line has the correct public IP address and the m= line > containing the correct port. > > > > Example of incorrect SDP descriptor being sent by FreeSWITCH in > re-INVITES: > > v=0 > > o=- 121397 121398 IN IP4 192.168.1.4 > > s=- > > c=IN IP4 192.168.1.4 > > t=0 0 > > m=audio 16404 RTP/AVP 18 0 8 101 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=sendonly > > a=ptime:20 > > > > Note that the c= line contains the wrong IP address, but the m= line > contains the correct RTP port. > > > > Example of wrong re-INVITE message being sent to the number that the call > was being transferred to: > > INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 > > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj > > Max-Forwards: 69 > > From: "John Platts" ;tag=c61Drt38KF72m > > To: ;tag=2B1339E0-1A2C > > Call-ID: 1c095553-5741-122d-33a8-00185167f91d > > CSeq: 123615824 INVITE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 183 > > X-FS-Support: update_display > > Remote-Party-ID: "John Platts" ;party=calling;screen=yes;privacy=off > > > > v=0 > > o=- 123576 123577 IN IP4 192.168.1.4 > > s=- > > c=IN IP4 168.75.202.212 > > t=0 0 > > m=audio 30186 RTP/AVP 101 13 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:13 CN/8000 > > > > Here is the correct re-INVITE for the call that was unsuccessfully > transferred (after the transfer was completed): > > INVITE sip:19729555871 at 168.75.202.246:5060 SIP/2.0 > > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKgaDHFKZrc06vD > > Max-Forwards: 16 > > From: ;tag=BX8mpZj5p6ggS > > To: ;tag=2B12D184-BEC > > Call-ID: 15A1F95-DBD611DE-8C95D9DF-3419A306 at 168.75.202.246 > > CSeq: 123615820 INVITE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 222 > > X-FS-Support: update_display > > > > v=0 > > o=- 121397 121399 IN IP4 192.168.1.4 > > s=- > > c=IN IP4 168.75.202.212 > > t=0 0 > > m=audio 26106 RTP/AVP 0 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > > > _________________________________________________________________ > > Windows 7: I wanted simpler, now it's simpler. I'm a rock star. > > > http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 > > _________________________________________________________________ > Hotmail: Trusted email with powerful SPAM protection. > http://clk.atdmt.com/GBL/go/177141665/direct/01/ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/127d1dd5/attachment-0001.html From moises.silva at gmail.com Mon Nov 30 10:05:21 2009 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 30 Nov 2009 13:05:21 -0500 Subject: [Freeswitch-users] Sangoma RTP TAP In-Reply-To: <4B13F841.1080905@ewetel.de> References: <4B13F841.1080905@ewetel.de> Message-ID: Hello Helmut, On Mon, Nov 30, 2009 at 11:52 AM, Helmut Kuper wrote: > Each try to do some kind of printf debugging in wanpipe-driver doesn't > succeed. > > Any ideas? > The way the rtp tapping works right now is kinda hackish and pretty much Asterisk/Zaptel-based. We depend on the application (either Asterisk or FreeSWITCH) to enable/disable echo cancellation via zaptel commands. When echo cancellation is enabled we assume a call started and enable the tapping, when echo cancellation is disabled we stop the tapping. This behavior has yet to be implemented for FreeSWITCH. An easy way to do it is just to have the wanpipe card to work in zaptel mode and then add a call to zap_channel_command(tech_pvt->zchan, ZAP_COMMAND_ENABLE_ECHOCANCEL) on call start and ZAP_COMMAND_DISABLE_ECHOCANCEL on call stop in mod_openzap.c. The right way to do it is via new API in libsangoma to start tapping and stop tapping. I will add the new libsangoma API to my todo list, hopefully will be done sometime this month. If you want to test the first quick approach send me an off-line message with ssh connection information to get into your box to do these changes so you can test them. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/fa0ef0de/attachment.html From info at daccii.it Mon Nov 30 11:46:54 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Mon, 30 Nov 2009 20:46:54 +0100 Subject: [Freeswitch-users] Questions on ISDN support for Freeswitch Message-ID: <4B14212E.7060003@daccii.it> Hi to all, shortly i'll make a pbx for a customer that uses a couple of isdn bri lines and, looking for hardware, i've seen that not too much expensive isdn cards that works well are the ones that uses hfc-4/8s controller (specifically i'll use a OpenVox B200P that has 2 ISDN ports and use an HFC-4S controller). I've seen that FreeSwitch doesn't support mISDN but uses openzap (trough ozmod_isdn.so). I got some serious troubles using OpenZAP on analogical lines (bad dtmf recognition on fxs ports (like press 4 and get 44), annoying noise, busy/hangup tone unrecognized [tones, in italy, differs by cadency and not frequency], and more). Using tone_detect i bypassed the tone recognition problem and the noise was however acceptable: the blocking problem was the bad dtmf recognition. In the end (hope god forgive me) i put asterisk as (*1*)(*2*)sip proxy between zaptel and freeswich: i need to fix in config hangup cause detection but it seems to works fine. So my questions are: - Do freeswitch supports mISDN? - If it doesn't support mISDN, tone and dtmf recognition will be done by the isdn card, the kernel module or will be done by openzap? - There are alternative ways to use (*3*) mISDN with freeswitch, apart put asterisk as proxy? Thank for your support! Best Regards, Daniele --- (*1*) At beginning i tried using IAX but freeswitch segfaults when it try to answer the call and when i reload the module (trought reload mod_iax) shutdown routing didn't get called(the used iax library, smartly, start using another port without saying anything), however i need to do more testing: hope to open some tickets on jira in short. (*2*) I've seen that mod_iax config file support a "context" variable, but it isn't used so i wrote a small fix to use it if context isn't specified in the iax request (*3*) I need to use mISDN, rather other things, because i should use octasis soft echo cancellation and them supports only mISDN and Zaptel modules -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/e485b4ff/attachment.vcf From andrew at hijacked.us Mon Nov 30 11:51:53 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 30 Nov 2009 14:51:53 -0500 Subject: [Freeswitch-users] Holiday routing examples Message-ID: <20091130195153.GD8574@hijacked.us> Tony committed my patch for doing 'week of month' conditions in the XML dialplan along with some holiday routing examples to the default dialplan. Now you can detect all the major US holidays in pure dialplan XML without having to do any nasty math or anything (I did it all for you). I've also added a page to the wiki describing how to use it for other dates (like non-US holidays): http://wiki.freeswitch.org/wiki/Holiday_Routing Hope this helps some people. Andrew From pjintheusa at gmail.com Mon Nov 30 14:03:12 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 30 Nov 2009 17:03:12 -0500 Subject: [Freeswitch-users] Holiday routing examples In-Reply-To: <20091130195153.GD8574@hijacked.us> References: <20091130195153.GD8574@hijacked.us> Message-ID: <367751820911301403m118bdd46g6c40d3d3492db5e3@mail.gmail.com> Thanks for this goodness. I am sure to use it so it is appreciated. On Mon, Nov 30, 2009 at 2:51 PM, Andrew Thompson wrote: > Tony committed my patch for doing 'week of month' conditions in the XML > dialplan along with some holiday routing examples to the default > dialplan. Now you can detect all the major US holidays in pure dialplan > XML without having to do any nasty math or anything (I did it all for > you). > > I've also added a page to the wiki describing how to use it for other > dates (like non-US holidays): > > http://wiki.freeswitch.org/wiki/Holiday_Routing > > Hope this helps some people. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/68c78249/attachment.html From yehavi.bourvine at gmail.com Mon Nov 30 21:59:59 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 1 Dec 2009 07:59:59 +0200 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller Message-ID: Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing & OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/e0bc2364/attachment.html From anthony.minessale at gmail.com Mon Nov 30 22:15:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 00:15:54 -0600 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> Message-ID: <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> Just set the variables effective_callee_id_name and effective_callee_id_number in your dp before you answer the call On Dec 1, 2009 12:08 AM, "Yehavi Bourvine" wrote: Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing & OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/5f707482/attachment.html From yehavi.bourvine at gmail.com Mon Nov 30 22:42:09 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 1 Dec 2009 08:42:09 +0200 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> Message-ID: Hello Anthony, I think I did not explain myself correctly: The destination sends the Remote-Party-ID in the Ringing and OK replies, but they are not relayed to the original caller. Thanks! __Yehavi: 2009/12/1 Anthony Minessale > Just set the variables effective_callee_id_name and > effective_callee_id_number in your dp before you answer the call > > On Dec 1, 2009 12:08 AM, "Yehavi Bourvine" > wrote: > > Hello, > > I would like Freeswitch to pass the Remote-Party-ID field of the called > party (sent in the Ringing & OK when answering the call) back to the > originator's phone. How can I do that? > > The drive for this is: Our Freeswitch is connected via a Cisco gateway and > PRI to the university's phone exchange. When we call some university's > extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK > which includes the called party's name. I would like Freeswitch to relay > this to the caller so he/she can see the name of the one who they called. > > Thanks! __Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/6fc00f1b/attachment-0001.html From mrene_lists at avgs.ca Mon Nov 30 22:49:30 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 1 Dec 2009 01:49:30 -0500 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> Message-ID: Are you on SVN trunk? As far as I recall the callee_id_number/name stuff isnt in 1.0.4. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 1-Dec-09, at 1:42 AM, Yehavi Bourvine wrote: > Hello Anthony, > > I think I did not explain myself correctly: The destination sends > the Remote-Party-ID in the Ringing and OK replies, but they are not > relayed to the original caller. > > Thanks! __Yehavi: > > 2009/12/1 Anthony Minessale > Just set the variables effective_callee_id_name and > effective_callee_id_number in your dp before you answer the call > > >> On Dec 1, 2009 12:08 AM, "Yehavi Bourvine" >> wrote: >> >> Hello, >> >> I would like Freeswitch to pass the Remote-Party-ID field of the >> called party (sent in the Ringing & OK when answering the call) >> back to the originator's phone. How can I do that? >> >> The drive for this is: Our Freeswitch is connected via a Cisco >> gateway and PRI to the university's phone exchange. When we call >> some university's extension the Cisco gateway adds Remote-Party-ID >> field to the Ringing and OK which includes the called party's name. >> I would like Freeswitch to relay this to the caller so he/she can >> see the name of the one who they called. >> >> Thanks! __Yehavi: >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/8807a857/attachment.html From yehavi.bourvine at gmail.com Mon Nov 30 23:04:07 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 1 Dec 2009 09:04:07 +0200 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> Message-ID: > Are you on SVN trunk? As far as I recall the callee_id_number/name stuff isnt in 1.0.4. No, because the SVN has problems with Emailing the voicemail... We use 1.0.4 and set sip_callee_id_number/name which works. I would like to not set it and get it from the other side... Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/4e160994/attachment.html From tculjaga at gmail.com Sun Nov 1 00:37:20 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sun, 1 Nov 2009 08:37:20 +0100 Subject: [Freeswitch-users] mod_t38gateway In-Reply-To: References: <65d96fc80910301452x2a831733h92b05861d6c94123@mail.gmail.com> Message-ID: <65d96fc80911010037g64c207dat2face4fca8de041a@mail.gmail.com> i tought so :PP T. On Sun, Nov 1, 2009 at 6:34 AM, Michael Jerris wrote: > This is a non working module, just a shell for development. > > Mike > > On Oct 30, 2009, at 5:52 PM, Tihomir Culjaga wrote: > > > does anybody know how does it work and how to use it in a dialplan? > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/6dc0ef76/attachment-0002.html From tculjaga at gmail.com Sun Nov 1 00:49:29 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sun, 1 Nov 2009 08:49:29 +0100 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179 In-Reply-To: References: <87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com> <87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com> <509453DC-57BF-43D0-B704-FF0F2BB58EC0@jerris.com> <87f2f3b90910312224s6968f278i86e35469c781d60c@mail.gmail.com> Message-ID: <65d96fc80911010049w65d59a34i615462f7d6f229a9@mail.gmail.com> and it works :P On Sun, Nov 1, 2009 at 6:38 AM, Michael Jerris wrote: > see rupa's explanation below. > > > On Nov 1, 2009, at 1:24 AM, Michael Collins wrote: > > How would you do an expression like: if $x < 24 in a condition tag? Just > curious. I would like to make sure that is properly documented. > -MC > > >> >> >> On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker wrote: >> >>> ${cond(${myvar} > 15 ? ERR : OK)} >>> >>> >>> If both sides of the comparison operator are numeric then it does >>> numeric comparison otherwise it does lexical string comparison. >>> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/a83e3912/attachment-0002.html From ivan at myrvold.org Sun Nov 1 02:40:05 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Sun, 1 Nov 2009 11:40:05 +0100 Subject: [Freeswitch-users] SIP provider with extern rtp server In-Reply-To: <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org> References: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com> <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org> <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org> <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org> Message-ID: No one have any idea why this is not working? I have combed through the log, but couldn't find any clue there. Incoming calls from my sip provider is working perfect, but for outgoing calls it looks like Freeswitch is not letting the incoming rtp to the local sip phone. Ivan On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote: > Yes, now I got a more detailed trace. Thank you for helping me with > this. > > A new pastebin at http://pastebin.freeswitch.org/10905 > > Ivan > > Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable: > >> fsctl loglevel debug >> console loglevel debug >> sofia profile internal siptrace on >> sofia profile external siptrace on >> sofia loglevel all 9 >> ^^^^^^^^^^^^^^^^^^^^^ >> >> Then run your call, then do this: >> >> sofia loglevel all 0 >> sofia profile external siptrace off >> sofia profile internal siptrace off >> fsctl loglevel warning >> console loglevel warning >> >> On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold >> wrote: >>> I have already set debug to 9, on both profiles. >>> >>> Ivan >>> >>> >>> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable: >>> >>>> See that 200 OK that keeps coming in over and over and over and >>>> over >>>> again? That's because they never received your ACK. If you can >>>> turn on >>>> sofia loglevel to 9 and then watch where you send the ACK, you will >>>> probably have your answer to why the other system did not receive >>>> it. >>>> If you're still not sure what's going on, post another pastebin >>>> with >>>> sofia loglevel set to 9. >>>> >>>> >>>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold >>>> wrote: >>>>> Oh, what happened to it? >>>>> Anyway, here is a new pb: >>>>> http://pastebin.freeswitch.org/10867 >>>>> Ivan >>>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins: >>>>> >>>>> >>>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold >>>>> wrote: >>>>>> >>>>>> Here is a debug log from a call from an internal phone out to an >>>>>> external (my iPhone with nbr 91316356): >>>>>> http://pastebin.freeswitch.org/108578 >>>>>> >>>>>> Ivan >>>>>> >>>>> Uh... you wanna try that PB number again? >>>>> -MC >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch- >>>>> users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch- >>>>> users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>> -- >>>> Eliot Gable >>>> >>>> "We do not inherit the Earth from our ancestors: we borrow it from >>>> our >>>> children." ~David Brower >>>> >>>> "I decided the words were too conservative for me. We're not >>>> borrowing >>>> from our children, we're stealing from them--and it's not even >>>> considered to be a crime." ~David Brower >>>> >>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >>>> live; not live to eat.) ~Marcus Tullius Cicero >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>>> users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Eliot Gable >> >> "We do not inherit the Earth from our ancestors: we borrow it from >> our >> children." ~David Brower >> >> "I decided the words were too conservative for me. We're not >> borrowing >> from our children, we're stealing from them--and it's not even >> considered to be a crime." ~David Brower >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >> live; not live to eat.) ~Marcus Tullius Cicero >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > From yehavi.bourvine at gmail.com Sun Nov 1 06:24:18 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 1 Nov 2009 16:24:18 +0200 Subject: [Freeswitch-users] Rejecting a call from JavaScript Message-ID: Hello, We would like to handle an incoming call to a busy phone according to user's prefference: Some want waiting call, some want to just reject the call, and others want to send the call to voicemail. We have a small JavaScript which tests the status of the destination and the user's will and tries to act accordingly. Our problem is how to send busy. I tried session.hangup("USER_BUSY") but it always sends "temporary unavailable" which causes the orignator to think that the destination is out of order. What is the correct way to do so? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/c428b1bd/attachment-0002.html From diego.viola at gmail.com Sun Nov 1 06:37:26 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 1 Nov 2009 14:37:26 +0000 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <200910312224.47236.chris@cloudtel.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> Message-ID: <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> Don't put me on the same leval as DelphiWorld please, I was just curious about how this SIP stack compares to sofia. Diego On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: > My favorite part of this 'civilized' discussion on IRC was when DelphiWord and > diegoviola sat around tryin to take the piss outta stkn on this issue for > seemingly no reason. Thanks for making the channel a cool place, guys ;) > > On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: >> Anthony Minessale a ?crit : >> > Meftah, >> > Feel free. >> > >> > thanks >> > >> > P.S. >> > >> > STKN was the guy who made the first mod_pjsip for FS that we abandoned >> > years ago. So you should believe him. >> > Both him and I agreed it was not working out. ?So if you don't believe >> > me, find out for yourself. >> > anthony, why i don't believe ?you? >> >> never say that. >> i believe you and all Freeswitch Staf and thank you and to all >> Freeswitch Staf. >> >> > On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb > > > wrote: >> > >> > ? ? hi Anthony >> > ? ? i agry >> > ? ? i say that because STKN hate all my suggestions. >> > ? ? about pjsip, i will contribute aditional module in the contrib. >> > ? ? thanks Anthony >> > >> > ? ? Anthony Minessale a ?crit : >> >> ? ? Meftah, >> >> >> >> ? ? He is 100% correct. ?Please do not insult my volunteer >> >> ? ? developers. Without help from him you would not have any >> >> ? ? FreeSWITCH right now so please drop this subject we are not using >> >> ? ? pjsip. >> >> >> >> >> >> >> >> ? ? On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb >> >> ? ? > wrote: >> >> >> >> ? ? ? ? hi, >> >> ? ? ? ? Pjsip support ICE, STUN and TURN! >> >> ? ? ? ? to STKN: >> >> ? ? ? ? if you don't pjsip, please stop talking or exit the discution >> >> ? ? ? ? we want to kype Freeswitch Clean and universal >> >> >> >> ? ? ? ? Stefan Knoblich a ?crit : >> >>> ? ? ? ? Michael S Collins wrote: >> >>>> ? ? ? ? I can guarantee that the FS devs are well aware of pj-sip. If >> >>>> it was/ is a viable alternative then it would be considered. The fact >> >>>> that it isn't being used is a pretty good indication that it isn't >> >>>> suitable for FS at this time. >> >>>> >> >>>> ? ? ? ? -MV >> >>>> >> >>>> ? ? ? ? Sent from my iPhone >> >>> >> >>> ? ? ? ? We already mentioned some of the reasons why it did get >> >>> ? ? ? ? dropped 3 years ago (first two points from memory, last two >> >>> ? ? ? ? from old IRC logs): [License incompatible (GPL), but i think >> >>> ? ? ? ? tony tried to negotiate on alternate license terms] Not >> >>> ? ? ? ? possible to have multiple SIP profiles (due to global >> >>> ? ? ? ? variables being used in the lib). A race-condition under >> >>> ? ? ? ? high load, that couldn't be resolved back then (with the >> >>> ? ? ? ? help of the pjsip developers). And the sofia module "just >> >>> ? ? ? ? working" and surviving the scalability tests, so all efforts >> >>> ? ? ? ? were focussed on mod_sofia and pjsip got dropped. stkn >> >>> ? ? ? ? _______________________________________________ >> >>> ? ? ? ? FreeSWITCH-users mailing list >> >>> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >> >>> ? ? ? ? >> >>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user >> >>>s http://www.freeswitch.org >> >>> >> >>> >> >>> >> >>> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >> >>> virus signature database 4539 (20091024) __________ >> >>> >> >>> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >> >>> >> >>> ? ? ? ? http://www.eset.com >> >> >> >> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >> >> ? ? ? ? virus signature database 4539 (20091024) __________ >> >> >> >> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >> >> >> >> ? ? ? ? http://www.eset.com >> >> >> >> ? ? ? ? _______________________________________________ >> >> ? ? ? ? FreeSWITCH-users mailing list >> >> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >> >> ? ? ? ? >> >> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> ? ? -- >> >> ? ? Anthony Minessale II >> >> >> >> ? ? FreeSWITCH http://www.freeswitch.org/ >> >> ? ? ClueCon http://www.cluecon.com/ >> >> ? ? Twitter: http://twitter.com/FreeSWITCH_wire >> >> >> >> ? ? AIM: anthm >> >> ? ? MSN:anthony_minessale at hotmail.com >> >> ? ? >> >> ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> ? ? >> >> ? ? IRC: irc.freenode.net #freeswitch >> >> >> >> ? ? FreeSWITCH Developer Conference >> >> ? ? sip:888 at conference.freeswitch.org >> >> ? ? >> >> ? ? iax:guest at conference.freeswitch.org/888 >> >> ? ? >> >> ? ? googletalk:conf+888 at conference.freeswitch.org >> >> ? ? >> >> ? ? pstn:213-799-1400 >> >> >> >> ? ? _______________________________________________ FreeSWITCH-users >> >> ? ? mailing list FreeSWITCH-users at lists.freeswitch.org >> >> ? ? >> >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org __________ Information from ESET NOD32 >> >> Antivirus, version of virus signature database 4539 (20091024) >> >> __________ The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> > >> > ? ? __________ Information from ESET NOD32 Antivirus, version of virus >> > ? ? signature database 4539 (20091024) __________ >> > >> > ? ? The message was checked by ESET NOD32 Antivirus. >> > >> > ? ? http://www.eset.com >> > >> > ? ? _______________________________________________ >> > ? ? FreeSWITCH-users mailing list >> > ? ? FreeSWITCH-users at lists.freeswitch.org >> > ? ? >> > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> > googletalk:conf+888 at conference.freeswitch.org >> > >> > pstn:213-799-1400 >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > __________ Information from ESET NOD32 Antivirus, version of virus >> > signature database 4539 (20091024) __________ >> > >> > The message was checked by ESET NOD32 Antivirus. >> > >> > http://www.eset.com >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From JCasale at activenetwerx.com Sun Nov 1 06:43:58 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Sun, 1 Nov 2009 14:43:58 +0000 Subject: [Freeswitch-users] multihomed help Message-ID: I am setting up fs in pfsense. Following the multihomed tutorial (I also have a dedicated wan/lan int) if I set directory/default.xml domain=10.0.0.1 (my lan int ip) it breaks everything, but if I set in conf/vars.xml I now get audio working correctly, *9999 etc plays MOH. Is there still something I have done wrong? conf/sip_profiles/internal.xml has the int lan ip's entered, and external.xml does not have any edits. Thanks for any info! jlc From jonas.gauffin at gmail.com Sun Nov 1 07:06:04 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Sun, 1 Nov 2009 16:06:04 +0100 Subject: [Freeswitch-users] Small bug in switch_ivr_record_file (in trunk) Message-ID: switch_ivr_play_say.c, line 486. file = switch_core_session_sprintf(session, "%s%s%s%s", switch_str_nil(tfile), tfile ? "]" : "", prefix, SWITCH_PATH_SEPARATOR, file); There should be five %s, not four. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/0b9018b1/attachment-0002.html From jonas.gauffin at gmail.com Sun Nov 1 07:14:57 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Sun, 1 Nov 2009 16:14:57 +0100 Subject: [Freeswitch-users] Small bug in switch_ivr_record_file (in trunk) In-Reply-To: References: Message-ID: Same bug in switch_ivr_async.c, method switch_ivr_record_session. On Sun, Nov 1, 2009 at 4:06 PM, Jonas Gauffin wrote: > switch_ivr_play_say.c, line 486. > > file = switch_core_session_sprintf(session, "%s%s%s%s", > switch_str_nil(tfile), tfile ? "]" : "", prefix, SWITCH_PATH_SEPARATOR, > file); > > There should be five %s, not four. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/b823c318/attachment-0002.html From gromovd at gmail.com Sun Nov 1 07:23:14 2009 From: gromovd at gmail.com (Dmitry Gromov) Date: Sun, 1 Nov 2009 11:23:14 -0400 Subject: [Freeswitch-users] SIP Proxy with direct media path In-Reply-To: <20091101052119.GA28137@jdc.jasonjgw.net> References: <20091101052119.GA28137@jdc.jasonjgw.net> Message-ID: Hi! On Sun, Nov 1, 2009 at 01:21, Jason White wrote: > > http://wiki.freeswitch.org/ is the best there is. > > It is being improved by the community over time. > > You can also take advantage of the IRC channel, the FreeSWITCH conference > and > of course the mailing list. > > > Thank you very much for the reply... I had some experience with FreeSwitch about a year ago, but did not have more time to check all the features. And there wasn't much documentation available then. I'll give it another try this time. Thanks, Dmitry -- DG NJ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/818192da/attachment-0002.html From gmaruzz at celliax.org Sun Nov 1 08:04:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 1 Nov 2009 17:04:36 +0100 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> Message-ID: <7b197bef0911010804n7bcf1d56n180ab963a13a1e39@mail.gmail.com> On Sun, Nov 1, 2009 at 3:37 PM, Diego Viola wrote: > Don't put me on the same leval as DelphiWorld please, I was just > curious about how this SIP stack compares to sofia. Smile Diego, smile. We're all just jocking! :) -gm > > Diego > > On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: >> My favorite part of this 'civilized' discussion on IRC was when DelphiWord and >> diegoviola sat around tryin to take the piss outta stkn on this issue for >> seemingly no reason. Thanks for making the channel a cool place, guys ;) >> >> On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: >>> Anthony Minessale a ?crit : >>> > Meftah, >>> > Feel free. >>> > >>> > thanks >>> > >>> > P.S. >>> > >>> > STKN was the guy who made the first mod_pjsip for FS that we abandoned >>> > years ago. So you should believe him. >>> > Both him and I agreed it was not working out. ?So if you don't believe >>> > me, find out for yourself. >>> > anthony, why i don't believe ?you? >>> >>> never say that. >>> i believe you and all Freeswitch Staf and thank you and to all >>> Freeswitch Staf. >>> >>> > On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb >> > > wrote: >>> > >>> > ? ? hi Anthony >>> > ? ? i agry >>> > ? ? i say that because STKN hate all my suggestions. >>> > ? ? about pjsip, i will contribute aditional module in the contrib. >>> > ? ? thanks Anthony >>> > >>> > ? ? Anthony Minessale a ?crit : >>> >> ? ? Meftah, >>> >> >>> >> ? ? He is 100% correct. ?Please do not insult my volunteer >>> >> ? ? developers. Without help from him you would not have any >>> >> ? ? FreeSWITCH right now so please drop this subject we are not using >>> >> ? ? pjsip. >>> >> >>> >> >>> >> >>> >> ? ? On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb >>> >> ? ? > wrote: >>> >> >>> >> ? ? ? ? hi, >>> >> ? ? ? ? Pjsip support ICE, STUN and TURN! >>> >> ? ? ? ? to STKN: >>> >> ? ? ? ? if you don't pjsip, please stop talking or exit the discution >>> >> ? ? ? ? we want to kype Freeswitch Clean and universal >>> >> >>> >> ? ? ? ? Stefan Knoblich a ?crit : >>> >>> ? ? ? ? Michael S Collins wrote: >>> >>>> ? ? ? ? I can guarantee that the FS devs are well aware of pj-sip. If >>> >>>> it was/ is a viable alternative then it would be considered. The fact >>> >>>> that it isn't being used is a pretty good indication that it isn't >>> >>>> suitable for FS at this time. >>> >>>> >>> >>>> ? ? ? ? -MV >>> >>>> >>> >>>> ? ? ? ? Sent from my iPhone >>> >>> >>> >>> ? ? ? ? We already mentioned some of the reasons why it did get >>> >>> ? ? ? ? dropped 3 years ago (first two points from memory, last two >>> >>> ? ? ? ? from old IRC logs): [License incompatible (GPL), but i think >>> >>> ? ? ? ? tony tried to negotiate on alternate license terms] Not >>> >>> ? ? ? ? possible to have multiple SIP profiles (due to global >>> >>> ? ? ? ? variables being used in the lib). A race-condition under >>> >>> ? ? ? ? high load, that couldn't be resolved back then (with the >>> >>> ? ? ? ? help of the pjsip developers). And the sofia module "just >>> >>> ? ? ? ? working" and surviving the scalability tests, so all efforts >>> >>> ? ? ? ? were focussed on mod_sofia and pjsip got dropped. stkn >>> >>> ? ? ? ? _______________________________________________ >>> >>> ? ? ? ? FreeSWITCH-users mailing list >>> >>> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >>> >>> ? ? ? ? >>> >>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user >>> >>>s http://www.freeswitch.org >>> >>> >>> >>> >>> >>> >>> >>> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >>> >>> virus signature database 4539 (20091024) __________ >>> >>> >>> >>> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >>> >>> >>> >>> ? ? ? ? http://www.eset.com >>> >> >>> >> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >>> >> ? ? ? ? virus signature database 4539 (20091024) __________ >>> >> >>> >> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >>> >> >>> >> ? ? ? ? http://www.eset.com >>> >> >>> >> ? ? ? ? _______________________________________________ >>> >> ? ? ? ? FreeSWITCH-users mailing list >>> >> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >>> >> ? ? ? ? >>> >> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> >> >>> >> ? ? -- >>> >> ? ? Anthony Minessale II >>> >> >>> >> ? ? FreeSWITCH http://www.freeswitch.org/ >>> >> ? ? ClueCon http://www.cluecon.com/ >>> >> ? ? Twitter: http://twitter.com/FreeSWITCH_wire >>> >> >>> >> ? ? AIM: anthm >>> >> ? ? MSN:anthony_minessale at hotmail.com >>> >> ? ? >>> >> ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >> ? ? >>> >> ? ? IRC: irc.freenode.net #freeswitch >>> >> >>> >> ? ? FreeSWITCH Developer Conference >>> >> ? ? sip:888 at conference.freeswitch.org >>> >> ? ? >>> >> ? ? iax:guest at conference.freeswitch.org/888 >>> >> ? ? >>> >> ? ? googletalk:conf+888 at conference.freeswitch.org >>> >> ? ? >>> >> ? ? pstn:213-799-1400 >>> >> >>> >> ? ? _______________________________________________ FreeSWITCH-users >>> >> ? ? mailing list FreeSWITCH-users at lists.freeswitch.org >>> >> ? ? >>> >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org __________ Information from ESET NOD32 >>> >> Antivirus, version of virus signature database 4539 (20091024) >>> >> __________ The message was checked by ESET NOD32 Antivirus. >>> >> http://www.eset.com >>> > >>> > ? ? __________ Information from ESET NOD32 Antivirus, version of virus >>> > ? ? signature database 4539 (20091024) __________ >>> > >>> > ? ? The message was checked by ESET NOD32 Antivirus. >>> > >>> > ? ? http://www.eset.com >>> > >>> > ? ? _______________________________________________ >>> > ? ? FreeSWITCH-users mailing list >>> > ? ? FreeSWITCH-users at lists.freeswitch.org >>> > ? ? >>> > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > -- >>> > Anthony Minessale II >>> > >>> > FreeSWITCH http://www.freeswitch.org/ >>> > ClueCon http://www.cluecon.com/ >>> > Twitter: http://twitter.com/FreeSWITCH_wire >>> > >>> > AIM: anthm >>> > MSN:anthony_minessale at hotmail.com >>> > >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> > >>> > IRC: irc.freenode.net #freeswitch >>> > >>> > FreeSWITCH Developer Conference >>> > sip:888 at conference.freeswitch.org >>> > >>> > iax:guest at conference.freeswitch.org/888 >>> > >>> > googletalk:conf+888 at conference.freeswitch.org >>> > >>> > pstn:213-799-1400 >>> > ------------------------------------------------------------------------ >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> > >>> > __________ Information from ESET NOD32 Antivirus, version of virus >>> > signature database 4539 (20091024) __________ >>> > >>> > The message was checked by ESET NOD32 Antivirus. >>> > >>> > http://www.eset.com >>> >>> __________ Information from ESET NOD32 Antivirus, version of virus >>> signature database 4539 (20091024) __________ >>> >>> The message was checked by ESET NOD32 Antivirus. >>> >>> http://www.eset.com >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From tayeb.meftah at gmail.com Sun Nov 1 09:46:19 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sun, 01 Nov 2009 17:46:19 +0000 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> Message-ID: <4AEDC96B.5090009@gmail.com> hi diego, what you mean? so my level is nothing? my level is bad? my level is zero? thank to gmaruzz/MikeJ that understand me quickly/easyly Diego Viola a ?crit : > Don't put me on the same leval as DelphiWorld please, I was just > curious about how this SIP stack compares to sofia. > > Diego > > On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: > >> My favorite part of this 'civilized' discussion on IRC was when DelphiWord and >> diegoviola sat around tryin to take the piss outta stkn on this issue for >> seemingly no reason. Thanks for making the channel a cool place, guys ;) >> >> On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: >> >>> Anthony Minessale a ?crit : >>> >>>> Meftah, >>>> Feel free. >>>> >>>> thanks >>>> >>>> P.S. >>>> >>>> STKN was the guy who made the first mod_pjsip for FS that we abandoned >>>> years ago. So you should believe him. >>>> Both him and I agreed it was not working out. So if you don't believe >>>> me, find out for yourself. >>>> anthony, why i don't believe you? >>>> >>> never say that. >>> i believe you and all Freeswitch Staf and thank you and to all >>> Freeswitch Staf. >>> >>> >>>> On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb >>> > wrote: >>>> >>>> hi Anthony >>>> i agry >>>> i say that because STKN hate all my suggestions. >>>> about pjsip, i will contribute aditional module in the contrib. >>>> thanks Anthony >>>> >>>> Anthony Minessale a ?crit : >>>> >>>>> Meftah, >>>>> >>>>> He is 100% correct. Please do not insult my volunteer >>>>> developers. Without help from him you would not have any >>>>> FreeSWITCH right now so please drop this subject we are not using >>>>> pjsip. >>>>> >>>>> >>>>> >>>>> On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb >>>>> > wrote: >>>>> >>>>> hi, >>>>> Pjsip support ICE, STUN and TURN! >>>>> to STKN: >>>>> if you don't pjsip, please stop talking or exit the discution >>>>> we want to kype Freeswitch Clean and universal >>>>> >>>>> Stefan Knoblich a ?crit : >>>>> >>>>>> Michael S Collins wrote: >>>>>> >>>>>>> I can guarantee that the FS devs are well aware of pj-sip. If >>>>>>> it was/ is a viable alternative then it would be considered. The fact >>>>>>> that it isn't being used is a pretty good indication that it isn't >>>>>>> suitable for FS at this time. >>>>>>> >>>>>>> -MV >>>>>>> >>>>>>> Sent from my iPhone >>>>>>> >>>>>> We already mentioned some of the reasons why it did get >>>>>> dropped 3 years ago (first two points from memory, last two >>>>>> from old IRC logs): [License incompatible (GPL), but i think >>>>>> tony tried to negotiate on alternate license terms] Not >>>>>> possible to have multiple SIP profiles (due to global >>>>>> variables being used in the lib). A race-condition under >>>>>> high load, that couldn't be resolved back then (with the >>>>>> help of the pjsip developers). And the sofia module "just >>>>>> working" and surviving the scalability tests, so all efforts >>>>>> were focussed on mod_sofia and pjsip got dropped. stkn >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user >>>>>> s http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> __________ Information from ESET NOD32 Antivirus, version of >>>>>> virus signature database 4539 (20091024) __________ >>>>>> >>>>>> The message was checked by ESET NOD32 Antivirus. >>>>>> >>>>>> http://www.eset.com >>>>>> >>>>> __________ Information from ESET NOD32 Antivirus, version of >>>>> virus signature database 4539 (20091024) __________ >>>>> >>>>> The message was checked by ESET NOD32 Antivirus. >>>>> >>>>> http://www.eset.com >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> >>>>> pstn:213-799-1400 >>>>> >>>>> _______________________________________________ FreeSWITCH-users >>>>> mailing list FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org __________ Information from ESET NOD32 >>>>> Antivirus, version of virus signature database 4539 (20091024) >>>>> __________ The message was checked by ESET NOD32 Antivirus. >>>>> http://www.eset.com >>>>> >>>> __________ Information from ESET NOD32 Antivirus, version of virus >>>> signature database 4539 (20091024) __________ >>>> >>>> The message was checked by ESET NOD32 Antivirus. >>>> >>>> http://www.eset.com >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> >>>> iax:guest at conference.freeswitch.org/888 >>>> >>>> googletalk:conf+888 at conference.freeswitch.org >>>> >>>> pstn:213-799-1400 >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> __________ Information from ESET NOD32 Antivirus, version of virus >>>> signature database 4539 (20091024) __________ >>>> >>>> The message was checked by ESET NOD32 Antivirus. >>>> >>>> http://www.eset.com >>>> >>> __________ Information from ESET NOD32 Antivirus, version of virus >>> signature database 4539 (20091024) __________ >>> >>> The message was checked by ESET NOD32 Antivirus. >>> >>> http://www.eset.com >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/bc8f9c19/attachment-0002.html From diego.viola at gmail.com Sun Nov 1 08:56:23 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 1 Nov 2009 16:56:23 +0000 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <7b197bef0911010804n7bcf1d56n180ab963a13a1e39@mail.gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> <7b197bef0911010804n7bcf1d56n180ab963a13a1e39@mail.gmail.com> Message-ID: <86a32abc0911010856s173a45a4m9e1d2b0fee9f165a@mail.gmail.com> =D On Sun, Nov 1, 2009 at 4:04 PM, Giovanni Maruzzelli wrote: > On Sun, Nov 1, 2009 at 3:37 PM, Diego Viola wrote: >> Don't put me on the same leval as DelphiWorld please, I was just >> curious about how this SIP stack compares to sofia. > > Smile Diego, smile. We're all just jocking! ?:) > > -gm > > >> >> Diego >> >> On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: >>> My favorite part of this 'civilized' discussion on IRC was when DelphiWord and >>> diegoviola sat around tryin to take the piss outta stkn on this issue for >>> seemingly no reason. Thanks for making the channel a cool place, guys ;) >>> >>> On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: >>>> Anthony Minessale a ?crit : >>>> > Meftah, >>>> > Feel free. >>>> > >>>> > thanks >>>> > >>>> > P.S. >>>> > >>>> > STKN was the guy who made the first mod_pjsip for FS that we abandoned >>>> > years ago. So you should believe him. >>>> > Both him and I agreed it was not working out. ?So if you don't believe >>>> > me, find out for yourself. >>>> > anthony, why i don't believe ?you? >>>> >>>> never say that. >>>> i believe you and all Freeswitch Staf and thank you and to all >>>> Freeswitch Staf. >>>> >>>> > On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb >>> > > wrote: >>>> > >>>> > ? ? hi Anthony >>>> > ? ? i agry >>>> > ? ? i say that because STKN hate all my suggestions. >>>> > ? ? about pjsip, i will contribute aditional module in the contrib. >>>> > ? ? thanks Anthony >>>> > >>>> > ? ? Anthony Minessale a ?crit : >>>> >> ? ? Meftah, >>>> >> >>>> >> ? ? He is 100% correct. ?Please do not insult my volunteer >>>> >> ? ? developers. Without help from him you would not have any >>>> >> ? ? FreeSWITCH right now so please drop this subject we are not using >>>> >> ? ? pjsip. >>>> >> >>>> >> >>>> >> >>>> >> ? ? On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb >>>> >> ? ? > wrote: >>>> >> >>>> >> ? ? ? ? hi, >>>> >> ? ? ? ? Pjsip support ICE, STUN and TURN! >>>> >> ? ? ? ? to STKN: >>>> >> ? ? ? ? if you don't pjsip, please stop talking or exit the discution >>>> >> ? ? ? ? we want to kype Freeswitch Clean and universal >>>> >> >>>> >> ? ? ? ? Stefan Knoblich a ?crit : >>>> >>> ? ? ? ? Michael S Collins wrote: >>>> >>>> ? ? ? ? I can guarantee that the FS devs are well aware of pj-sip. If >>>> >>>> it was/ is a viable alternative then it would be considered. The fact >>>> >>>> that it isn't being used is a pretty good indication that it isn't >>>> >>>> suitable for FS at this time. >>>> >>>> >>>> >>>> ? ? ? ? -MV >>>> >>>> >>>> >>>> ? ? ? ? Sent from my iPhone >>>> >>> >>>> >>> ? ? ? ? We already mentioned some of the reasons why it did get >>>> >>> ? ? ? ? dropped 3 years ago (first two points from memory, last two >>>> >>> ? ? ? ? from old IRC logs): [License incompatible (GPL), but i think >>>> >>> ? ? ? ? tony tried to negotiate on alternate license terms] Not >>>> >>> ? ? ? ? possible to have multiple SIP profiles (due to global >>>> >>> ? ? ? ? variables being used in the lib). A race-condition under >>>> >>> ? ? ? ? high load, that couldn't be resolved back then (with the >>>> >>> ? ? ? ? help of the pjsip developers). And the sofia module "just >>>> >>> ? ? ? ? working" and surviving the scalability tests, so all efforts >>>> >>> ? ? ? ? were focussed on mod_sofia and pjsip got dropped. stkn >>>> >>> ? ? ? ? _______________________________________________ >>>> >>> ? ? ? ? FreeSWITCH-users mailing list >>>> >>> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >>>> >>> ? ? ? ? >>>> >>> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user >>>> >>>s http://www.freeswitch.org >>>> >>> >>>> >>> >>>> >>> >>>> >>> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >>>> >>> virus signature database 4539 (20091024) __________ >>>> >>> >>>> >>> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >>>> >>> >>>> >>> ? ? ? ? http://www.eset.com >>>> >> >>>> >> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >>>> >> ? ? ? ? virus signature database 4539 (20091024) __________ >>>> >> >>>> >> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >>>> >> >>>> >> ? ? ? ? http://www.eset.com >>>> >> >>>> >> ? ? ? ? _______________________________________________ >>>> >> ? ? ? ? FreeSWITCH-users mailing list >>>> >> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >>>> >> ? ? ? ? >>>> >> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> >>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> ? ? -- >>>> >> ? ? Anthony Minessale II >>>> >> >>>> >> ? ? FreeSWITCH http://www.freeswitch.org/ >>>> >> ? ? ClueCon http://www.cluecon.com/ >>>> >> ? ? Twitter: http://twitter.com/FreeSWITCH_wire >>>> >> >>>> >> ? ? AIM: anthm >>>> >> ? ? MSN:anthony_minessale at hotmail.com >>>> >> ? ? >>>> >> ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> >> ? ? >>>> >> ? ? IRC: irc.freenode.net #freeswitch >>>> >> >>>> >> ? ? FreeSWITCH Developer Conference >>>> >> ? ? sip:888 at conference.freeswitch.org >>>> >> ? ? >>>> >> ? ? iax:guest at conference.freeswitch.org/888 >>>> >> ? ? >>>> >> ? ? googletalk:conf+888 at conference.freeswitch.org >>>> >> ? ? >>>> >> ? ? pstn:213-799-1400 >>>> >> >>>> >> ? ? _______________________________________________ FreeSWITCH-users >>>> >> ? ? mailing list FreeSWITCH-users at lists.freeswitch.org >>>> >> ? ? >>>> >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> >>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org __________ Information from ESET NOD32 >>>> >> Antivirus, version of virus signature database 4539 (20091024) >>>> >> __________ The message was checked by ESET NOD32 Antivirus. >>>> >> http://www.eset.com >>>> > >>>> > ? ? __________ Information from ESET NOD32 Antivirus, version of virus >>>> > ? ? signature database 4539 (20091024) __________ >>>> > >>>> > ? ? The message was checked by ESET NOD32 Antivirus. >>>> > >>>> > ? ? http://www.eset.com >>>> > >>>> > ? ? _______________________________________________ >>>> > ? ? FreeSWITCH-users mailing list >>>> > ? ? FreeSWITCH-users at lists.freeswitch.org >>>> > ? ? >>>> > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> > >>>> > >>>> > -- >>>> > Anthony Minessale II >>>> > >>>> > FreeSWITCH http://www.freeswitch.org/ >>>> > ClueCon http://www.cluecon.com/ >>>> > Twitter: http://twitter.com/FreeSWITCH_wire >>>> > >>>> > AIM: anthm >>>> > MSN:anthony_minessale at hotmail.com >>>> > >>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> > >>>> > IRC: irc.freenode.net #freeswitch >>>> > >>>> > FreeSWITCH Developer Conference >>>> > sip:888 at conference.freeswitch.org >>>> > >>>> > iax:guest at conference.freeswitch.org/888 >>>> > >>>> > googletalk:conf+888 at conference.freeswitch.org >>>> > >>>> > pstn:213-799-1400 >>>> > ------------------------------------------------------------------------ >>>> > >>>> > _______________________________________________ >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> > >>>> > __________ Information from ESET NOD32 Antivirus, version of virus >>>> > signature database 4539 (20091024) __________ >>>> > >>>> > The message was checked by ESET NOD32 Antivirus. >>>> > >>>> > http://www.eset.com >>>> >>>> __________ Information from ESET NOD32 Antivirus, version of virus >>>> signature database 4539 (20091024) __________ >>>> >>>> The message was checked by ESET NOD32 Antivirus. >>>> >>>> http://www.eset.com >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From diego.viola at gmail.com Sun Nov 1 09:09:16 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 1 Nov 2009 17:09:16 +0000 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <4AEDC96B.5090009@gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> <4AEDC96B.5090009@gmail.com> Message-ID: <86a32abc0911010909lcddf914u1b477b22ee65ed75@mail.gmail.com> Hi Meftah, No, of course is not, and it will never be, I actually quite admire how you are able to do what you do. I just wanted to say that I don't want people to put me into this problem, because I don't have anything to do with it, I was just curious about how pjsip compares to sofia, etc. That's all. Apologies if I didn't expressed myself correctly. Regards, Diego On Sun, Nov 1, 2009 at 5:46 PM, Meftah Tayeb wrote: > hi diego, > what you mean? > so my level is nothing? > my level is bad? > my level is zero? > thank to gmaruzz/MikeJ that understand me quickly/easyly > Diego Viola a ?crit?: > > Don't put me on the same leval as DelphiWorld please, I was just > curious about how this SIP stack compares to sofia. > > Diego > > On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: > > > My favorite part of this 'civilized' discussion on IRC was when DelphiWord > and > diegoviola sat around tryin to take the piss outta stkn on this issue for > seemingly no reason. Thanks for making the channel a cool place, guys ;) > > On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: > > > Anthony Minessale a ?crit : > > > Meftah, > Feel free. > > thanks > > P.S. > > STKN was the guy who made the first mod_pjsip for FS that we abandoned > years ago. So you should believe him. > Both him and I agreed it was not working out. ?So if you don't believe > me, find out for yourself. > anthony, why i don't believe ?you? > > > never say that. > i believe you and all Freeswitch Staf and thank you and to all > Freeswitch Staf. > > > > On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb > wrote: > > ? ? hi Anthony > ? ? i agry > ? ? i say that because STKN hate all my suggestions. > ? ? about pjsip, i will contribute aditional module in the contrib. > ? ? thanks Anthony > > ? ? Anthony Minessale a ?crit : > > > ? ? Meftah, > > ? ? He is 100% correct. ?Please do not insult my volunteer > ? ? developers. Without help from him you would not have any > ? ? FreeSWITCH right now so please drop this subject we are not using > ? ? pjsip. > > > > ? ? On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb > ? ? > wrote: > > ? ? ? ? hi, > ? ? ? ? Pjsip support ICE, STUN and TURN! > ? ? ? ? to STKN: > ? ? ? ? if you don't pjsip, please stop talking or exit the discution > ? ? ? ? we want to kype Freeswitch Clean and universal > > ? ? ? ? Stefan Knoblich a ?crit : > > > ? ? ? ? Michael S Collins wrote: > > > ? ? ? ? I can guarantee that the FS devs are well aware of pj-sip. If > it was/ is a viable alternative then it would be considered. The fact > that it isn't being used is a pretty good indication that it isn't > suitable for FS at this time. > > ? ? ? ? -MV > > ? ? ? ? Sent from my iPhone > > > ? ? ? ? We already mentioned some of the reasons why it did get > ? ? ? ? dropped 3 years ago (first two points from memory, last two > ? ? ? ? from old IRC logs): [License incompatible (GPL), but i think > ? ? ? ? tony tried to negotiate on alternate license terms] Not > ? ? ? ? possible to have multiple SIP profiles (due to global > ? ? ? ? variables being used in the lib). A race-condition under > ? ? ? ? high load, that couldn't be resolved back then (with the > ? ? ? ? help of the pjsip developers). And the sofia module "just > ? ? ? ? working" and surviving the scalability tests, so all efforts > ? ? ? ? were focussed on mod_sofia and pjsip got dropped. stkn > ? ? ? ? _______________________________________________ > ? ? ? ? FreeSWITCH-users mailing list > ? ? ? ? FreeSWITCH-users at lists.freeswitch.org > ? ? ? ? > ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > s http://www.freeswitch.org > > > > ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of > virus signature database 4539 (20091024) __________ > > ? ? ? ? The message was checked by ESET NOD32 Antivirus. > > ? ? ? ? http://www.eset.com > > > ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of > ? ? ? ? virus signature database 4539 (20091024) __________ > > ? ? ? ? The message was checked by ESET NOD32 Antivirus. > > ? ? ? ? http://www.eset.com > > ? ? ? ? _______________________________________________ > ? ? ? ? FreeSWITCH-users mailing list > ? ? ? ? FreeSWITCH-users at lists.freeswitch.org > ? ? ? ? > ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ? ? -- > ? ? Anthony Minessale II > > ? ? FreeSWITCH http://www.freeswitch.org/ > ? ? ClueCon http://www.cluecon.com/ > ? ? Twitter: http://twitter.com/FreeSWITCH_wire > > ? ? AIM: anthm > ? ? MSN:anthony_minessale at hotmail.com > ? ? > ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > ? ? > ? ? IRC: irc.freenode.net #freeswitch > > ? ? FreeSWITCH Developer Conference > ? ? sip:888 at conference.freeswitch.org > ? ? > ? ? iax:guest at conference.freeswitch.org/888 > ? ? > ? ? googletalk:conf+888 at conference.freeswitch.org > ? ? > ? ? pstn:213-799-1400 > > ? ? _______________________________________________ FreeSWITCH-users > ? ? mailing list FreeSWITCH-users at lists.freeswitch.org > ? ? > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org __________ Information from ESET NOD32 > Antivirus, version of virus signature database 4539 (20091024) > __________ The message was checked by ESET NOD32 Antivirus. > http://www.eset.com > > > ? ? __________ Information from ESET NOD32 Antivirus, version of virus > ? ? signature database 4539 (20091024) __________ > > ? ? The message was checked by ESET NOD32 Antivirus. > > ? ? http://www.eset.com > > ? ? _______________________________________________ > ? ? FreeSWITCH-users mailing list > ? ? FreeSWITCH-users at lists.freeswitch.org > ? ? > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > __________ Information from ESET NOD32 Antivirus, version of virus signature > database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature > database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Sun Nov 1 09:25:07 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 1 Nov 2009 17:25:07 +0000 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <86a32abc0911010909lcddf914u1b477b22ee65ed75@mail.gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> <4AEDC96B.5090009@gmail.com> <86a32abc0911010909lcddf914u1b477b22ee65ed75@mail.gmail.com> Message-ID: <86a32abc0911010925s19986a84ga20466aef0379cd4@mail.gmail.com> I never said your level is bad or anything, I just said that I don't want people to involve me into that problem. Diego On Sun, Nov 1, 2009 at 5:09 PM, Diego Viola wrote: > Hi Meftah, > > No, of course is not, and it will never be, I actually quite admire > how you are able to do what you do. > > I just wanted to say that I don't want people to put me into this > problem, because I don't have anything to do with it, I was just > curious about how pjsip compares to sofia, etc. That's all. > > Apologies if I didn't expressed myself correctly. > > Regards, > > Diego > > On Sun, Nov 1, 2009 at 5:46 PM, Meftah Tayeb wrote: >> hi diego, >> what you mean? >> so my level is nothing? >> my level is bad? >> my level is zero? >> thank to gmaruzz/MikeJ that understand me quickly/easyly >> Diego Viola a ?crit?: >> >> Don't put me on the same leval as DelphiWorld please, I was just >> curious about how this SIP stack compares to sofia. >> >> Diego >> >> On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: >> >> >> My favorite part of this 'civilized' discussion on IRC was when DelphiWord >> and >> diegoviola sat around tryin to take the piss outta stkn on this issue for >> seemingly no reason. Thanks for making the channel a cool place, guys ;) >> >> On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: >> >> >> Anthony Minessale a ?crit : >> >> >> Meftah, >> Feel free. >> >> thanks >> >> P.S. >> >> STKN was the guy who made the first mod_pjsip for FS that we abandoned >> years ago. So you should believe him. >> Both him and I agreed it was not working out. ?So if you don't believe >> me, find out for yourself. >> anthony, why i don't believe ?you? >> >> >> never say that. >> i believe you and all Freeswitch Staf and thank you and to all >> Freeswitch Staf. >> >> >> >> On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb > > wrote: >> >> ? ? hi Anthony >> ? ? i agry >> ? ? i say that because STKN hate all my suggestions. >> ? ? about pjsip, i will contribute aditional module in the contrib. >> ? ? thanks Anthony >> >> ? ? Anthony Minessale a ?crit : >> >> >> ? ? Meftah, >> >> ? ? He is 100% correct. ?Please do not insult my volunteer >> ? ? developers. Without help from him you would not have any >> ? ? FreeSWITCH right now so please drop this subject we are not using >> ? ? pjsip. >> >> >> >> ? ? On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb >> ? ? > wrote: >> >> ? ? ? ? hi, >> ? ? ? ? Pjsip support ICE, STUN and TURN! >> ? ? ? ? to STKN: >> ? ? ? ? if you don't pjsip, please stop talking or exit the discution >> ? ? ? ? we want to kype Freeswitch Clean and universal >> >> ? ? ? ? Stefan Knoblich a ?crit : >> >> >> ? ? ? ? Michael S Collins wrote: >> >> >> ? ? ? ? I can guarantee that the FS devs are well aware of pj-sip. If >> it was/ is a viable alternative then it would be considered. The fact >> that it isn't being used is a pretty good indication that it isn't >> suitable for FS at this time. >> >> ? ? ? ? -MV >> >> ? ? ? ? Sent from my iPhone >> >> >> ? ? ? ? We already mentioned some of the reasons why it did get >> ? ? ? ? dropped 3 years ago (first two points from memory, last two >> ? ? ? ? from old IRC logs): [License incompatible (GPL), but i think >> ? ? ? ? tony tried to negotiate on alternate license terms] Not >> ? ? ? ? possible to have multiple SIP profiles (due to global >> ? ? ? ? variables being used in the lib). A race-condition under >> ? ? ? ? high load, that couldn't be resolved back then (with the >> ? ? ? ? help of the pjsip developers). And the sofia module "just >> ? ? ? ? working" and surviving the scalability tests, so all efforts >> ? ? ? ? were focussed on mod_sofia and pjsip got dropped. stkn >> ? ? ? ? _______________________________________________ >> ? ? ? ? FreeSWITCH-users mailing list >> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >> ? ? ? ? >> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user >> s http://www.freeswitch.org >> >> >> >> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >> virus signature database 4539 (20091024) __________ >> >> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >> >> ? ? ? ? http://www.eset.com >> >> >> ? ? ? ? __________ Information from ESET NOD32 Antivirus, version of >> ? ? ? ? virus signature database 4539 (20091024) __________ >> >> ? ? ? ? The message was checked by ESET NOD32 Antivirus. >> >> ? ? ? ? http://www.eset.com >> >> ? ? ? ? _______________________________________________ >> ? ? ? ? FreeSWITCH-users mailing list >> ? ? ? ? FreeSWITCH-users at lists.freeswitch.org >> ? ? ? ? >> ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> ? ? -- >> ? ? Anthony Minessale II >> >> ? ? FreeSWITCH http://www.freeswitch.org/ >> ? ? ClueCon http://www.cluecon.com/ >> ? ? Twitter: http://twitter.com/FreeSWITCH_wire >> >> ? ? AIM: anthm >> ? ? MSN:anthony_minessale at hotmail.com >> ? ? >> ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> ? ? >> ? ? IRC: irc.freenode.net #freeswitch >> >> ? ? FreeSWITCH Developer Conference >> ? ? sip:888 at conference.freeswitch.org >> ? ? >> ? ? iax:guest at conference.freeswitch.org/888 >> ? ? >> ? ? googletalk:conf+888 at conference.freeswitch.org >> ? ? >> ? ? pstn:213-799-1400 >> >> ? ? _______________________________________________ FreeSWITCH-users >> ? ? mailing list FreeSWITCH-users at lists.freeswitch.org >> ? ? >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org __________ Information from ESET NOD32 >> Antivirus, version of virus signature database 4539 (20091024) >> __________ The message was checked by ESET NOD32 Antivirus. >> http://www.eset.com >> >> >> ? ? __________ Information from ESET NOD32 Antivirus, version of virus >> ? ? signature database 4539 (20091024) __________ >> >> ? ? The message was checked by ESET NOD32 Antivirus. >> >> ? ? http://www.eset.com >> >> ? ? _______________________________________________ >> ? ? FreeSWITCH-users mailing list >> ? ? FreeSWITCH-users at lists.freeswitch.org >> ? ? >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus signature >> database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> >> >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus signature >> database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From anthony.minessale at gmail.com Sun Nov 1 10:51:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 12:51:57 -0600 Subject: [Freeswitch-users] Small bug in switch_ivr_record_file (in trunk) In-Reply-To: References: Message-ID: <191c3a030911011051w63f1e9e0ibb03be66ac8d53eb@mail.gmail.com> thank you fixed in r15308 We do prefer jira for this kind of thing so we can track and make accurate change logs. On Sun, Nov 1, 2009 at 9:14 AM, Jonas Gauffin wrote: > Same bug in switch_ivr_async.c, method switch_ivr_record_session. > > > On Sun, Nov 1, 2009 at 4:06 PM, Jonas Gauffin wrote: > >> switch_ivr_play_say.c, line 486. >> >> file = switch_core_session_sprintf(session, "%s%s%s%s", >> switch_str_nil(tfile), tfile ? "]" : "", prefix, SWITCH_PATH_SEPARATOR, >> file); >> >> There should be five %s, not four. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/213a1ae9/attachment-0002.html From anthony.minessale at gmail.com Sun Nov 1 10:54:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 12:54:34 -0600 Subject: [Freeswitch-users] Mod_pjsip In-Reply-To: <86a32abc0911010925s19986a84ga20466aef0379cd4@mail.gmail.com> References: <4AEC9C40.502@gmail.com> <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com> <4AECC8F3.7090208@gmail.com> <200910312224.47236.chris@cloudtel.com> <86a32abc0911010637w7defc89aic32cae1ad8a0de4@mail.gmail.com> <4AEDC96B.5090009@gmail.com> <86a32abc0911010909lcddf914u1b477b22ee65ed75@mail.gmail.com> <86a32abc0911010925s19986a84ga20466aef0379cd4@mail.gmail.com> Message-ID: <191c3a030911011054j131f6710s742c8033c3ac71eb@mail.gmail.com> Meftah and Diego, If you continue to argue on our mailing lists I will be forced to moderate you to stop you from bothering other people. I do not want to see one more reply to this thread from either of you. Please do not reply to apologize, simply stop sending any more email to this topic. On Sun, Nov 1, 2009 at 11:25 AM, Diego Viola wrote: > I never said your level is bad or anything, I just said that I don't > want people to involve me into that problem. > > Diego > > On Sun, Nov 1, 2009 at 5:09 PM, Diego Viola wrote: > > Hi Meftah, > > > > No, of course is not, and it will never be, I actually quite admire > > how you are able to do what you do. > > > > I just wanted to say that I don't want people to put me into this > > problem, because I don't have anything to do with it, I was just > > curious about how pjsip compares to sofia, etc. That's all. > > > > Apologies if I didn't expressed myself correctly. > > > > Regards, > > > > Diego > > > > On Sun, Nov 1, 2009 at 5:46 PM, Meftah Tayeb > wrote: > >> hi diego, > >> what you mean? > >> so my level is nothing? > >> my level is bad? > >> my level is zero? > >> thank to gmaruzz/MikeJ that understand me quickly/easyly > >> Diego Viola a ?crit : > >> > >> Don't put me on the same leval as DelphiWorld please, I was just > >> curious about how this SIP stack compares to sofia. > >> > >> Diego > >> > >> On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote: > >> > >> > >> My favorite part of this 'civilized' discussion on IRC was when > DelphiWord > >> and > >> diegoviola sat around tryin to take the piss outta stkn on this issue > for > >> seemingly no reason. Thanks for making the channel a cool place, guys ;) > >> > >> On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: > >> > >> > >> Anthony Minessale a ?crit : > >> > >> > >> Meftah, > >> Feel free. > >> > >> thanks > >> > >> P.S. > >> > >> STKN was the guy who made the first mod_pjsip for FS that we abandoned > >> years ago. So you should believe him. > >> Both him and I agreed it was not working out. So if you don't believe > >> me, find out for yourself. > >> anthony, why i don't believe you? > >> > >> > >> never say that. > >> i believe you and all Freeswitch Staf and thank you and to all > >> Freeswitch Staf. > >> > >> > >> > >> On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb >> > wrote: > >> > >> hi Anthony > >> i agry > >> i say that because STKN hate all my suggestions. > >> about pjsip, i will contribute aditional module in the contrib. > >> thanks Anthony > >> > >> Anthony Minessale a ?crit : > >> > >> > >> Meftah, > >> > >> He is 100% correct. Please do not insult my volunteer > >> developers. Without help from him you would not have any > >> FreeSWITCH right now so please drop this subject we are not using > >> pjsip. > >> > >> > >> > >> On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb > >> > wrote: > >> > >> hi, > >> Pjsip support ICE, STUN and TURN! > >> to STKN: > >> if you don't pjsip, please stop talking or exit the discution > >> we want to kype Freeswitch Clean and universal > >> > >> Stefan Knoblich a ?crit : > >> > >> > >> Michael S Collins wrote: > >> > >> > >> I can guarantee that the FS devs are well aware of pj-sip. If > >> it was/ is a viable alternative then it would be considered. The fact > >> that it isn't being used is a pretty good indication that it isn't > >> suitable for FS at this time. > >> > >> -MV > >> > >> Sent from my iPhone > >> > >> > >> We already mentioned some of the reasons why it did get > >> dropped 3 years ago (first two points from memory, last two > >> from old IRC logs): [License incompatible (GPL), but i think > >> tony tried to negotiate on alternate license terms] Not > >> possible to have multiple SIP profiles (due to global > >> variables being used in the lib). A race-condition under > >> high load, that couldn't be resolved back then (with the > >> help of the pjsip developers). And the sofia module "just > >> working" and surviving the scalability tests, so all efforts > >> were focussed on mod_sofia and pjsip got dropped. stkn > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > >> s http://www.freeswitch.org > >> > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of > >> virus signature database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of > >> virus signature database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> > > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> > > > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > >> > > > >> iax:guest at conference.freeswitch.org/888 > >> > >> googletalk:conf+888 at conference.freeswitch.org > >> > > > >> pstn:213-799-1400 > >> > >> _______________________________________________ FreeSWITCH-users > >> mailing list FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org __________ Information from ESET NOD32 > >> Antivirus, version of virus signature database 4539 (20091024) > >> __________ The message was checked by ESET NOD32 Antivirus. > >> http://www.eset.com > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of virus > >> signature database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> > > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> > > > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > >> > > > >> iax:guest at conference.freeswitch.org/888 > >> > >> googletalk:conf+888 at conference.freeswitch.org > >> > > > >> pstn:213-799-1400 > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of virus > >> signature database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of virus > >> signature database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of virus > signature > >> database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> > >> > >> > >> > >> __________ Information from ESET NOD32 Antivirus, version of virus > signature > >> database 4539 (20091024) __________ > >> > >> The message was checked by ESET NOD32 Antivirus. > >> > >> http://www.eset.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/88ae4771/attachment-0002.html From anthony.minessale at gmail.com Sun Nov 1 11:27:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 13:27:31 -0600 Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session In-Reply-To: <8ccbff060910311945j5be2fa85j95e7e1b422dcc3a9@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com> <191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com> <8ccbff060910311945j5be2fa85j95e7e1b422dcc3a9@mail.gmail.com> Message-ID: <191c3a030911011127g543d8464g8a7933f9e4860951@mail.gmail.com> #4 makes no sense to me. Are you just trying to create a call that the channel does not participate in? Then for sure you want to use bgapi to cause the originate to happen in the background. Also your gcore report has to be taken while you have the stuck channels to see why they are stuck. I can promise you this problem should be filed under misuse/abuse. On Sat, Oct 31, 2009 at 8:45 PM, Dome Charoenyost wrote: > How to use bgapi in my flow. > 1. user call did > 2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey) > 3. Hangup (by use or timeout) > 4. FS callback to user and bridge to IVR > > I'm not sure bgapi can do after channel hangup. > > Dome C. > > > 2009/11/1 Anthony Minessale : > > Use bgapi originate ... > > > > On Oct 31, 2009 9:09 PM, "Dome Charoenyost" wrote: > > > > Yes. i user api_hangup_hook for do callback. > > > > > > may be need originate_timeout > > > > Dome C. > > > > > > 2009/11/1 Rupa Schomaker : > > > >> fscore_pb has been updated. > > Next time put in the pastebin url. This > >> one was 10911. > > Ok, a ... > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/ad4a0733/attachment-0002.html From anthony.minessale at gmail.com Sun Nov 1 11:31:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 13:31:59 -0600 Subject: [Freeswitch-users] Rejecting a call from JavaScript In-Reply-To: References: Message-ID: <191c3a030911011131h6310f312u555af860889488c8@mail.gmail.com> try session.execute("hangup", "user_busy"); On Sun, Nov 1, 2009 at 8:24 AM, Yehavi Bourvine wrote: > Hello, > > We would like to handle an incoming call to a busy phone according > to user's prefference: Some want waiting call, some want to just reject the > call, and others want to send the call to voicemail. > > We have a small JavaScript which tests the status of the destination and > the user's will and tries to act accordingly. Our problem is how to send > busy. I tried session.hangup("USER_BUSY") but it always sends "temporary > unavailable" which causes the orignator to think that the destination is out > of order. > > What is the correct way to do so? > > Thanks! __Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/8865a324/attachment-0002.html From anthony.minessale at gmail.com Sun Nov 1 11:40:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 13:40:21 -0600 Subject: [Freeswitch-users] CDR CSV variables In-Reply-To: <609781.88322.qm@web37502.mail.mud.yahoo.com> References: <609781.88322.qm@web37502.mail.mud.yahoo.com> Message-ID: <191c3a030911011140q505868c3p5a9c94b5decbf857@mail.gmail.com> you can make up your own variable and set whatever you want in there then add it to the template. On Fri, Oct 30, 2009 at 10:04 AM, DJB wrote: > I wonder if I don't want to have b-leg in cdr csv, is there any variables > that can give me the actual gateway ip address that is actually went out. > > For instance, if I have this in my dialplan: > > the only value that I can think of from cdr csv is to get > remote_ip_last_arg, but it would contains the whole line of both ip > addresses. > > Thank you, > Dorn B. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/f53517af/attachment-0002.html From dome at tel.co.th Sun Nov 1 11:53:01 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 2 Nov 2009 02:53:01 +0700 Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session In-Reply-To: <191c3a030911011127g543d8464g8a7933f9e4860951@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com> <191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com> <8ccbff060910311945j5be2fa85j95e7e1b422dcc3a9@mail.gmail.com> <191c3a030911011127g543d8464g8a7933f9e4860951@mail.gmail.com> Message-ID: <8ccbff060911011153h63bed954qe154b716a2bf8099@mail.gmail.com> 2009/11/2 Anthony Minessale : > #4 makes no sense to me. This solution call "miss callback". some contry cost for toll free number is so high. but call to user cheaper. so when user want to use international call. they call to my DID and my system callback to them. > Are you just trying to create a call that the channel does not participate > in? > Then for sure you want to use bgapi to cause the originate to? happen in the > background. Give me example how bgapi work when user hangup call Dome C. > > Also your gcore report has to be taken while you have the stuck channels to > see why they are stuck. > I can promise you this problem should be filed under misuse/abuse. > > > On Sat, Oct 31, 2009 at 8:45 PM, Dome Charoenyost wrote: >> >> How to use bgapi in my flow. >> 1. user call did >> 2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey) >> 3. Hangup (by use or timeout) >> 4. FS callback to user and bridge to IVR >> >> I'm not sure ?bgapi can do after channel hangup. >> >> Dome C. >> >> >> 2009/11/1 Anthony Minessale : >> > Use bgapi originate ... >> > >> > On Oct 31, 2009 9:09 PM, "Dome Charoenyost" wrote: >> > >> > Yes. i user api_hangup_hook for do callback. >> > >> > >> > may be need originate_timeout >> > >> > Dome C. >> > >> > >> > 2009/11/1 Rupa Schomaker : >> > >> >> fscore_pb has been updated. > > Next time put in the pastebin url. >> >> ?This >> >> one was 10911. > > Ok, a ... >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Sun Nov 1 12:04:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 14:04:41 -0600 Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session In-Reply-To: <8ccbff060911011153h63bed954qe154b716a2bf8099@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com> <191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com> <8ccbff060910311945j5be2fa85j95e7e1b422dcc3a9@mail.gmail.com> <191c3a030911011127g543d8464g8a7933f9e4860951@mail.gmail.com> <8ccbff060911011153h63bed954qe154b716a2bf8099@mail.gmail.com> Message-ID: <191c3a030911011204r6a74b4e2u108c2943d14d0ee4@mail.gmail.com> if you set the variable to: "originate foo bar baz" "bgapi originate foo bar baz" bgapi is itself an api command that takes the argument string and runs it in a separate thread and returns instantly. This will stop it from hanging at that point. On Sun, Nov 1, 2009 at 1:53 PM, Dome Charoenyost wrote: > 2009/11/2 Anthony Minessale : > > #4 makes no sense to me. > This solution call "miss callback". some contry cost for toll free > number is so high. but call to user cheaper. so when user want to > use international call. they call to my DID and my system callback to them. > > > Are you just trying to create a call that the channel does not > participate > > in? > > Then for sure you want to use bgapi to cause the originate to happen in > the > > background. > Give me example how bgapi work when user hangup call > > > > Dome C. > > > > > Also your gcore report has to be taken while you have the stuck channels > to > > see why they are stuck. > > I can promise you this problem should be filed under misuse/abuse. > > > > > > On Sat, Oct 31, 2009 at 8:45 PM, Dome Charoenyost > wrote: > >> > >> How to use bgapi in my flow. > >> 1. user call did > >> 2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey) > >> 3. Hangup (by use or timeout) > >> 4. FS callback to user and bridge to IVR > >> > >> I'm not sure bgapi can do after channel hangup. > >> > >> Dome C. > >> > >> > >> 2009/11/1 Anthony Minessale : > >> > Use bgapi originate ... > >> > > >> > On Oct 31, 2009 9:09 PM, "Dome Charoenyost" wrote: > >> > > >> > Yes. i user api_hangup_hook for do callback. > >> > > >> > > >> > may be need originate_timeout > >> > > >> > Dome C. > >> > > >> > > >> > 2009/11/1 Rupa Schomaker : > >> > > >> >> fscore_pb has been updated. > > Next time put in the pastebin url. > >> >> This > >> >> one was 10911. > > Ok, a ... > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/38463b48/attachment-0002.html From anthony.minessale at gmail.com Sun Nov 1 12:20:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 14:20:55 -0600 Subject: [Freeswitch-users] SIP provider with extern rtp server In-Reply-To: References: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com> <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org> <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org> <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org> Message-ID: <191c3a030911011220m22d7b515kda412e1fd9408f59@mail.gmail.com> Session-Expires: -1;refresher=uas nta: 200 OK has fatal syntax errors This is a know-bug in asterisk. see: https://issues.asterisk.org/view.php?id=15621 On Sun, Nov 1, 2009 at 4:40 AM, Ivan C Myrvold wrote: > No one have any idea why this is not working? I have combed through > the log, but couldn't find any clue there. > Incoming calls from my sip provider is working perfect, but for > outgoing calls it looks like Freeswitch is not letting the incoming > rtp to the local sip phone. > > Ivan > > On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote: > > > Yes, now I got a more detailed trace. Thank you for helping me with > > this. > > > > A new pastebin at http://pastebin.freeswitch.org/10905 > > > > Ivan > > > > Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable: > > > >> fsctl loglevel debug > >> console loglevel debug > >> sofia profile internal siptrace on > >> sofia profile external siptrace on > >> sofia loglevel all 9 > >> ^^^^^^^^^^^^^^^^^^^^^ > >> > >> Then run your call, then do this: > >> > >> sofia loglevel all 0 > >> sofia profile external siptrace off > >> sofia profile internal siptrace off > >> fsctl loglevel warning > >> console loglevel warning > >> > >> On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold > >> wrote: > >>> I have already set debug to 9, on both profiles. > >>> > >>> Ivan > >>> > >>> > >>> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable: > >>> > >>>> See that 200 OK that keeps coming in over and over and over and > >>>> over > >>>> again? That's because they never received your ACK. If you can > >>>> turn on > >>>> sofia loglevel to 9 and then watch where you send the ACK, you will > >>>> probably have your answer to why the other system did not receive > >>>> it. > >>>> If you're still not sure what's going on, post another pastebin > >>>> with > >>>> sofia loglevel set to 9. > >>>> > >>>> > >>>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold > >>>> wrote: > >>>>> Oh, what happened to it? > >>>>> Anyway, here is a new pb: > >>>>> http://pastebin.freeswitch.org/10867 > >>>>> Ivan > >>>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins: > >>>>> > >>>>> > >>>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold > >>>>> wrote: > >>>>>> > >>>>>> Here is a debug log from a call from an internal phone out to an > >>>>>> external (my iPhone with nbr 91316356): > >>>>>> http://pastebin.freeswitch.org/108578 > >>>>>> > >>>>>> Ivan > >>>>>> > >>>>> Uh... you wanna try that PB number again? > >>>>> -MC > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > >>>>> freeswitch- > >>>>> users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > >>>>> freeswitch- > >>>>> users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>> -- > >>>> Eliot Gable > >>>> > >>>> "We do not inherit the Earth from our ancestors: we borrow it from > >>>> our > >>>> children." ~David Brower > >>>> > >>>> "I decided the words were too conservative for me. We're not > >>>> borrowing > >>>> from our children, we're stealing from them--and it's not even > >>>> considered to be a crime." ~David Brower > >>>> > >>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > >>>> live; not live to eat.) ~Marcus Tullius Cicero > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>>> users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Eliot Gable > >> > >> "We do not inherit the Earth from our ancestors: we borrow it from > >> our > >> children." ~David Brower > >> > >> "I decided the words were too conservative for me. We're not > >> borrowing > >> from our children, we're stealing from them--and it's not even > >> considered to be a crime." ~David Brower > >> > >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > >> live; not live to eat.) ~Marcus Tullius Cicero > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/4b8be022/attachment-0002.html From dujinfang at gmail.com Sun Nov 1 17:37:26 2009 From: dujinfang at gmail.com (Seven Du) Date: Mon, 2 Nov 2009 09:37:26 +0800 Subject: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session In-Reply-To: <23f91030910311924of53ae2ega246cf367f4a98a7@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms> <191c3a030910311426of22a4fdyb6be7cabfb992bd6@mail.gmail.com> <23f91030910311924of53ae2ega246cf367f4a98a7@mail.gmail.com> Message-ID: <23f91030911011737l70a39195xb7a5e69df51f4865@mail.gmail.com> Just suspicious would be possible that happened on sqlite stage? I manually deleted the channels from sqlite and nothing bad happend. just FYI. ---------- Forwarded message ---------- From: Seven Du Date: Sun, 1 Nov 2009 10:24:32 +0800 Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session To: Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound and outbound. I must be very careful if I upgrade to trunk and turn on rwlock debug because it's on production and the the problem not happening that much so would be hard to trace. But I will find time to test and report back. FYI, I also noticed that in some zombile channels, couple of INVITEs sent but never got a response. However, it sent out an ACK at last. From lei.tlfly at gmail.com Sun Nov 1 18:26:59 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Mon, 2 Nov 2009 10:26:59 +0800 Subject: [Freeswitch-users] Get error "415 Unsupported Media Type" when receiving call from softswitch Message-ID: <50c41b4e0911011826s3431a0cex79d4ba7eee79c872@mail.gmail.com> Hi all, I get a "415 Unsupported Media Type" when FS receiving call from a softswitch. I captured some packets, It seems that the softswitch use SIP-I protocol, does FS can handle SIP-I message? ===here is the invite messagefrom softswitch INVITE sip:xxxxx at xxxx:5060;user=phone SIP/2.0 Contact: MIME-version: 1.0 Content-Type: multipart/mixed;boundary=Alcatel-boundary To: From: xxxx;tag=73D332463135364195291201 P-Asserted-Identity: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE Supported: 100rel,timer,replaces,diversion Expires: 155 Session-Expires: 1800 Min-SE: 90 Call-ID: 01FD034872814000000230A1 at sip-3 Max-Forwards: 70 CSeq: 1 INVITE Timestamp: 10645 Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E Content-Length: 542 --Alcatel-boundary Content-Type: application/sdp v=0 o=- 5 8 IN IP4 xxxxxx s=SDP Data c=IN IP4 xxxxx t=0 0 m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97 a=rtpmap:110 speex/8000/1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 TELEPHONE-EVENT/8000 a=fmtp:110 mode=3 a=rtpmap:111 speex/8000/1 a=rtpmap:3 GSM/8000 a=ptime:30 a=fmtp:111 mode=2 a=rtpmap:112 speex/8000/1 --Alcatel-boundary Content-Type: application/ISUP;version=N/A .. .... ...0.D... ...V..0......1...n..p... --Alcatel-boundary-- =====response message from FS SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E From: xxxxx ;tag=73D332463135364195291201 To: ;tag=m6XS44BKF4pSS Call-ID: 01FD034872814000000230A1 at sip-3 CSeq: 1 INVITE Timestamp: 10645 0.000000 User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/d8ab7972/attachment-0002.html From lei.tlfly at gmail.com Sun Nov 1 20:09:25 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Mon, 2 Nov 2009 12:09:25 +0800 Subject: [Freeswitch-users] Get error "415 Unsupported Media Type" when receiving call from softswitch In-Reply-To: <50c41b4e0911011826s3431a0cex79d4ba7eee79c872@mail.gmail.com> References: <50c41b4e0911011826s3431a0cex79d4ba7eee79c872@mail.gmail.com> Message-ID: <50c41b4e0911012009y2b2bbfa3qa4713854843c968f@mail.gmail.com> FYI, Here is the log when I set sofia loglevel all 9 ============== tport_wakeup_pri(00DFE3E8): events IN tport_recv_event(00DFE3E8) tport(00DFE3E8) msg 01B2E0C0 from (udp/MyIP:5060) has 1315 bytes, veclen = 1 tport(00DFE3E8): msg 01B2E0C0 (1315 bytes) from udp/SSIP:5060/sip next=000 00000 nta: received INVITE sip:AAA at MyIP:5060;user=phone SIP/2.0 (CSeq 1) nta: canonizing sip:AAA at MyIP:5060 with contact nta: INVITE (1) going to a default leg nta: timer shortened to 200 ms tport_tsend(00DFE3E8) tpn = UDP/SSIP:5061 tport_resolve addrinfo = SSIP:5061 tport(00DFE3E8): not found by name UDP/SSIP:5061 tport_vsend(00DFE3E8): 652 bytes of 652 to udp/SSIP:5061 tport_vsend returned 652 nta: sent 415 Unsupported Media Type for INVITE (1) tport_wakeup_pri(00DFE3E8): events IN tport_recv_event(00DFE3E8) tport(00DFE3E8) msg 0192C510 from (udp/MyIP:5060) has 388 bytes, veclen = 1 tport(00DFE3E8): msg 0192C510 (388 bytes) from udp/SSIP:5060/sip next=0000 0000 nta: received ACK sip:AAA at MyIP:5060;user=phone SIP/2.0 (CSeq 1) nta: ACK (1) is going to INVITE (1) nta: timer set next to 937 ms nta: timer J fired, terminate 200 response incoming_reclaim_all(00000000, 00000000, 02E2FEB8) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 3859 ms nta: timer I fired, terminate 415 response incoming_reclaim_all(00000000, 00000000, 02E2FEB8) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set 2009/11/2 Lei Tang > Hi all, I get a "415 Unsupported Media Type" when FS receiving call from a > softswitch. I captured some packets, It seems that the softswitch use SIP-I > protocol, does FS can handle SIP-I message? > > ===here is the invite messagefrom softswitch > INVITE sip:xxxxx at xxxx:5060;user=phone SIP/2.0 > Contact: > MIME-version: 1.0 > Content-Type: multipart/mixed;boundary=Alcatel-boundary > To: > From: xxxx;tag=73D332463135364195291201 > P-Asserted-Identity: > Allow: > INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE > Supported: 100rel,timer,replaces,diversion > Expires: 155 > Session-Expires: 1800 > Min-SE: 90 > Call-ID: 01FD034872814000000230A1 at sip-3 > Max-Forwards: 70 > CSeq: 1 INVITE > Timestamp: 10645 > Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E > Content-Length: 542 > > --Alcatel-boundary > Content-Type: application/sdp > > v=0 > o=- 5 8 IN IP4 xxxxxx > s=SDP Data > c=IN IP4 xxxxx > t=0 0 > m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97 > a=rtpmap:110 speex/8000/1 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:96 TELEPHONE-EVENT/8000 > a=fmtp:110 mode=3 > a=rtpmap:111 speex/8000/1 > a=rtpmap:3 GSM/8000 > a=ptime:30 > a=fmtp:111 mode=2 > a=rtpmap:112 speex/8000/1 > > --Alcatel-boundary > Content-Type: application/ISUP;version=N/A > > .. .... > ...0.D... > ...V..0......1...n..p... > --Alcatel-boundary-- > > =====response message from FS > SIP/2.0 415 Unsupported Media Type > Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E > From: xxxxx ;tag=73D332463135364195291201 > To: ;tag=m6XS44BKF4pSS > Call-ID: 01FD034872814000000230A1 at sip-3 > CSeq: 1 INVITE > Timestamp: 10645 0.000000 > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 > Accept: application/sdp > Accept-Encoding: > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Length: 0 > -- > Lei.Tang > lei.tlfly at gmail.com > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/daaead31/attachment-0002.html From yehavi.bourvine at gmail.com Sun Nov 1 20:15:32 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 2 Nov 2009 06:15:32 +0200 Subject: [Freeswitch-users] Rejecting a call from JavaScript In-Reply-To: <191c3a030911011131h6310f312u555af860889488c8@mail.gmail.com> References: <191c3a030911011131h6310f312u555af860889488c8@mail.gmail.com> Message-ID: Thanks! It works! __Yehavi: 2009/11/1 Anthony Minessale > try session.execute("hangup", "user_busy"); > > > On Sun, Nov 1, 2009 at 8:24 AM, Yehavi Bourvine < > yehavi.bourvine at gmail.com> wrote: > >> Hello, >> >> We would like to handle an incoming call to a busy phone according >> to user's prefference: Some want waiting call, some want to just reject the >> call, and others want to send the call to voicemail. >> >> We have a small JavaScript which tests the status of the destination and >> the user's will and tries to act accordingly. Our problem is how to send >> busy. I tried session.hangup("USER_BUSY") but it always sends "temporary >> unavailable" which causes the orignator to think that the destination is out >> of order. >> >> What is the correct way to do so? >> >> Thanks! __Yehavi: >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/81bb8333/attachment-0002.html From anthony.minessale at gmail.com Sun Nov 1 20:56:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Nov 2009 22:56:55 -0600 Subject: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session In-Reply-To: <191c3a030911012056j2991b3c6y3cda1b705b5f9df9@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms> <191c3a030910311426of22a4fdyb6be7cabfb992bd6@mail.gmail.com> <23f91030910311924of53ae2ega246cf367f4a98a7@mail.gmail.com> <23f91030911011737l70a39195xb7a5e69df51f4865@mail.gmail.com> <191c3a030911012056j2991b3c6y3cda1b705b5f9df9@mail.gmail.com> Message-ID: <191c3a030911012056y66441b8aya672a16fabe89f33@mail.gmail.com> We already concluded its your unacceptabe use of originate in hangup hook right? On Nov 1, 2009 7:45 PM, "Seven Du" wrote: Just suspicious would be possible that happened on sqlite stage? I manually deleted the channels from sqlite and nothing bad happend. just FYI. ---------- Forwarded message ---------- From: Seven Du Date: Sun, 1 Nov 2009 10:24:32 +0800 Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session To: Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound and outbound. I must be very careful if I upgrade to trunk and turn on rwlock debug because it's on production and the the problem not happening that much so would be hard to trace. But I will find time to test and report back. FYI, I also noticed that in some zombile channels, couple of INVITEs sent but never got a response. However, it sent out an ACK at last. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at list... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091101/2da278e1/attachment-0002.html From lei.tlfly at gmail.com Mon Nov 2 01:24:52 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Mon, 2 Nov 2009 17:24:52 +0800 Subject: [Freeswitch-users] Get error "415 Unsupported Media Type" whenreceiving call from softswitch In-Reply-To: <50c41b4e0911020117w27ad3ca2w99590117cb1925ad@mail.gmail.com> References: <50c41b4e0911011826s3431a0cex79d4ba7eee79c872@mail.gmail.com> <2CEBE489DC2CE140B7983073B17FB3D6E5B075@301081ANEX2.global.avaya.com> <50c41b4e0911020117w27ad3ca2w99590117cb1925ad@mail.gmail.com> Message-ID: <50c41b4e0911020124s27b7203bj5d5224288e1bb790@mail.gmail.com> Hi all, The problem is solved. I ask the softswitch to send only sdp in INVITE message, then It works. I think sofia doesn't support multipart content currently. is it right? 2009/11/2 Lei Tang > Hi Daniel. > Sure. pls email me to tlfly at hotmail.com. > > 2009/11/2 Zeng, Qinglan (Daniel) > > Lei, >> >> This is Daniel Zeng and I got your email address from the maillist of FS. >> I have personal interestings on FS and if possible can we have a talk on >> this? >> >> Thanks >> Daniel Zeng >> >> ------------------------------ >> *From:* Lei Tang [mailto:lei.tlfly at gmail.com] >> *Sent:* Monday, November 02, 2009 10:27 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Get error "415 Unsupported Media Type" >> whenreceiving call from softswitch >> >> Hi all, I get a "415 Unsupported Media Type" when FS receiving call from >> a softswitch. I captured some packets, It seems that the softswitch use >> SIP-I protocol, does FS can handle SIP-I message? >> >> ===here is the invite messagefrom softswitch >> INVITE sip:xxxxx at xxxx:5060;user=phone SIP/2.0 >> Contact: >> MIME-version: 1.0 >> Content-Type: multipart/mixed;boundary=Alcatel-boundary >> To: >> From: xxxx;tag=73D332463135364195291201 >> P-Asserted-Identity: >> Allow: >> INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE >> Supported: 100rel,timer,replaces,diversion >> Expires: 155 >> Session-Expires: 1800 >> Min-SE: 90 >> Call-ID: 01FD034872814000000230A1 at sip-3 >> Max-Forwards: 70 >> CSeq: 1 INVITE >> Timestamp: 10645 >> Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E >> Content-Length: 542 >> >> --Alcatel-boundary >> Content-Type: application/sdp >> >> v=0 >> o=- 5 8 IN IP4 xxxxxx >> s=SDP Data >> c=IN IP4 xxxxx >> t=0 0 >> m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97 >> a=rtpmap:110 speex/8000/1 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:96 TELEPHONE-EVENT/8000 >> a=fmtp:110 mode=3 >> a=rtpmap:111 speex/8000/1 >> a=rtpmap:3 GSM/8000 >> a=ptime:30 >> a=fmtp:111 mode=2 >> a=rtpmap:112 speex/8000/1 >> >> --Alcatel-boundary >> Content-Type: application/ISUP;version=N/A >> >> .. .... >> ...0.D... >> ...V..0......1...n..p... >> --Alcatel-boundary-- >> >> =====response message from FS >> SIP/2.0 415 Unsupported Media Type >> Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E >> From: xxxxx ;tag=73D332463135364195291201 >> To: ;tag=m6XS44BKF4pSS >> Call-ID: 01FD034872814000000230A1 at sip-3 >> CSeq: 1 INVITE >> Timestamp: 10645 0.000000 >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 >> Accept: application/sdp >> Accept-Encoding: >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Length: 0 >> -- >> Lei.Tang >> lei.tlfly at gmail.com >> > > > > -- > Lei.Tang > lei.tlfly at gmail.com > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/3504a8c7/attachment-0002.html From dujinfang at gmail.com Mon Nov 2 02:03:39 2009 From: dujinfang at gmail.com (Seven Du) Date: Mon, 2 Nov 2009 18:03:39 +0800 Subject: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session In-Reply-To: <191c3a030911012056y66441b8aya672a16fabe89f33@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com> <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms> <191c3a030910311426of22a4fdyb6be7cabfb992bd6@mail.gmail.com> <23f91030910311924of53ae2ega246cf367f4a98a7@mail.gmail.com> <23f91030911011737l70a39195xb7a5e69df51f4865@mail.gmail.com> <191c3a030911012056j2991b3c6y3cda1b705b5f9df9@mail.gmail.com> <191c3a030911012056y66441b8aya672a16fabe89f33@mail.gmail.com> Message-ID: <23f91030911020203i182536bfn2e51604e5909a5c3@mail.gmail.com> No, I'm Seven and never used hangup hook. you must had though I was Dome. Sorry, I'm not tend to hijack this thread, just though it's the same topic. 2009/11/2 Anthony Minessale > We already concluded its your unacceptabe use of originate in hangup hook > right? > > On Nov 1, 2009 7:45 PM, "Seven Du" wrote: > > Just suspicious would be possible that happened on sqlite stage? I > manually deleted the channels from sqlite and nothing bad happend. > just FYI. > > ---------- Forwarded message ---------- > From: Seven Du > Date: Sun, 1 Nov 2009 10:24:32 +0800 > Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session > To: > > Thank you Anthony. We are on r14696 and no non-standard mods loaded. we > even > unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both > inbound > and outbound. I must be very careful if I upgrade to trunk and turn on > rwlock debug because it's on production and the the problem not happening > that much so would be hard to trace. But I will find time to test and > report > back. > > FYI, I also noticed that in some zombile channels, couple of INVITEs sent > but never got a response. However, it sent out an ACK at last. > > _______________________________________________ FreeSWITCH-users mailing > list FreeSWITCH-users at list... > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/6e73cdf0/attachment-0002.html From mariusz_kolo at wp.pl Mon Nov 2 03:32:56 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=) Date: Mon, 02 Nov 2009 12:32:56 +0100 Subject: [Freeswitch-users] Problem with hangin bri In-Reply-To: <87f2f3b90910281528t633b765x7c2bb858f41ce154@mail.gmail.com> References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com> <87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com> <4AE8B3B6.9000106@wp.pl> <87f2f3b90910281528t633b765x7c2bb858f41ce154@mail.gmail.com> Message-ID: <4AEEC368.5020204@wp.pl> Hi pastebin: http://pastebin.freeswitch.org/10926 and http://pastebin.freeswitch.org/10927 .We invoke calls from one voip phone to cell phone, and vice versa, but when i make inbound and outbound connection in nearly same time something goes wrong with chanells Thanks Michael Collins pisze: > Thanks. Can you collect debug logs of this happening? See > http://wiki.freeswitch.org/wiki/Reporting_Bugs for helpful tips on > collecting debug information. Use pastebin to dump all the log info > and reply here with the link. We don't have too many BRI users but I > believe there are a few so hopefully we can help you get up and running. > -MC > > 2009/10/28 Mariusz Ko?odziejczyk > > > Hi > > I'm also working on this project, so i can answer your questions > > Which version of FreeSWITCH are you running? > > FreeSWITCH Version 1.0.trunk (15246) > > Which PRI library are you using? > openzap Native stack > > openzap.conf > > [span zt BRI1] > trunk_type => bri > b-channel => 1-2 > d-channel=> 3 > > openzap.conf > > > > > > > > > > > > > > > > > Which BRI card are you using? > > Producer: http://www.phoniceq.com/ > card model: http://quadbri.phoniceq.com/ > > Card instalation process (instruction from producer) > > 1) download bristuff staff from > > http://junghanns.net/downloads/bristuff-0.4.0-RC3h.tar.gz > or > http://junghanns.net/downloads/bristuff-0.3.0-PRE-1y-z.tar.gz > > unpack it and go to bristuff-* > > 2) download patcher from > http://quadbri.phoniceq.com/driver/bristuff/qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch > > patch it using > > patch -p0 < qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch > > 3) you can check card using zttest (result should be 99.x) > > Producer has said, that we are first client, it wants to use this > card in freeswitch > > we are using 1 port (S/T interface). Our NT is "NT1 plus 2b1q" > > > Thanks > > Michael Collins pisze: > > Okay, obligatory questions: > > Which version of FreeSWITCH are you running? > > Which PRI library are you using? > > Which BRI card are you using? > > > > -MC > > > > On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawli?ski > > > >> wrote: > > > > Hi, > > I have some problems with bri status. I have 3 chanel isdn > modem, > > and zaptel compatible quad bri card. I can invoke calls from my > > voip phone to cell phone, and vice versa, but when i make > inbound > > and outbound connection in nearly same time something goes wrong > > with chanells and after few calls all of them has hangup status. > > > > There is log about that in attachement, see "is already in use > > waiting for it to become available." phrase. Time of this > event is > > about 14:43:35. Unload and Load open_zap module helped, but its > > not an solution because of lost connections. > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Mariusz Ko?odziejczyk > > Advanced Developing Architecture S.C. > > tel. : +48 609 381 316 > e-mail : mariusz_kolo at wp.pl > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Mariusz Ko?odziejczyk Advanced Developing Architecture S.C. tel. : +48 609 381 316 e-mail : mariusz_kolo at wp.pl From dipen at entvoice.com Mon Nov 2 03:53:51 2009 From: dipen at entvoice.com (dipen at entvoice.com) Date: Mon, 2 Nov 2009 06:53:51 -0500 Subject: [Freeswitch-users] Java example Message-ID: <44498.1257162831@entvoice.com> Hi, Can you please paste me your sample java dialplan code that work for you ? ..coz m also facing the same problem. My mod_java is loaded properly. Also /usr/lib/jvm/java-1.5.0-gcj-4.3-1.5.0.0/jre/lib/i386/client/libjvm.so and freeswitch.jar in java.conf.xml is specified properly. I have written a java code to print HIIIIIIIIII on the console but its not printing. Level mentioned is INFO. on FS console it just prints EXECUTE sofia/internal/1004 at 192.168.1.144:5061 java(testing.class) I am attaching my java code herewith. Can u just tell me where more i should do the modification to get my dialplan work. Waiting for your kind reply. Thanks & Regards, Dipen Velani On Fri 19/12/08 4:09 AM , kriko wrote: > Seems like my dialplan was a bit problematic, it works now. > Thanks. > On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale > wrote: > > did you turn up your console log level high enough to see it? The > default > > level is "INFO" > > > > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > -- > kriko > _______________________________________________ > Freeswitch-users mailing list > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ---- Msg sent via @Mail - http://atmail.com/ -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: java_sample_code.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/9b737d5a/attachment-0004.txt From dipen at entvoice.com Mon Nov 2 03:53:51 2009 From: dipen at entvoice.com (dipen at entvoice.com) Date: Mon, 2 Nov 2009 06:53:51 -0500 Subject: [Freeswitch-users] Java example Message-ID: <44498.1257162831@entvoice.com> Hi, Can you please paste me your sample java dialplan code that work for you ? ..coz m also facing the same problem. My mod_java is loaded properly. Also /usr/lib/jvm/java-1.5.0-gcj-4.3-1.5.0.0/jre/lib/i386/client/libjvm.so and freeswitch.jar in java.conf.xml is specified properly. I have written a java code to print HIIIIIIIIII on the console but its not printing. Level mentioned is INFO. on FS console it just prints EXECUTE sofia/internal/1004 at 192.168.1.144:5061 java(testing.class) I am attaching my java code herewith. Can u just tell me where more i should do the modification to get my dialplan work. Waiting for your kind reply. Thanks & Regards, Dipen Velani On Fri 19/12/08 4:09 AM , kriko wrote: > Seems like my dialplan was a bit problematic, it works now. > Thanks. > On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale > wrote: > > did you turn up your console log level high enough to see it? The > default > > level is "INFO" > > > > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > -- > kriko > _______________________________________________ > Freeswitch-users mailing list > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ---- Msg sent via @Mail - http://atmail.com/ -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: java_sample_code.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/9b737d5a/attachment-0005.txt From info at daccii.it Mon Nov 2 05:04:24 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Mon, 02 Nov 2009 14:04:24 +0100 Subject: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone after the first call using OpenZAP (analog spans) In-Reply-To: <4AEC303F.4020207@daccii.it> References: <4AE8B20D.4040608@daccii.it> <191c3a030910290912x9489784s4cc8e224363f3796@mail.gmail.com> <87f2f3b90910291026u2c62ded6v5b9458a9c2e85877@mail.gmail.com> <4AEC303F.4020207@daccii.it> Message-ID: <4AEED8D8.6050303@daccii.it> Hi, i've done more and more tests ... the result is the same :\ I've tried previous freeswitch version (1.0.2, 1.0.3), lastest stable (1.0.4) and with svn (updated at revision 15315 while openzap revision is 847). I've tried with ubuntu zaptel modules (1.4.10), with and without octvqe soft echo, with another card, that uses wctdm instead of opvxa1200, with lastest manually compiled zaptel modules (1.4.12.1) but, yet, nothing to do. I tested openzap test utilities but them works well (only a little change to testanalog.c to use it tones instead of us). The problem is ever the same, i can do the first call but i can't do more. I put logs and config files into freeswitch pastebin, here links: - FULL STARTUP http://pastebin.freeswitch.org/10930 - ZAPTEL STARTUP http://pastebin.freeswitch.org/10929 - FIRST CALL http://pastebin.freeswitch.org/10928 - SECOND CALL http://pastebin.freeswitch.org/10931 - INCOMING CALL http://pastebin.freeswitch.org/10939 - SHUTDOWN http://pastebin.freeswitch.org/10932 - OPENVOX DMESG http://pastebin.freeswitch.org/10933 - zaptel.conf http://pastebin.freeswitch.org/10934 - openzap.conf http://pastebin.freeswitch.org/10935 - zt.conf http://pastebin.freeswitch.org/10936 - openzap.conf.xml http://pastebin.freeswitch.org/10937 - openzap dialplan for outgoing/incoming calls http://pastebin.freeswitch.org/10938 As you can see from logs freeswitch doesn't reknow free dial tone after the first call, but i don't understand why. I'll join on irc in short, hope to find some help :) Best Regards, Daniele Albano Daniele Salvatore - Lavoro ha scritto: > Hi, > > i've done more tests, with svn too, but nothing to do. The strange thing > is when freeswitch shutdown and close zaptel channel i get back the dial > tone from my provider (i'm checking trought ztmonitor 1 -vv). Can be > something is missing when span get closed? > > PS1: with svn i can do only one call, but i can receive calls without > problems > PS2: i joined yesterday on irc but i had few time ... hope to join monday > > > Best Regards, > Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/cbc035a6/attachment-0002.vcf From hjqlopez at hotmail.com Mon Nov 2 06:36:50 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Mon, 2 Nov 2009 09:36:50 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media Message-ID: Thanks for you answers guys, I test the parameters you suggested but still no audio due to the lack of reINVITE.? By the way I'm using 1.0.4 but I also tried 1.0.5pre3. One particular condition is that there is no on-hold before the Blind Transfer. Regards, Humberto >? >? >> My scenario is as follows: >> >> inbound-bypass-media is set in the profile because we dont want FS handling >> the media. >> >> 1. A calls B >> 2. FS sends to B the A's SDP >> 3. B answers >> 4. FS sends to A the B's SDP >> 5. Media going directly between A and B >> 6. B REFERs the call to C (blind transfer with no reINVITE for Hold) >> 7. FS accepts(202) the REFER and sends the NOTIFY >> 7a. B and FS send the BYE >> 8. FS sends an INVITE? to C with A's SDP >> 9. C answers >> 10. FS doesn't send a reINVITE to A to let it know about C's SDP >> >> >> Is that the expected FS behavior or is this a bug? _________________________________________________________________ Ready for a deal-of-a-lifetime? See fantastic offers on Windows 7, in one convenient place. http://go.microsoft.com/?linkid=9691634 From mike at jerris.com Mon Nov 2 06:45:26 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 2 Nov 2009 09:45:26 -0500 Subject: [Freeswitch-users] Get error "415 Unsupported Media Type" whenreceiving call from softswitch In-Reply-To: <50c41b4e0911020124s27b7203bj5d5224288e1bb790@mail.gmail.com> References: <50c41b4e0911011826s3431a0cex79d4ba7eee79c872@mail.gmail.com> <2CEBE489DC2CE140B7983073B17FB3D6E5B075@301081ANEX2.global.avaya.com> <50c41b4e0911020117w27ad3ca2w99590117cb1925ad@mail.gmail.com> <50c41b4e0911020124s27b7203bj5d5224288e1bb790@mail.gmail.com> Message-ID: <724DAF7A-537D-4588-AE73-7B06076DE78E@jerris.com> That is correct. Mike On Nov 2, 2009, at 4:24 AM, Lei Tang wrote: > Hi all, > The problem is solved. I ask the softswitch to send only sdp in > INVITE message, then It works. > I think sofia doesn't support multipart content currently. is it > right? From mike at jerris.com Mon Nov 2 06:49:22 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 2 Nov 2009 09:49:22 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <177068F1-7F95-4AB2-AF60-E1B367B49213@jerris.com> Please re-try with latest svn trunk. Mike On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote: > > Thanks for you answers guys, > > I test the parameters you suggested > but still no audio due to the lack of reINVITE. By the way I'm using > 1.0.4 but I also tried 1.0.5pre3. > > One particular condition is that there is no on-hold before the > Blind Transfer. > > Regards, > > Humberto > >> >> From anthony.minessale at gmail.com Mon Nov 2 07:32:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Nov 2009 09:32:54 -0600 Subject: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session In-Reply-To: <23f91030911020203i182536bfn2e51604e5909a5c3@mail.gmail.com> References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com> <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms> <191c3a030910311426of22a4fdyb6be7cabfb992bd6@mail.gmail.com> <23f91030910311924of53ae2ega246cf367f4a98a7@mail.gmail.com> <23f91030911011737l70a39195xb7a5e69df51f4865@mail.gmail.com> <191c3a030911012056j2991b3c6y3cda1b705b5f9df9@mail.gmail.com> <191c3a030911012056y66441b8aya672a16fabe89f33@mail.gmail.com> <23f91030911020203i182536bfn2e51604e5909a5c3@mail.gmail.com> Message-ID: <191c3a030911020732r4b043c52r975ea31a7b35e4d0@mail.gmail.com> Every time you have stuck channels at the last state it means something took control of the thread and did not release it. revisions other that current SVN trunk are not possible to debug because over one thousand changes have occurred since then. On Mon, Nov 2, 2009 at 4:03 AM, Seven Du wrote: > No, I'm Seven and never used hangup hook. you must had though I was Dome. > > Sorry, I'm not tend to hijack this thread, just though it's the same topic. > > 2009/11/2 Anthony Minessale > >> We already concluded its your unacceptabe use of originate in hangup hook >> right? >> >> On Nov 1, 2009 7:45 PM, "Seven Du" wrote: >> >> Just suspicious would be possible that happened on sqlite stage? I >> manually deleted the channels from sqlite and nothing bad happend. >> just FYI. >> >> ---------- Forwarded message ---------- >> From: Seven Du >> Date: Sun, 1 Nov 2009 10:24:32 +0800 >> Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session >> To: >> >> Thank you Anthony. We are on r14696 and no non-standard mods loaded. we >> even >> unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both >> inbound >> and outbound. I must be very careful if I upgrade to trunk and turn on >> rwlock debug because it's on production and the the problem not happening >> that much so would be hard to trace. But I will find time to test and >> report >> back. >> >> FYI, I also noticed that in some zombile channels, couple of INVITEs sent >> but never got a response. However, it sent out an ACK at last. >> >> _______________________________________________ FreeSWITCH-users mailing >> list FreeSWITCH-users at list... >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/61fc2204/attachment-0002.html From ujjval at simplesignal.com Mon Nov 2 07:54:33 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Mon, 2 Nov 2009 07:54:33 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> , <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> Message-ID: <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> Yes, I think I did. However here is what furthur testing revelas. If I dial in from AT&T cell phone, I do not see any DTMF using Don's IVR.xml.conf to call my conf app. But when I dial the same number using a Verizon Cell, it works. When I dial a number that is provisioned to call the Conf App directly from the public.xml dialplan...it works even with the same AT&T cell phone... Strange behaviour ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris [mike at jerris.com] Sent: Saturday, October 31, 2009 11:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Have you answered the call? On Oct 30, 2009, at 11:34 AM, Rob Forman wrote: > Hm, strange. I haven't seen that before. Can you pastebin your logs > at debug level? > > On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > >> It's strange... a tcpdump tells me that there is no DTMF from my >> provider when using IVR, but when I call into a TN that goes >> directly into the Conference App, I see DTMF from the provider. >> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Mon Nov 2 08:08:29 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Nov 2009 10:08:29 -0600 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> , <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> Message-ID: <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> you know I have heard this before... It seems to ONLY be AT&T /b On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote: > Yes, I think I did. However here is what furthur testing revelas. If > I dial in from AT&T cell phone, I do not see any DTMF using Don's > IVR.xml.conf to call my conf app. But when I dial the same number > using a Verizon Cell, it works. > > When I dial a number that is provisioned to call the Conf App > directly from the public.xml dialplan...it works even with the same > AT&T cell phone... > > Strange behaviour From djbinter at yahoo.com Mon Nov 2 08:52:34 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 2 Nov 2009 08:52:34 -0800 (PST) Subject: [Freeswitch-users] CDR CSV variables In-Reply-To: <191c3a030911011140q505868c3p5a9c94b5decbf857@mail.gmail.com> References: <609781.88322.qm@web37502.mail.mud.yahoo.com> <191c3a030911011140q505868c3p5a9c94b5decbf857@mail.gmail.com> Message-ID: <430543.96247.qm@web37502.mail.mud.yahoo.com> Anthony, Yes, if you can advise, how would I detect whether it's going out to 192.168.1.4 or 192.168.1.5 without having to activate b-leg of the CDRs. Thank you, Dorn B. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Sun, November 1, 2009 11:40:21 AM Subject: Re: [Freeswitch-users] CDR CSV variables you can make up your own variable and set whatever you want in there then add it to the template. On Fri, Oct 30, 2009 at 10:04 AM, DJB wrote: I wonder if I don't want to have b-leg in cdr csv, is there any variables that can give me the actual gateway ip address that is actually went out. > >>For instance, if I have this in my dialplan: >> >>the only value that I can think of from cdr csv is to get remote_ip_last_arg, but it would contains the whole line of both ip addresses. > >>Thank you, >>Dorn B. > > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/f0173889/attachment-0002.html From djbinter at yahoo.com Mon Nov 2 08:56:34 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 2 Nov 2009 08:56:34 -0800 (PST) Subject: [Freeswitch-users] Freeswitch in signaling path only In-Reply-To: <312533.76921.qm@web37505.mail.mud.yahoo.com> References: <429925.60305.qm@web37508.mail.mud.yahoo.com> <200910301300.01766.chris@cloudtel.com> <312533.76921.qm@web37505.mail.mud.yahoo.com> Message-ID: <612508.14697.qm@web37508.mail.mud.yahoo.com> Any suggestion from anyone please? Thank you, Dorn B ----- Original Message ---- From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Fri, October 30, 2009 11:31:12 AM Subject: Re: [Freeswitch-users] Freeswitch in signaling path only Now i have as follows, but it's still the same result. By the way, I am running: FreeSWITCH Version 1.0.4 (exported) . . . session:execute("set","hangup_after_bridge=true") session:execute("set","continue_on_fail=true") session:execute("set","originate_timeout=2") session:execute("set","originate_retries=3") session:execute("set","progress_timeout=15") . . . while row do local gw_ip_address = row.gw_ip_address local cust_name = row.cust_name session:execute("set", "accountcode=" ..cust_name .. "") session:execute("set","bypass_media=true") session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.146") session:execute("set","bypass_media=true") session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.105") -- Block for testing -- session:execute("bridge","sofia/external/" .. called_num .. "@" .. gw_ip_address .."") row = cur:fetch (row, "a") end Here is the debug for switch_ivr_originate.c: 2009-10-30 11:09:52.877832 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:09:53.17811 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.1 46] 2009-10-30 11:09:54.285453 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:09:54.422426 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 105] 2009-10-30 11:09:55.694761 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:09:55.836036 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 146] 2009-10-30 11:09:57.107697 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:09:57.254664 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 105] 2009-10-30 11:12:03.129097 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:12:03.273055 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 146] 2009-10-30 11:12:04.546410 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:12:04.682661 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 105] 2009-10-30 11:12:15.781701 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 16 [NORMAL_CLEARING] 2009-10-30 11:12:33.349162 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C S_CONSUME_MEDIA 2009-10-30 11:12:33.470989 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. 146] 2009-10-30 11:12:34.724641 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2009-10-30 11:12:34.730634 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2009-10-30 11:12:34.750637 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] FIRST ROUTE: XX.XX.XX.146 and I tried to failed the first route and it gave 500 back, then it goes to the next one. SECOND ROUTE: XX.XX.XX.105 Thank you, Dorn B. ----- Original Message ---- From: Chris Burns To: freeswitch-users at lists.freeswitch.org Sent: Fri, October 30, 2009 10:00:01 AM Subject: Re: [Freeswitch-users] Freeswitch in signaling path only Do you have a debug level message from switch_ivr_originate.c in your log? "Channel is already up, delaying proxy mode 'till both legs are answered." Set bypass_media b4 each bridge. It is unsetting on you and setting bypass_media_after_bridge because you already answered the channel running the lua script. On October 30, 2009 12:03:29 pm DJB wrote: > I am wondering why I cannot do as condition#2. > > For Lua in dialplan, when I have the followings: > > > --WORKING-- > (Condition#1) > . > . > session:execute("set","bypass_media=true") > session:execute("set","hangup_after_bridge=true") > session:execute("set","continue_on_fail=true") > . > . > session:execute("bridge","sofia/external/" .. called_num .. > "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") . > . > > --NOT WORKING-- > (Condition#2) > Note: FS tries to be in media path and send re-invite. > . > . > session:execute("set","bypass_media=true") > session:execute("set","hangup_after_bridge=true") > session:execute("set","continue_on_fail=true") > . > . > session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1") > session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2") > . > . > > Thank you, > Dorn B. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Mon Nov 2 09:07:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Nov 2009 11:07:26 -0600 Subject: [Freeswitch-users] Freeswitch in signaling path only In-Reply-To: <612508.14697.qm@web37508.mail.mud.yahoo.com> References: <429925.60305.qm@web37508.mail.mud.yahoo.com> <200910301300.01766.chris@cloudtel.com> <312533.76921.qm@web37505.mail.mud.yahoo.com> <612508.14697.qm@web37508.mail.mud.yahoo.com> Message-ID: <191c3a030911020907o7718bc3ci7ebfee40519e01cb@mail.gmail.com> yes please use our latest SVN trunk if you plan to report issues because our first questions always is "can you reproduce this issue with our latest code" And please don't reply twice to the same email asking us to hurry up an answer you faster its only monday morning here. On Mon, Nov 2, 2009 at 10:56 AM, DJB wrote: > Any suggestion from anyone please? > > Thank you, > Dorn B > > > ----- Original Message ---- > From: DJB > To: freeswitch-users at lists.freeswitch.org > Sent: Fri, October 30, 2009 11:31:12 AM > Subject: Re: [Freeswitch-users] Freeswitch in signaling path only > > Now i have as follows, but it's still the same result. By the way, I am > running: FreeSWITCH Version 1.0.4 (exported) > . > . > . > session:execute("set","hangup_after_bridge=true") > session:execute("set","continue_on_fail=true") > session:execute("set","originate_timeout=2") > session:execute("set","originate_retries=3") > session:execute("set","progress_timeout=15") > . > . > . > while row do > local gw_ip_address = row.gw_ip_address > local cust_name = row.cust_name > session:execute("set", "accountcode=" ..cust_name .. "") > session:execute("set","bypass_media=true") > session:execute("bridge","sofia/external/" .. called_num .. > "@XX.XX.XX.XX.146") > session:execute("set","bypass_media=true") > session:execute("bridge","sofia/external/" .. called_num .. > "@XX.XX.XX.XX.105") > -- Block for testing -- session:execute("bridge","sofia/external/" .. > called_num .. "@" .. gw_ip_address .."") > row = cur:fetch (row, "a") > end > > Here is the debug for switch_ivr_originate.c: > > 2009-10-30 11:09:52.877832 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:09:53.17811 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.1 > 46] > 2009-10-30 11:09:54.285453 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:09:54.422426 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 105] > 2009-10-30 11:09:55.694761 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:09:55.836036 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 146] > 2009-10-30 11:09:57.107697 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:09:57.254664 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 105] > 2009-10-30 11:12:03.129097 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:12:03.273055 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 146] > 2009-10-30 11:12:04.546410 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:12:04.682661 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 105] > 2009-10-30 11:12:15.781701 [DEBUG] switch_ivr_originate.c:2138 Originate > Resulted in Error Cause: 16 [NORMAL_CLEARING] > 2009-10-30 11:12:33.349162 [DEBUG] switch_ivr_originate.c:63 > (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C > S_CONSUME_MEDIA > 2009-10-30 11:12:33.470989 [DEBUG] switch_ivr_originate.c:2061 Originate > Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. > 146] > 2009-10-30 11:12:34.724641 [DEBUG] switch_ivr_originate.c:2138 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > 2009-10-30 11:12:34.730634 [DEBUG] switch_ivr_originate.c:2138 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > 2009-10-30 11:12:34.750637 [DEBUG] switch_ivr_originate.c:2138 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > > > FIRST ROUTE: XX.XX.XX.146 > and I tried to failed the first route and it gave 500 back, then it goes to > the next one. > SECOND ROUTE: XX.XX.XX.105 > > > Thank you, > Dorn B. > > > > ----- Original Message ---- > From: Chris Burns > To: freeswitch-users at lists.freeswitch.org > Sent: Fri, October 30, 2009 10:00:01 AM > Subject: Re: [Freeswitch-users] Freeswitch in signaling path only > > Do you have a debug level message from switch_ivr_originate.c in your log? > "Channel is already up, delaying proxy mode 'till both legs are answered." > > Set bypass_media b4 each bridge. It is unsetting on you and setting > bypass_media_after_bridge because you already answered the channel running > the lua script. > > On October 30, 2009 12:03:29 pm DJB wrote: > > I am wondering why I cannot do as condition#2. > > > > For Lua in dialplan, when I have the followings: > > > > > > --WORKING-- > > (Condition#1) > > . > > . > > session:execute("set","bypass_media=true") > > session:execute("set","hangup_after_bridge=true") > > session:execute("set","continue_on_fail=true") > > . > > . > > session:execute("bridge","sofia/external/" .. called_num .. > > "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") . > > . > > > > --NOT WORKING-- > > (Condition#2) > > Note: FS tries to be in media path and send re-invite. > > . > > . > > session:execute("set","bypass_media=true") > > session:execute("set","hangup_after_bridge=true") > > session:execute("set","continue_on_fail=true") > > . > > . > > session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1") > > session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2") > > . > > . > > > > Thank you, > > Dorn B. > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/8a3ef38e/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 2 09:09:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Nov 2009 11:09:44 -0600 Subject: [Freeswitch-users] CDR CSV variables In-Reply-To: <430543.96247.qm@web37502.mail.mud.yahoo.com> References: <609781.88322.qm@web37502.mail.mud.yahoo.com> <191c3a030911011140q505868c3p5a9c94b5decbf857@mail.gmail.com> <430543.96247.qm@web37502.mail.mud.yahoo.com> Message-ID: <191c3a030911020909r39a3687fq733189943bb07cd7@mail.gmail.com> if you enable debug on the cdr_csv module you will get a big dump of all the data you have available and you may be able to pick something out that indicates which one it was. On Mon, Nov 2, 2009 at 10:52 AM, DJB wrote: > Anthony, > > Yes, if you can advise, how would I detect whether it's going out to > 192.168.1.4 or 192.168.1.5 without having to activate b-leg of the CDRs. > > Thank you, > Dorn B. > > ------------------------------ > *From:* Anthony Minessale > > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Sun, November 1, 2009 11:40:21 AM > *Subject:* Re: [Freeswitch-users] CDR CSV variables > > you can make up your own variable and set whatever you want in there then > add it to the template. > > > On Fri, Oct 30, 2009 at 10:04 AM, DJB wrote: > >> I wonder if I don't want to have b-leg in cdr csv, is there any variables >> that can give me the actual gateway ip address that is actually went out. >> >> For instance, if I have this in my dialplan: >> >> the only value that I can think of from cdr csv is to get >> remote_ip_last_arg, but it would contains the whole line of both ip >> addresses. >> >> Thank you, >> Dorn B. >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/b1b99b88/attachment-0002.html From djbinter at yahoo.com Mon Nov 2 09:39:28 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 2 Nov 2009 09:39:28 -0800 (PST) Subject: [Freeswitch-users] Freeswitch in signaling path only In-Reply-To: <191c3a030911020907o7718bc3ci7ebfee40519e01cb@mail.gmail.com> References: <429925.60305.qm@web37508.mail.mud.yahoo.com> <200910301300.01766.chris@cloudtel.com> <312533.76921.qm@web37505.mail.mud.yahoo.com> <612508.14697.qm@web37508.mail.mud.yahoo.com> <191c3a030911020907o7718bc3ci7ebfee40519e01cb@mail.gmail.com> Message-ID: <147606.16838.qm@web37506.mail.mud.yahoo.com> I am really sorry. I did not mean to rush or anything. I've had a problem with my email many times, so I just want to make sure that my email gets there. Regards, Dorn B. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Mon, November 2, 2009 9:07:26 AM Subject: Re: [Freeswitch-users] Freeswitch in signaling path only yes please use our latest SVN trunk if you plan to report issues because our first questions always is "can you reproduce this issue with our latest code" And please don't reply twice to the same email asking us to hurry up an answer you faster its only monday morning here. On Mon, Nov 2, 2009 at 10:56 AM, DJB wrote: >Any suggestion from anyone please? > > >>Thank you, >>Dorn B > > >>----- Original Message ---- > >From: DJB >>To: freeswitch-users at lists.freeswitch.org > >Sent: Fri, October 30, 2009 11:31:12 AM >>Subject: Re: [Freeswitch-users] Freeswitch in signaling path only > >>Now i have as follows, but it's still the same result. By the way, I am running: FreeSWITCH Version 1.0.4 (exported) >>. >>. >>. >>session:execute("set","hangup_after_bridge=true") >>session:execute("set","continue_on_fail=true") >>session:execute("set","originate_timeout=2") >>session:execute("set","originate_retries=3") >>session:execute("set","progress_timeout=15") >>. >>. >>. >>while row do >>local gw_ip_address = row.gw_ip_address >>local cust_name = row.cust_name >>session:execute("set", "accountcode=" ..cust_name .. "") >>session:execute("set","bypass_media=true") >>session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.146") >>session:execute("set","bypass_media=true") >>session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.105") >>-- Block for testing -- session:execute("bridge","sofia/external/" .. called_num .. "@" .. gw_ip_address .."") >>row = cur:fetch (row, "a") >>end > >>Here is the debug for switch_ivr_originate.c: > >>2009-10-30 11:09:52.877832 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:09:53.17811 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.1 >>46] >>2009-10-30 11:09:54.285453 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:09:54.422426 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>105] >>2009-10-30 11:09:55.694761 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:09:55.836036 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>146] >>2009-10-30 11:09:57.107697 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:09:57.254664 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>105] >>2009-10-30 11:12:03.129097 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:12:03.273055 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>146] >>2009-10-30 11:12:04.546410 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:12:04.682661 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>105] >>2009-10-30 11:12:15.781701 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 16 [NORMAL_CLEARING] >>2009-10-30 11:12:33.349162 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C >>S_CONSUME_MEDIA >>2009-10-30 11:12:33.470989 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX. >>146] >>2009-10-30 11:12:34.724641 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >>2009-10-30 11:12:34.730634 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >>2009-10-30 11:12:34.750637 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > > >>FIRST ROUTE: XX.XX.XX.146 >>and I tried to failed the first route and it gave 500 back, then it goes to the next one. >>SECOND ROUTE: XX.XX.XX.105 > > >>Thank you, >>Dorn B. > > > >>----- Original Message ---- >>From: Chris Burns >>To: freeswitch-users at lists.freeswitch.org >>Sent: Fri, October 30, 2009 10:00:01 AM >>Subject: Re: [Freeswitch-users] Freeswitch in signaling path only > >>Do you have a debug level message from switch_ivr_originate.c in your log? >>"Channel is already up, delaying proxy mode 'till both legs are answered." > >>Set bypass_media b4 each bridge. It is unsetting on you and setting >>bypass_media_after_bridge because you already answered the channel running >>the lua script. > >>On October 30, 2009 12:03:29 pm DJB wrote: >>> I am wondering why I cannot do as condition#2. >>> >>> For Lua in dialplan, when I have the followings: >>> >>> >>> --WORKING-- >>> (Condition#1) >>> . >>> . >>> session:execute("set","bypass_media=true") >>> session:execute("set","hangup_after_bridge=true") >>> session:execute("set","continue_on_fail=true") >>> . >>> . >>> session:execute("bridge","sofia/external/" .. called_num .. >>> "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") . >>> . >>> >>> --NOT WORKING-- >>> (Condition#2) >>> Note: FS tries to be in media path and send re-invite. >>> . >>> . >>> session:execute("set","bypass_media=true") >>> session:execute("set","hangup_after_bridge=true") >>> session:execute("set","continue_on_fail=true") >>> . >>> . >>> session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1") >>> session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2") >>> . >>> . >>> >>> Thank you, >>> Dorn B. >>> >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/9fc97856/attachment-0002.html From shiyanov at gmail.com Mon Nov 2 10:51:59 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Mon, 2 Nov 2009 21:51:59 +0300 Subject: [Freeswitch-users] Java example In-Reply-To: <44498.1257162831@entvoice.com> References: <44498.1257162831@entvoice.com> Message-ID: Here is rather big and, let's say, complete example of mod_java usage: https://starpound.svn.sourceforge.net/svnroot/starpound/trunk/src/fs2agi The goal of this project is to be a proxy between FreeSwitch and server application which knows Asterisk AGI. On Mon, Nov 2, 2009 at 2:53 PM, wrote: > > Hi, > > Can you please paste me your sample java dialplan code that work for you ? > ..coz m also facing the same problem. > > My mod_java is loaded properly. > Also /usr/lib/jvm/java-1.5.0-gcj-4.3-1.5.0.0/jre/lib/i386/client/libjvm.so > and freeswitch.jar in java.conf.xml is specified properly. > > I have written a java code to print HIIIIIIIIII on the console but its not > printing. Level mentioned is INFO. > > on FS console it just prints > EXECUTE sofia/internal/1004 at 192.168.1.144:5061 java(testing.class) > > > I am attaching my java code herewith. > Can u just tell me where more i should do the modification to get my > dialplan work. > > Waiting for your kind reply. > > Thanks & Regards, > Dipen Velani > > On Fri 19/12/08 4:09 AM , kriko wrote: > > > Seems like my dialplan was a bit problematic, it works now. > > Thanks. > > On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale > > wrote: > > > did you turn up your console log level high enough to see it? The > > default > > > level is "INFO" > > > > > > > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > > > > > > > > > > -- > > kriko > > _______________________________________________ > > Freeswitch-users mailing list > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ---- Msg sent via @Mail - http://atmail.com/ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/ed780c43/attachment-0002.html From brian at freeswitch.org Mon Nov 2 11:07:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Nov 2009 13:07:08 -0600 Subject: [Freeswitch-users] Java example In-Reply-To: References: <44498.1257162831@entvoice.com> Message-ID: <85845D7B-9D9D-4BFA-ACCA-0F28DA4EBA9E@freeswitch.org> Is starpound involved in the FS Community? /b On Nov 2, 2009, at 12:51 PM, Artem Shiyanov wrote: > Here is rather big and, let's say, complete example of mod_java usage: > https://starpound.svn.sourceforge.net/svnroot/starpound/trunk/src/fs2agi > The goal of this project is to be a proxy between FreeSwitch and > server application which knows Asterisk AGI. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/3c5a6fc8/attachment-0002.html From ujjval at simplesignal.com Mon Nov 2 11:51:39 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Mon, 2 Nov 2009 11:51:39 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> Message-ID: <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> Rob: Once I have the Moderator and Participants logged on, how do I invoke the moderator previlidges, LIk esay muting everyone/someone or kicking someone out of the Conf and the like? Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO? 80112 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Friday, October 30, 2009 9:34 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hm, strange. I haven't seen that before. Can you pastebin your logs at debug level? On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > It's strange... a tcpdump tells me that there is no DTMF from my > provider when using IVR, but when I call into a TN that goes > directly into the Conference App, I see DTMF from the provider. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 7:23 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > I've never had any problem with that. Is your logging at debug level > so you can see the RECV DTFM in the log/fs_cli? Are you calling from > a SIP phone on the pbx, or via a PSTN provider? Maybe your provider > isn't passing them through. > > Make sure your logging is turned up then try something simpler, like > calling the echo application, and see if DTFM comes through. > > Rob > > On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: > >> Rob: >> >> For some reason, I don't see the DTMF appear on the fs_CLI when >> using the below configuration....so it basically timesout. >> >> UK >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, October 26, 2009 9:21 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Thx a lot Rob, reading the wiki your way or using IVR seems correct.. >> =============== >> The wiki also says that the wait-mod might be "used in conjunction >> with an IVR where the moderators are authenticated with an extra >> pass- >> code", which is what I did. I guess that's why I didn't understand >> the point of the +pin. >> ====================== >> >> I will try it out. >> >> Again thx a lot for your help. Will keep everyone posted. >> >> ________________________________________ >> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >> Sent: Friday, October 23, 2009 12:22 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I just re-tested with the pin in my dial plan: >> >> >> >> And it doesn't challenge me for the pin. I just drop right in. I >> figured this is how it was intended, since the wiki says the pin is >> set initially and only challenged in later attempts [by future >> callers]: >> >> "The first time a conference name (confname) is used, it will be >> created on demand, and the pin will be set to what ever is specified >> at that time: the pin in the data string if specified, or if not, the >> "pin" setting in the conference profile, and if that is also >> unspecified, then there is no pin protection. Any later attempt to >> join the conference must specify the same pin number, if one existed >> when it was created. " >> >> >> The wiki also says that the wait-mod might be "used in conjunction >> with an IVR where the moderators are authenticated with an extra >> pass- >> code", which is what I did. I guess that's why I didn't understand >> the point of the +pin. >> >> I'm sure there's a scenario where its used and useful, the wiki just >> doesn't explain it. >> >> Rob >> >> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >> >>> Well first off you're not defining a pine here... >>> >>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>> pin >>> number] >>> >>> That might be why its not asking for a pin. >>> >>> /b >>> >>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>> >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ivan at myrvold.org Mon Nov 2 12:58:22 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Mon, 2 Nov 2009 21:58:22 +0100 Subject: [Freeswitch-users] SIP provider with extern rtp server In-Reply-To: <191c3a030911011220m22d7b515kda412e1fd9408f59@mail.gmail.com> References: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com> <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org> <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org> <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org> <191c3a030911011220m22d7b515kda412e1fd9408f59@mail.gmail.com> Message-ID: <03E9A3AC-F501-4FAE-8199-A4AEC4D60891@myrvold.org> That was it. My sip provider applied the patch to his Asterisk server that was referenced in the link you was so kind to provide, and again everything worked as it should. Thank you very much! Ivan Den 1. nov. 2009 kl. 21:20 skrev Anthony Minessale: > Session-Expires: -1;refresher=uas > > nta: 200 OK has fatal syntax errors > > This is a know-bug in asterisk. > > see: https://issues.asterisk.org/view.php?id=15621 > > > > On Sun, Nov 1, 2009 at 4:40 AM, Ivan C Myrvold > wrote: > No one have any idea why this is not working? I have combed through > the log, but couldn't find any clue there. > Incoming calls from my sip provider is working perfect, but for > outgoing calls it looks like Freeswitch is not letting the incoming > rtp to the local sip phone. > > Ivan > > On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote: > > > Yes, now I got a more detailed trace. Thank you for helping me with > > this. > > > > A new pastebin at http://pastebin.freeswitch.org/10905 > > > > Ivan > > > > Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable: > > > >> fsctl loglevel debug > >> console loglevel debug > >> sofia profile internal siptrace on > >> sofia profile external siptrace on > >> sofia loglevel all 9 > >> ^^^^^^^^^^^^^^^^^^^^^ > >> > >> Then run your call, then do this: > >> > >> sofia loglevel all 0 > >> sofia profile external siptrace off > >> sofia profile internal siptrace off > >> fsctl loglevel warning > >> console loglevel warning > >> > >> On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold > >> wrote: > >>> I have already set debug to 9, on both profiles. > >>> > >>> Ivan > >>> > >>> > >>> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable: > >>> > >>>> See that 200 OK that keeps coming in over and over and over and > >>>> over > >>>> again? That's because they never received your ACK. If you can > >>>> turn on > >>>> sofia loglevel to 9 and then watch where you send the ACK, you > will > >>>> probably have your answer to why the other system did not receive > >>>> it. > >>>> If you're still not sure what's going on, post another pastebin > >>>> with > >>>> sofia loglevel set to 9. > >>>> > >>>> > >>>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold > > >>>> wrote: > >>>>> Oh, what happened to it? > >>>>> Anyway, here is a new pb: > >>>>> http://pastebin.freeswitch.org/10867 > >>>>> Ivan > >>>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins: > >>>>> > >>>>> > >>>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold > > >>>>> wrote: > >>>>>> > >>>>>> Here is a debug log from a call from an internal phone out to > an > >>>>>> external (my iPhone with nbr 91316356): > >>>>>> http://pastebin.freeswitch.org/108578 > >>>>>> > >>>>>> Ivan > >>>>>> > >>>>> Uh... you wanna try that PB number again? > >>>>> -MC > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > >>>>> freeswitch- > >>>>> users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > >>>>> freeswitch- > >>>>> users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>> -- > >>>> Eliot Gable > >>>> > >>>> "We do not inherit the Earth from our ancestors: we borrow it > from > >>>> our > >>>> children." ~David Brower > >>>> > >>>> "I decided the words were too conservative for me. We're not > >>>> borrowing > >>>> from our children, we're stealing from them--and it's not even > >>>> considered to be a crime." ~David Brower > >>>> > >>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat > to > >>>> live; not live to eat.) ~Marcus Tullius Cicero > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > freeswitch- > >>>> users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Eliot Gable > >> > >> "We do not inherit the Earth from our ancestors: we borrow it from > >> our > >> children." ~David Brower > >> > >> "I decided the words were too conservative for me. We're not > >> borrowing > >> from our children, we're stealing from them--and it's not even > >> considered to be a crime." ~David Brower > >> > >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > >> live; not live to eat.) ~Marcus Tullius Cicero > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/e9ab9ee5/attachment-0002.html From msc at freeswitch.org Mon Nov 2 13:04:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Nov 2009 13:04:12 -0800 Subject: [Freeswitch-users] SIP provider with extern rtp server In-Reply-To: <03E9A3AC-F501-4FAE-8199-A4AEC4D60891@myrvold.org> References: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com> <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org> <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org> <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org> <191c3a030911011220m22d7b515kda412e1fd9408f59@mail.gmail.com> <03E9A3AC-F501-4FAE-8199-A4AEC4D60891@myrvold.org> Message-ID: <87f2f3b90911021304yb972754gc4daa504d2304f92@mail.gmail.com> On Mon, Nov 2, 2009 at 12:58 PM, Ivan C Myrvold wrote: > That was it. My sip provider applied the patch to his Asterisk server that > was referenced in the link you was so kind to provide, and again everything > worked as it should. > > Thank you very much! > > This is why Tony's Asterisk karma is still so high! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/d8b1611d/attachment-0002.html From hjqlopez at hotmail.com Mon Nov 2 14:30:38 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Mon, 2 Nov 2009 17:30:38 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media Message-ID: Hi Mike, I re-tried with trunk rev 15319 but I got almost the same behavior: There is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But still there is no reINVITE for A (with C's SDP) after the call from FS to C is established. Anyway, we decided for now to do a different implementation but if you want to explore more in this issue count me in ;-) Thank you very much! Humberto >Please re-try with latest svn trunk. > >Mike > >On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote: > >> >> Thanks for you answers guys, >> >> I test the parameters you suggested >> but still no audio due to the lack of reINVITE. By the way I'm using >> 1.0.4 but I also tried 1.0.5pre3. >> >> One particular condition is that there is no on-hold before the >> Blind Transfer. >> >> Regards, >> >> Humberto >> >>> >>> _________________________________________________________________ Lots of fantastic Windows 7 offers, in one convenient place. Get the perfect deal for you now. http://go.microsoft.com/?linkid=9691633 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/e4cb94ad/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 2 16:01:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Nov 2009 18:01:33 -0600 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <191c3a030911021601l660c294dp16179dbc5e23206e@mail.gmail.com> please try r15326 I think i have it working. I recommend for optimal results you set bypass_media_after_bridge=true either as a global or in your DP in place of bypass_media=true On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana wrote: > Hi Mike, > > I re-tried with trunk rev 15319 but I got almost the same behavior: There > is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But > still there is no reINVITE for A (with C's SDP) after the call from FS to C > is established. > > Anyway, we decided for now to do a different implementation but if you want > to explore more in this issue count me in ;-) > > > Thank you very much! > > Humberto > > > > > >Please re-try with latest svn trunk. > > > >Mike > > > >On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote: > > > >> > >> Thanks for you answers guys, > >> > >> I test the parameters you suggested > >> but still no audio due to the lack of reINVITE. By the way I'm using > >> 1.0.4 but I also tried 1.0.5pre3. > >> > >> One particular condition is that there is no on-hold before the > >> Blind Transfer. > >> > >> Regards, > >> > >> Humberto > >> > >>> > >>> > > ------------------------------ > Lots of fantastic offers on Windows 7, in one convenient place. Get a deal > on Windows 7 now > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091102/ec9fffdb/attachment-0002.html From maciej.aniserowicz at gmail.com Mon Nov 2 21:48:45 2009 From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz) Date: Mon, 2 Nov 2009 21:48:45 -0800 (PST) Subject: [Freeswitch-users] "Can not record session. Media not enabled on channel." In-Reply-To: <1256746903296-3906568.post@n2.nabble.com> References: <1256034659005-3857858.post@n2.nabble.com> <87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com> <1256283391890-3877285.post@n2.nabble.com> <87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com> <1256541677550-3890610.post@n2.nabble.com> <87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com> <1256655334215-3899478.post@n2.nabble.com> <191c3a030910270818y79f436d9h8f6311c4671502c2@mail.gmail.com> <1256746903296-3906568.post@n2.nabble.com> Message-ID: <1257227325746-3936705.post@n2.nabble.com> Hi, Unfortunately getting the newest version did not solve the problem: "Can not record session. Media not enabled on channel." error still appears sometimes. MA Maciej Aniserowicz wrote: > > Correct - compiled but did not run. Works fine now. > > I'll see if the error shows up again and let you know if it does. > Thanks, > MA > > > > Anthony Minessale wrote: >> >> won't compile or won't run? >> maybe you should try rebuilding it. >> >> >> On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz < >> maciej.aniserowicz at gmail.com> wrote: >> >>> Sorry, trunk does not compile on win7, here are the details: >>> >>> >>> rev.15247 >>> >>> --------------------------- >>> Microsoft Visual C++ Debug Library >>> --------------------------- >>> Debug Assertion Failed! >>> >>> >>> >>> >>> >>> ----- Original Message ----- >>> *From:* [hidden >>> email] >>> *To:* [hidden >>> email] >>> *Sent:* Monday, October 26, 2009 10:32 PM >>> *Subject:* Re: [Freeswitch-users] "Can not record session. Media not >>> enabled on channel." >>> >>> >>> >>> On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden >>> email] >>> > wrote: >>> >>>> >>>> Yes, I can confirm - this exact error occurs each time when I start >>>> recording >>>> before the call is answered (just after sending ORIGINATE command) - >>>> but I >>>> think that's completely understandable that media is not ready for an >>>> unanswered call. >>>> But... is there any other event that guarantees media to be ready? >>>> >>>> Update to latest SVN and try again. >>> -MC >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> [hidden >>> email] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------ >>> View this message in context: Re: [Freeswitch-users] "Can not record >>> session. Media not enabled on >>> channel." >>> >>> Sent from the freeswitch-users mailing list >>> archiveat >>> Nabble.com. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3936705.html Sent from the freeswitch-users mailing list archive at Nabble.com. From lei.tlfly at gmail.com Tue Nov 3 01:05:24 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 3 Nov 2009 17:05:24 +0800 Subject: [Freeswitch-users] How to get digitals and stop play when speak tts? Just like session:playAndGetDigits do Message-ID: <50c41b4e0911030105k5585b8f2q3b355fbb488750ed@mail.gmail.com> Hi all, I'm writing lua ivr scirpt, Does some known how to get digitals and stop play when speak tts? Just like session:playAndGetDigits do. Thanks lots! Best Regards! -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/4ef9368f/attachment-0002.html From brian.stafford at lattice-voice.com Tue Nov 3 01:28:18 2009 From: brian.stafford at lattice-voice.com (Brian Stafford) Date: Tue, 03 Nov 2009 09:28:18 +0000 Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg of call In-Reply-To: <4AEB1166.80002@lattice-voice.com> References: <4AEB0A9C.7010907@lattice-voice.com> <4AEB1166.80002@lattice-voice.com> Message-ID: <4AEFF7B2.3080607@lattice-voice.com> Brian Stafford wrote: > Brian West wrote: > >> You have to be doing it wrong then. >> >> Can you show us your dialplan you should have two extensions one for >> the lot range and one to attended transfer someone into the lot. >> >> /b >> >> > The relevant excerpt from the dialplan is > > > > > > > > > > > > > > > > x410-419 are the slots and 420 parks a call. Parking by picking one of > 410-419 works fine and subsequently dialling them from another works > fine, I added x420 for the auto feature. > > Regards > Brian > > _ Any clues what I'm doing wrong? Is more information needed? Brian From maciej.aniserowicz at gmail.com Tue Nov 3 02:28:13 2009 From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz) Date: Tue, 3 Nov 2009 02:28:13 -0800 (PST) Subject: [Freeswitch-users] Users hanged up for unknown reason Message-ID: <1257244093831-3937601.post@n2.nabble.com> Hi, I have a strange problem. I control FS with commands sent by tcp in response to events published via tcp. I do something like: 1) call 1st user 2) call 2nd user 3) 1st and 2nd talk 4) call another user 5) 1st and another talk etc... Sometimes (quite regularly) users are hanged up (with cause NORMAL_CLEARING) even if they do not hangup manually. I pasted one such scenario in pastebin (http://pastebin.freeswitch.org/10955), it includes logs from commands sent by me and events received from FS. Could someone take a look and see what am I doing wrong? The scenario includes 3 users 1st user (Unique-ID: f076261a-4537-40f2-b46d-933141320314) is supposed to be connected all the time but gets diconnected 2nd user (Unique-ID: ebdfb398-ec82-4760-9f79-81364e0f37b6) is supposed to talk for a few seconds and get killed 3rd user (Unique-ID: d5cd839e-793c-4b3c-adda-327841672a5f) is supposed to work like 2nd user All of them are simulated by dialplan extensions (using answer and playback tools), but the same thing happends for xlite or cisco phone. Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Users-hanged-up-for-unknown-reason-tp3937601p3937601.html Sent from the freeswitch-users mailing list archive at Nabble.com. From stevendt at primrosebank.net Tue Nov 3 03:28:23 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 11:28:23 -0000 Subject: [Freeswitch-users] Precompiled Windows Binaries Message-ID: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> Hi, I have read the Docs on the Wiki (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) but am still not sure of what the different Windows install files are. Currently, the Windows Installer directory contains :- LATEST_SVN_15106 - 6 Bytes freeswitch-1.0.4.exe - 42 Megabytes freeswitch.exe - 32 Megabytes I have installed the freeswitch-1.0.4.exe file which is dated 3rd September. The freeswitch.exe file is dated 7th October and think that it contains the minor updates since 3rd September ? Could someone who knows FreeSwitch under windows help me understand the two files please ? I chickened out of running the later exe in case it did something to the running install of FreeSwitch 1.0.4, is it safe to run the newer exe with the old one already installed ? What will it actually do ? regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/8518c608/attachment-0002.html From jlenk at frontiernet.net Tue Nov 3 06:48:34 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 3 Nov 2009 06:48:34 -0800 (PST) Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> Message-ID: <1257259714704-3938887.post@n2.nabble.com> Hi Dave, These are supported by "Carlos Talbot" . They also include Freepbx v3 Just as you said freeswitch-1.0.4.exe is the tagged release and freeswitch.exe is a newer svn snapshot. There should be no problems installing the new version allthough best to just try and see! Not sure why the newest one is from October 7th. Jeff Dave Stevenson wrote: > > Hi, > > I have read the Docs on the Wiki > (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) > but am still not sure of what the different Windows install files are. > Currently, the Windows Installer directory contains :- > > LATEST_SVN_15106 - 6 Bytes > > freeswitch-1.0.4.exe - 42 Megabytes > > freeswitch.exe - 32 Megabytes > > I have installed the freeswitch-1.0.4.exe file which is dated 3rd > September. The freeswitch.exe file is dated 7th October and think that it > contains the minor updates since 3rd September ? > > Could someone who knows FreeSwitch under windows help me understand the > two files please ? > > I chickened out of running the later exe in case it did something to the > running install of FreeSwitch 1.0.4, is it safe to run the newer exe with > the old one already installed ? > What will it actually do ? > > regards > Dave > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html Sent from the freeswitch-users mailing list archive at Nabble.com. From stevendt at primrosebank.net Tue Nov 3 07:27:28 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 15:27:28 -0000 Subject: [Freeswitch-users] Precompiled Windows Binaries References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> Message-ID: Jeff, thanks a lot for the reply. I was a little confused by the fact that the "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried that I might lose something. As you say though, think that I'll cross my fingers and try the updated release. I am running FreeSwitch on a test machine at the moment until the target hardware arrives - hopefully tomorrow, so I can afford to have a little play. You mentioned FreePBX V3. I had been fumbling around trying to work out what this is and from what I've read, it seems to provide a GUI Front End for configuring FreeSwitch ? I am guessing that while it has been installed with FreeSwitch, I then need to run the FreePBX Installer to update the FreePBX/FreeSwitch configuration on my hardware ? When I start FreeSwitch, it does not automatically load the WAMPServer. When I start WAMPServer manually, and open up localhost (127.0.0.1) in a web browser, I can see the WampServer logo and various tools such as phpinfo() and phpmyadmin. FreePBX is there under Your Projects. When I opened this up the first time, it appeared to want to install FreePBX over FreeSwitch, I tried to abort this when it was going to overwrite some FreeSwitch conf files and I thought I'd better not go on until I had a better idea what was happening. I backed out of the FreePBX install and now I can't get the FreePBX or phpmyadmin pages up again (missing files) so it looks like I'm going to have to reinstall anyway. So, for next time,am I right in thinking that I should proceed with running the FreePBX install from the WAMPServer menu ? regards Dave ----- Original Message ----- From: "Jeff Lenk" To: Sent: Tuesday, November 03, 2009 2:48 PM Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > Hi Dave, > > These are supported by "Carlos Talbot" . They also include Freepbx v3 > > Just as you said freeswitch-1.0.4.exe is the tagged release and > freeswitch.exe is a newer svn snapshot. > > There should be no problems installing the new version allthough best to > just try and see! > > Not sure why the newest one is from October 7th. > > Jeff > > > Dave Stevenson wrote: >> >> Hi, >> >> I have read the Docs on the Wiki >> (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> but am still not sure of what the different Windows install files are. >> Currently, the Windows Installer directory contains :- >> >> LATEST_SVN_15106 - 6 Bytes >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> freeswitch.exe - 32 Megabytes >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> September. The freeswitch.exe file is dated 7th October and think that it >> contains the minor updates since 3rd September ? >> >> Could someone who knows FreeSwitch under windows help me understand the >> two files please ? >> >> I chickened out of running the later exe in case it did something to the >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe with >> the old one already installed ? >> What will it actually do ? >> >> regards >> Dave >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From odermann at googlemail.com Tue Nov 3 07:57:24 2009 From: odermann at googlemail.com (Dennis) Date: Tue, 3 Nov 2009 16:57:24 +0100 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com> <5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com> <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com> <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> Message-ID: <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> hi anthony, i believe, that there is no problem with the communication between fs and the cirpack (everything works to smooth as if this could be possible). if fs sends the 484, the cirpack sends more digits to fs (if there are some), so this works as it should. the problem is, that fs ends the session/socket after a 484, so that the cirpack sends the following digits into another socket. you wrote about a "1 line patch", which might not have been implemented - at least it seems so. is there a way to get someone of the sofia devs to fix this small problem, so that fs sends the 484 without ending the session/socket and waiting for an answer of the cirpack? we would take care of the rest. kind regards, dennis 2009/10/15 Anthony Minessale : > right you can reply 484 in your dp at any time > > > then it should try again. > > The bit i can't remember is if we committed a certain 1 line patch that > makes sofia parse the next invite to the same call properly, the patch was > to the sofia lib itself so test it and see. I may need to dig up the answer > again from the sofia dev. From msc at freeswitch.org Tue Nov 3 08:38:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Nov 2009 08:38:41 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.0.5 Scheduled For November 10; 1.0.5pre5 Now Available Message-ID: <87f2f3b90911030838h505a47b7qd5333fab525fc65b@mail.gmail.com> Greetings! The latest FreeSWITCH pre-release is now available: http://www.freeswitch.org/node/215 Please update and test as soon as possible. With the community's help we should be able to hit our target of releasing version 1.0.5 next Tuesday November 10. The FreeSWITCH developers appreciate all the hard work that the community does on behalf of the project. Like most open source projects, FreeSWITCH needs the community to "give back" a little and you all certainly do that. Please keep up the great work. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/da32af15/attachment-0002.html From frank at carmickle.com Tue Nov 3 09:01:10 2009 From: frank at carmickle.com (Frank Carmickle) Date: Tue, 3 Nov 2009 12:01:10 -0500 Subject: [Freeswitch-users] portaudio error Message-ID: <20091103170110.GK10757@base.carmickle.com> Hello Debian lenny with svn15321 freeswitch at internal> load mod_portaudio -ERR [module load file routine returned an error] 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input devicefreeswitch at internal> 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 Cannot find an input device 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_portaudio.so **Module load routine returned an error** Any help would be appreciated. --FC From anthony.minessale at gmail.com Tue Nov 3 09:19:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 11:19:51 -0600 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com> <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com> <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> Message-ID: <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> The patch was it's ability to accept subsequent invites. Your problem is that in sip each new attempt to send an invite is another call. 484 is a final response so the call with too few digits is terminated. On Tue, Nov 3, 2009 at 9:57 AM, Dennis wrote: > hi anthony, > > i believe, that there is no problem with the communication between fs > and the cirpack (everything works to smooth as if this could be > possible). if fs sends the 484, the cirpack sends more digits to fs > (if there are some), so this works as it should. the problem is, that > fs ends the session/socket after a 484, so that the cirpack sends the > following digits into another socket. > > you wrote about a "1 line patch", which might not have been > implemented - at least it seems so. > > is there a way to get someone of the sofia devs to fix this small > problem, so that fs sends the 484 without ending the session/socket > and waiting for an answer of the cirpack? we would take care of the > rest. > > kind regards, > dennis > > > 2009/10/15 Anthony Minessale : > > right you can reply 484 in your dp at any time > > > > > > then it should try again. > > > > The bit i can't remember is if we committed a certain 1 line patch that > > makes sofia parse the next invite to the same call properly, the patch > was > > to the sofia lib itself so test it and see. I may need to dig up the > answer > > again from the sofia dev. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/21a896c6/attachment-0002.html From andrew at hijacked.us Tue Nov 3 09:24:38 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 3 Nov 2009 12:24:38 -0500 Subject: [Freeswitch-users] portaudio error In-Reply-To: <20091103170110.GK10757@base.carmickle.com> References: <20091103170110.GK10757@base.carmickle.com> Message-ID: <20091103172437.GA9418@hijacked.us> On Tue, Nov 03, 2009 at 12:01:10PM -0500, Frank Carmickle wrote: > Hello > > Debian lenny with svn15321 > > freeswitch at internal> load mod_portaudio > -ERR [module load file routine returned an error] > > 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input devicefreeswitch at internal> 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 Cannot find an input device > 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_portaudio.so > **Module load routine returned an error** > Try installing the alsa development headers, it's got some stupid name on debian like libasound2-devel or something. Then re-build the portaudio module and library (a couple well placed make cleans should do it). Andrew From stevendt at primrosebank.net Tue Nov 3 09:30:08 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 17:30:08 -0000 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch Message-ID: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension & password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/306a4add/attachment-0002.html From mariano.dellano at gmail.com Tue Nov 3 09:47:05 2009 From: mariano.dellano at gmail.com (Mariano de Llano) Date: Tue, 3 Nov 2009 14:47:05 -0300 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <24282895.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> Message-ID: Hi, I'm having a problem with a Sipura, it is sending for the G729 the tag "G729a" witch is not correct due the RFC. Media Attribute (a): rtpmap:18 G729a/8000 FS is returning (200OK) Media Attribute (a): rtpmap:96 G729/8000 I think that the problem is that FS is not matching the codec, so it returns the first dynamic payload which is 96. I think that I've seen post with a similar issue, and the solution was to change the tag before it hit the switch, so, what I've done is to change the "switch_r_sdp" (I have the rest of the parameters correct due I also use it to dynamically change the codecs order) and it's changing the SDP, but when FS sends the 200OK it is returning to the endpoint: Media Attribute (a): rtpmap:96 G729/8000 Which is exactly the same problem that I have without the transformation of the SDP. Is it correct? Do I have another solution? Thanks From anthony.minessale at gmail.com Tue Nov 3 09:55:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 11:55:21 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: References: <24251951.post@talk.nabble.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> Message-ID: <191c3a030911030955k6406e81byf96f18250eb3b835@mail.gmail.com> I think you can edit the prefs in your sipura and change it to the correct string. On Tue, Nov 3, 2009 at 11:47 AM, Mariano de Llano wrote: > Hi, > > I'm having a problem with a Sipura, it is sending for the G729 the > tag "G729a" witch is not correct due the RFC. > > Media Attribute (a): rtpmap:18 G729a/8000 > > FS is returning (200OK) > > Media Attribute (a): rtpmap:96 G729/8000 > > I think that the problem is that FS is not matching the codec, so it > returns the first dynamic payload which is 96. > > I think that I've seen post with a similar issue, and the solution was > to change the tag before it hit the switch, so, what I've done is to > change the "switch_r_sdp" (I have the rest of the parameters correct > due I also use it to dynamically change the codecs order) and it's > changing the SDP, but when FS sends the 200OK it is returning to the > endpoint: > > Media Attribute (a): rtpmap:96 G729/8000 > > Which is exactly the same problem that I have without the > transformation of the SDP. > > Is it correct? Do I have another solution? > > Thanks > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/10f301eb/attachment-0002.html From brian at freeswitch.org Tue Nov 3 09:58:48 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 11:58:48 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> Message-ID: <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> FIx your sipura to NOT include the a in the codec its in the admin section of the UI on the ATA. /b On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote: > Hi, > > I'm having a problem with a Sipura, it is sending for the G729 the > tag "G729a" witch is not correct due the RFC. > > Media Attribute (a): rtpmap:18 G729a/8000 > > FS is returning (200OK) > > Media Attribute (a): rtpmap:96 G729/8000 > > I think that the problem is that FS is not matching the codec, so it > returns the first dynamic payload which is 96. > > I think that I've seen post with a similar issue, and the solution was > to change the tag before it hit the switch, so, what I've done is to > change the "switch_r_sdp" (I have the rest of the parameters correct > due I also use it to dynamically change the codecs order) and it's > changing the SDP, but when FS sends the 200OK it is returning to the > endpoint: > > Media Attribute (a): rtpmap:96 G729/8000 > > Which is exactly the same problem that I have without the > transformation of the SDP. > > Is it correct? Do I have another solution? > > Thanks > From mariano.dellano at gmail.com Tue Nov 3 10:11:13 2009 From: mariano.dellano at gmail.com (Mariano de Llano) Date: Tue, 3 Nov 2009 15:11:13 -0300 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> Message-ID: <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Yes, that was my first option, but there many endpoints that I'm not able to configure. Basically it's a broadband solution where I have like 1000 endpoints that are out of my provisioning. Thanks, M On 03/11/2009, at 14:58, Brian West wrote: > FIx your sipura to NOT include the a in the codec its in the admin > section of the UI on the ATA. > > /b > > On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote: > >> Hi, >> >> I'm having a problem with a Sipura, it is sending for the G729 the >> tag "G729a" witch is not correct due the RFC. >> >> Media Attribute (a): rtpmap:18 G729a/8000 >> >> FS is returning (200OK) >> >> Media Attribute (a): rtpmap:96 G729/8000 >> >> I think that the problem is that FS is not matching the codec, so it >> returns the first dynamic payload which is 96. >> >> I think that I've seen post with a similar issue, and the solution >> was >> to change the tag before it hit the switch, so, what I've done is to >> change the "switch_r_sdp" (I have the rest of the parameters correct >> due I also use it to dynamically change the codecs order) and it's >> changing the SDP, but when FS sends the 200OK it is returning to the >> endpoint: >> >> Media Attribute (a): rtpmap:96 G729/8000 >> >> Which is exactly the same problem that I have without the >> transformation of the SDP. >> >> Is it correct? Do I have another solution? >> >> Thanks >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Nov 3 10:24:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 12:24:47 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> References: <24251951.post@talk.nabble.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Message-ID: <191c3a030911031024n1bf258abldbbb24f35f107f3b@mail.gmail.com> so imagine how much money all those sipuras cost. They get all the money *and* have a bug and we are free and are supposed to break the rules for them. On Tue, Nov 3, 2009 at 12:11 PM, Mariano de Llano wrote: > Yes, that was my first option, but there many endpoints that I'm not > able to configure. Basically it's a broadband solution where I have > like 1000 endpoints that are out of my provisioning. > > Thanks, > M > > On 03/11/2009, at 14:58, Brian West wrote: > > > FIx your sipura to NOT include the a in the codec its in the admin > > section of the UI on the ATA. > > > > /b > > > > On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote: > > > >> Hi, > >> > >> I'm having a problem with a Sipura, it is sending for the G729 the > >> tag "G729a" witch is not correct due the RFC. > >> > >> Media Attribute (a): rtpmap:18 G729a/8000 > >> > >> FS is returning (200OK) > >> > >> Media Attribute (a): rtpmap:96 G729/8000 > >> > >> I think that the problem is that FS is not matching the codec, so it > >> returns the first dynamic payload which is 96. > >> > >> I think that I've seen post with a similar issue, and the solution > >> was > >> to change the tag before it hit the switch, so, what I've done is to > >> change the "switch_r_sdp" (I have the rest of the parameters correct > >> due I also use it to dynamically change the codecs order) and it's > >> changing the SDP, but when FS sends the 200OK it is returning to the > >> endpoint: > >> > >> Media Attribute (a): rtpmap:96 G729/8000 > >> > >> Which is exactly the same problem that I have without the > >> transformation of the SDP. > >> > >> Is it correct? Do I have another solution? > >> > >> Thanks > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/def75bf7/attachment-0002.html From brian at freeswitch.org Tue Nov 3 10:27:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 12:27:39 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Message-ID: Sounds like bad planning. I would send out a memo to your users and have them fix it. I have raised a bug multiple times with Cisco g729a is NOT valid. /b On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: > Yes, that was my first option, but there many endpoints that I'm not > able to configure. Basically it's a broadband solution where I have > like 1000 endpoints that are out of my provisioning. > > Thanks, > M From anthony.minessale at gmail.com Tue Nov 3 10:45:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 12:45:53 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: References: <24251951.post@talk.nabble.com> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Message-ID: <191c3a030911031045ue8f9375ve0e76953d07d595d@mail.gmail.com> I am willing to support this with the note that its incorrect and will not support it by default but update to trunk and try: this should fix it for you, SIGH On Tue, Nov 3, 2009 at 12:27 PM, Brian West wrote: > Sounds like bad planning. I would send out a memo to your users and > have them fix it. I have raised a bug multiple times with Cisco g729a > is NOT valid. > > /b > > On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: > > > Yes, that was my first option, but there many endpoints that I'm not > > able to configure. Basically it's a broadband solution where I have > > like 1000 endpoints that are out of my provisioning. > > > > Thanks, > > M > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/751c5579/attachment-0002.html From kristian.kielhofner at gmail.com Tue Nov 3 11:11:10 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 3 Nov 2009 14:11:10 -0500 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> References: <24251951.post@talk.nabble.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Message-ID: <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> It appears that Tony has already added an option (amazing) BUT you should really be setup for central provisioning with an installed base that large... You'll eventually have issues that *NO* amount of Tony/FreeSWITCH magic can fix. On Tue, Nov 3, 2009 at 1:11 PM, Mariano de Llano wrote: > Yes, that was my first option, but there many endpoints that I'm not > able to configure. Basically it's a broadband solution where I have > like 1000 endpoints that are out of my provisioning. > > Thanks, > M > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From achaloyan at yahoo.com Tue Nov 3 11:12:52 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Tue, 3 Nov 2009 11:12:52 -0800 (PST) Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> Message-ID: <32896.60654.qm@web111308.mail.gq1.yahoo.com> Actually, there were a few more misinterpretations in earlier software of Cisco Gateways, which RFC implementers had to address in RFC3551, strange ... RTP Payload Type 19 remains reserved because "some implementations" wrongly interpreted 13 decimal as 13 hexadecimal value. Another issue is G726 bit packing. Again "some implementations" used wrong bit packing and RFC3551 tried to partially resolve this conflict introducing new payload format named AAL2-G726 ... ________________________________ From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Tue, November 3, 2009 10:27:39 PM Subject: Re: [Freeswitch-users] Sipura Codec Problem Sounds like bad planning. I would send out a memo to your users and have them fix it. I have raised a bug multiple times with Cisco g729a is NOT valid. /b On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: > Yes, that was my first option, but there many endpoints that I'm not > able to configure. Basically it's a broadband solution where I have > like 1000 endpoints that are out of my provisioning. > > Thanks, > M _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/a5ffdf9d/attachment-0002.html From anthony.minessale at gmail.com Tue Nov 3 11:21:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 13:21:57 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <32896.60654.qm@web111308.mail.gq1.yahoo.com> References: <24251951.post@talk.nabble.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <32896.60654.qm@web111308.mail.gq1.yahoo.com> Message-ID: <191c3a030911031121n714daa3dj834b77e01d13638a@mail.gmail.com> Don't forget the one where there was a typo in the one for G722 so now we are all required to emulate that typo by running a 16khz codec with 8khz timestamps and sdp params. On Tue, Nov 3, 2009 at 1:12 PM, Arsen Chaloyan wrote: > Actually, there were a few more misinterpretations in earlier software of > Cisco Gateways, which RFC implementers had to address in RFC3551, strange > ... > > RTP Payload Type 19 remains reserved because "some implementations" wrongly > interpreted 13 decimal as 13 hexadecimal value. > Another issue is G726 bit packing. Again "some implementations" used wrong > bit packing and RFC3551 tried to partially resolve this conflict introducing > new payload format named AAL2-G726 ... > > ------------------------------ > *From:* Brian West > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Tue, November 3, 2009 10:27:39 PM > *Subject:* Re: [Freeswitch-users] Sipura Codec Problem > > Sounds like bad planning. I would send out a memo to your users and > have them fix it. I have raised a bug multiple times with Cisco g729a > is NOT valid. > > /b > > On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: > > > Yes, that was my first option, but there many endpoints that I'm not > > able to configure. Basically it's a broadband solution where I have > > like 1000 endpoints that are out of my provisioning. > > > > Thanks, > > M > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/20a59753/attachment-0002.html From brian at freeswitch.org Tue Nov 3 11:22:50 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 13:22:50 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> References: <24251951.post@talk.nabble.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> Message-ID: At some point the paint will be rubbed off the magic lamp. /b On Nov 3, 2009, at 1:11 PM, Kristian Kielhofner wrote: > It appears that Tony has already added an option (amazing) BUT you > should really be setup for central provisioning with an installed base > that large... You'll eventually have issues that *NO* amount of > Tony/FreeSWITCH magic can fix. From brian at freeswitch.org Tue Nov 3 11:24:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 13:24:06 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <32896.60654.qm@web111308.mail.gq1.yahoo.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <32896.60654.qm@web111308.mail.gq1.yahoo.com> Message-ID: <2FEB08A4-FB22-4CC4-9BBC-E3732756B7DB@freeswitch.org> Yah this one is LLLAME!!!! :P We have some dyslexic engineers. /b On Nov 3, 2009, at 1:12 PM, Arsen Chaloyan wrote: > Another issue is G726 bit packing. Again "some implementations" used > wrong bit packing and RFC3551 tried to partially resolve this > conflict introducing new payload format named AAL2-G726 ... From brian at freeswitch.org Tue Nov 3 11:25:42 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 13:25:42 -0600 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <191c3a030911031121n714daa3dj834b77e01d13638a@mail.gmail.com> References: <24251951.post@talk.nabble.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <32896.60654.qm@web111308.mail.gq1.yahoo.com> <191c3a030911031121n714daa3dj834b77e01d13638a@mail.gmail.com> Message-ID: THE BLOODY MADNESS!!! I can only stop if people start saying 'NO'. :) /b On Nov 3, 2009, at 1:21 PM, Anthony Minessale wrote: > Don't forget the one where there was a typo in the one for G722 so > now we are all required to emulate that typo by running a 16khz > codec with 8khz timestamps and sdp params. From jlenk at frontiernet.net Tue Nov 3 11:26:16 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 3 Nov 2009 11:26:16 -0800 (PST) Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> Message-ID: <1257276376879-3940700.post@n2.nabble.com> Dave, Carlos can probably be a better help here too but yes Freepbx v3 is a web gui that is under heavy development - it probably is not yet ready for production but looks very promising! you can navigate to http://127.0.0.1/freepbx-v3/index.php/installer.html and restart the installer for freepbx if you want to experiment with it. The base FreeSWITCH installer does install and work well with windows and is quite easy to learn and configure. Their is a lot to learn though :) Regards, Jeff Dave Stevenson wrote: > > Jeff, > > thanks a lot for the reply. I was a little confused by the fact that the > "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried > that I might lose something. As you say though, think that I'll cross my > fingers and try the updated release. I am running FreeSwitch on a test > machine at the moment until the target hardware arrives - hopefully > tomorrow, so I can afford to have a little play. > > You mentioned FreePBX V3. I had been fumbling around trying to work out > what > this is and from what I've read, it seems to provide a GUI Front End for > configuring FreeSwitch ? > > I am guessing that while it has been installed with FreeSwitch, I then > need > to run the FreePBX Installer to update the FreePBX/FreeSwitch > configuration > on my hardware ? > > > When I start FreeSwitch, it does not automatically load the WAMPServer. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a > web > browser, I can see the WampServer logo and various tools such as phpinfo() > and phpmyadmin. FreePBX is there under Your Projects. > > When I opened this up the first time, it appeared to want to install > FreePBX > over FreeSwitch, I tried to abort this when it was going to overwrite some > FreeSwitch conf files and I thought I'd better not go on until I had a > better idea what was happening. I backed out of the FreePBX install and > now > I can't get the FreePBX or phpmyadmin pages up again (missing files) so it > looks like I'm going to have to reinstall anyway. > > So, for next time,am I right in thinking that I should proceed with > running > the FreePBX install from the WAMPServer menu ? > > regards > Dave > > > > ----- Original Message ----- > From: "Jeff Lenk" > To: > Sent: Tuesday, November 03, 2009 2:48 PM > Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > >> >> Hi Dave, >> >> These are supported by "Carlos Talbot" . They also include Freepbx v3 >> >> Just as you said freeswitch-1.0.4.exe is the tagged release and >> freeswitch.exe is a newer svn snapshot. >> >> There should be no problems installing the new version allthough best to >> just try and see! >> >> Not sure why the newest one is from October 7th. >> >> Jeff >> >> >> Dave Stevenson wrote: >>> >>> Hi, >>> >>> I have read the Docs on the Wiki >>> (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >>> but am still not sure of what the different Windows install files are. >>> Currently, the Windows Installer directory contains :- >>> >>> LATEST_SVN_15106 - 6 Bytes >>> >>> freeswitch-1.0.4.exe - 42 Megabytes >>> >>> freeswitch.exe - 32 Megabytes >>> >>> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >>> September. The freeswitch.exe file is dated 7th October and think that >>> it >>> contains the minor updates since 3rd September ? >>> >>> Could someone who knows FreeSwitch under windows help me understand the >>> two files please ? >>> >>> I chickened out of running the later exe in case it did something to the >>> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >>> with >>> the old one already installed ? >>> What will it actually do ? >>> >>> regards >>> Dave >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3940700.html Sent from the freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Tue Nov 3 11:53:26 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 3 Nov 2009 20:53:26 +0100 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch In-Reply-To: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> Message-ID: <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote: > Help please . . . . > > Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? > > I have got FreeSwitch up and running with the SoftPhone, but can't get a > 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP > Address from DHCP and it can see the FreeSwitch server but I can't find > anything in the phone to allow the extension & password to be configured. > Can FreeSwitch send this data to the phone (and if so, which configuration > files are involved) or must the phone be configured manually before it can > talk to FreeSwitch ? > > Any help would be really appreciated as I'm pulling my hair out here ! > > Regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/d428fa04/attachment-0002.html From stevendt at primrosebank.net Tue Nov 3 12:03:13 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 20:03:13 -0000 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> Message-ID: Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. >From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a "special" version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave ----- Original Message ----- From: Tihomir Culjaga To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 03, 2009 7:53 PM Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension & password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/5388a9bd/attachment-0002.html From chris.chen2004 at gmail.com Tue Nov 3 12:18:24 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Tue, 3 Nov 2009 15:18:24 -0500 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch In-Reply-To: References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> Message-ID: <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> I think you are most likely on the wrong track, 3COM phones are locked to either 3COM PBX or the special Asterisk edition locked-down by 3COM. You cannot make them work with either FreeSWITCH or any other open SIP server other than 3COM IP PBX systems. I learned this over one year ago by playing with 3COm 3102 phones myself. Chris On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson wrote: > Tihomir, > > thanks for the link, but actually, I had already found/downloaded/read and > almost understood that document ! > > However, the options to log into the phone and configure the extension > number etc. do not appear on my phone. > > From reading another post on the web, I don't think that the phone has the > SIP software loaded until it is downloaded from the Server - I think that > there is a "special" version of Asterix for 3Com that does this, maybe the > same functionality does not exist in FreeSwitch ? > > Maybe I should have been clearer in the post below, but I think that this > is the root of the problem. I think that the 3Com phone is looking for the > Switch to download the SIP firmware to it and FreeSwitch does not seem to do > that. > > Given that you have pointed me in the direction of that document, are you > using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, > but please let me know how you've made it work > > regards > Dave > > > > > ----- Original Message ----- > *From:* Tihomir Culjaga > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Tuesday, November 03, 2009 7:53 PM > *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with > FreeSwitch > > you might read this before you bigin :P > > http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf > > > T. > > > On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote: > >> Help please . . . . >> >> Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? >> >> I have got FreeSwitch up and running with the SoftPhone, but can't get a >> 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP >> Address from DHCP and it can see the FreeSwitch server but I can't find >> anything in the phone to allow the extension & password to be configured. >> Can FreeSwitch send this data to the phone (and if so, which configuration >> files are involved) or must the phone be configured manually before it can >> talk to FreeSwitch ? >> >> Any help would be really appreciated as I'm pulling my hair out here ! >> >> Regards >> Dave >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/b861e63f/attachment-0002.html From anthony.minessale at gmail.com Tue Nov 3 12:23:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 14:23:05 -0600 Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg of call In-Reply-To: <4AEFF7B2.3080607@lattice-voice.com> References: <4AEB0A9C.7010907@lattice-voice.com> <4AEB1166.80002@lattice-voice.com> <4AEFF7B2.3080607@lattice-voice.com> Message-ID: <191c3a030911031223p23835d6ev4c3c3ddd98193f50@mail.gmail.com> There are 2 ways to use the auto in one is to attended transfer the call into the extension with auto in the other is to bind_meta_app a call to valet_park + auto in blind transfer to auto in only has one leg so the guy you transferred is the only one who can hear it because when you press the blind xfer key you hangup the call on your side. On Tue, Nov 3, 2009 at 3:28 AM, Brian Stafford < brian.stafford at lattice-voice.com> wrote: > Brian Stafford wrote: > > Brian West wrote: > > > >> You have to be doing it wrong then. > >> > >> Can you show us your dialplan you should have two extensions one for > >> the lot range and one to attended transfer someone into the lot. > >> > >> /b > >> > >> > > The relevant excerpt from the dialplan is > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > x410-419 are the slots and 420 parks a call. Parking by picking one of > > 410-419 works fine and subsequently dialling them from another works > > fine, I added x420 for the auto feature. > > > > Regards > > Brian > > > > _ > > Any clues what I'm doing wrong? Is more information needed? > > Brian > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/53b92400/attachment-0002.html From stevendt at primrosebank.net Tue Nov 3 12:25:20 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 20:25:20 -0000 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com><65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> Message-ID: <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com> Chris, thanks a lot for the response. It's not the answer that I wanted, but it is what I was coming round to thinking. As much as I'm disappointed (particularly as I've just got the phone), but at least it's a definitive answer and I can avoid wasting any more time with it, so thanks again. Oh well, off to try and find some open SIP phones that will actually work for me, regards Dave ----- Original Message ----- From: Chris Chen To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 03, 2009 8:18 PM Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch I think you are most likely on the wrong track, 3COM phones are locked to either 3COM PBX or the special Asterisk edition locked-down by 3COM. You cannot make them work with either FreeSWITCH or any other open SIP server other than 3COM IP PBX systems. I learned this over one year ago by playing with 3COm 3102 phones myself. Chris On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson wrote: Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a "special" version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave ----- Original Message ----- From: Tihomir Culjaga To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 03, 2009 7:53 PM Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension & password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/55003312/attachment-0002.html From mastermind202 at gmail.com Tue Nov 3 09:52:09 2009 From: mastermind202 at gmail.com (mm_202) Date: Tue, 3 Nov 2009 12:52:09 -0500 Subject: [Freeswitch-users] FS and Skinny (SCCP) Message-ID: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> FS doesnt support SCCP (from what I gathered, just because no one has bothered coding it). Are there other users out there has use SCCP and FS? (with some middleware in between) If enough people would find a use for it, I'd be willing to actually code it (esp if someone offered a bounty). So, would anyone besides me want/use a SCCP endpoint in FS? -- mm_202. From tculjaga at gmail.com Tue Nov 3 12:53:42 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 3 Nov 2009 21:53:42 +0100 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch In-Reply-To: References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> Message-ID: <65d96fc80911031253j3ea587b3j48a71a5e2746ee23@mail.gmail.com> well, if it is a sip phone than you should be able to input your username&password somewhere. Usually, SIP phones downloads their configuration using dhcp/tftp|http method... the FW is downloaded just once if you need to upgrade the phone... I don't have any of these phones on my desk, just found the manual on the web. anyhow, freeswitch is expecting a SIP phone to register and thats it :P ... there is no specific phone provisioning from FS side. T. On Tue, Nov 3, 2009 at 9:03 PM, Dave Stevenson wrote: > Tihomir, > > thanks for the link, but actually, I had already found/downloaded/read and > almost understood that document ! > > However, the options to log into the phone and configure the extension > number etc. do not appear on my phone. > > From reading another post on the web, I don't think that the phone has the > SIP software loaded until it is downloaded from the Server - I think that > there is a "special" version of Asterix for 3Com that does this, maybe the > same functionality does not exist in FreeSwitch ? > > Maybe I should have been clearer in the post below, but I think that this > is the root of the problem. I think that the 3Com phone is looking for the > Switch to download the SIP firmware to it and FreeSwitch does not seem to do > that. > > Given that you have pointed me in the direction of that document, are you > using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, > but please let me know how you've made it work > > regards > Dave > > > > > ----- Original Message ----- > *From:* Tihomir Culjaga > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Tuesday, November 03, 2009 7:53 PM > *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with > FreeSwitch > > you might read this before you bigin :P > > http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf > > > T. > > > On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote: > >> Help please . . . . >> >> Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? >> >> I have got FreeSwitch up and running with the SoftPhone, but can't get a >> 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP >> Address from DHCP and it can see the FreeSwitch server but I can't find >> anything in the phone to allow the extension & password to be configured. >> Can FreeSwitch send this data to the phone (and if so, which configuration >> files are involved) or must the phone be configured manually before it can >> talk to FreeSwitch ? >> >> Any help would be really appreciated as I'm pulling my hair out here ! >> >> Regards >> Dave >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/f9f31ef1/attachment-0002.html From tculjaga at gmail.com Tue Nov 3 12:55:28 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 3 Nov 2009 21:55:28 +0100 Subject: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch In-Reply-To: <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> Message-ID: <65d96fc80911031255o3e7b7790ta61f3cb29107d786@mail.gmail.com> pity,the phone looks quite nice... On Tue, Nov 3, 2009 at 9:18 PM, Chris Chen wrote: > I think you are most likely on the wrong track, 3COM phones are locked to > either 3COM PBX or the special Asterisk edition locked-down by 3COM. You > cannot make them work with either FreeSWITCH or any other open SIP server > other than 3COM IP PBX systems. > I learned this over one year ago by playing with 3COm 3102 phones myself. > > Chris > > > > On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson wrote: > >> Tihomir, >> >> thanks for the link, but actually, I had already found/downloaded/read and >> almost understood that document ! >> >> However, the options to log into the phone and configure the extension >> number etc. do not appear on my phone. >> >> From reading another post on the web, I don't think that the phone has the >> SIP software loaded until it is downloaded from the Server - I think that >> there is a "special" version of Asterix for 3Com that does this, maybe the >> same functionality does not exist in FreeSwitch ? >> >> Maybe I should have been clearer in the post below, but I think that this >> is the root of the problem. I think that the 3Com phone is looking for >> the Switch to download the SIP firmware to it and FreeSwitch does not seem >> to do that. >> >> Given that you have pointed me in the direction of that document, are you >> using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, >> but please let me know how you've made it work >> >> regards >> Dave >> >> >> >> >> ----- Original Message ----- >> *From:* Tihomir Culjaga >> *To:* freeswitch-users at lists.freeswitch.org >> *Sent:* Tuesday, November 03, 2009 7:53 PM >> *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with >> FreeSwitch >> >> you might read this before you bigin :P >> >> http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf >> >> >> T. >> >> >> On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson > > wrote: >> >>> Help please . . . . >>> >>> Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? >>> >>> I have got FreeSwitch up and running with the SoftPhone, but can't get a >>> 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP >>> Address from DHCP and it can see the FreeSwitch server but I can't find >>> anything in the phone to allow the extension & password to be configured. >>> Can FreeSwitch send this data to the phone (and if so, which configuration >>> files are involved) or must the phone be configured manually before it can >>> talk to FreeSwitch ? >>> >>> Any help would be really appreciated as I'm pulling my hair out here ! >>> >>> Regards >>> Dave >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/789a3427/attachment-0002.html From stevendt at primrosebank.net Tue Nov 3 12:57:12 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 20:57:12 -0000 Subject: [Freeswitch-users] IP Phones with FreeSwitch References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com><65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com><507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com> Message-ID: <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> Hi again, sorry to be here again ! OK, now that I know that 3Com phones and FreeSwitch don't mix, my next question is about Cisco ! I see that the FreeSwitch Interoperability list includes Cisco phones such as the 7940 and 7960. I believe that these phones need user licenses to work with Cisco Call Manager. What I'd like to confirm is that I would not need any Cisco licenses or anything else to get a Cisco IP phone working with FreeSwitch. Again, I'd really appreciate feedback from anyone using either of these (or other) Cisco phones with FreeSwitch on whether any additional licenses or software are required to work with an "out of the box" FreeSwitch installation ? regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/33e4d45b/attachment-0002.html From sicfslist at gmail.com Tue Nov 3 13:09:09 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Tue, 03 Nov 2009 15:09:09 -0600 Subject: [Freeswitch-users] IP Phones with FreeSwitch In-Reply-To: <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com><65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com><507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com> <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> Message-ID: <4AF09BF5.4000802@gmail.com> Any of the Cisco phones with a SIP image should work fine ... no license required. SDR Dave Stevenson wrote: > Hi again, > > sorry to be here again ! > > OK, now that I know that 3Com phones and FreeSwitch don't mix, my next > question is about Cisco ! > > I see that the FreeSwitch Interoperability list includes Cisco phones > such as the 7940 and 7960. > > I believe that these phones need user licenses to work with Cisco Call > Manager. > > What I'd like to confirm is that I would not need any Cisco licenses > or anything else to get a Cisco IP phone working with FreeSwitch. > > Again, I'd really appreciate feedback from anyone using either of > these (or other) Cisco phones with FreeSwitch on whether any > additional licenses or software are required to work with an "out of > the box" FreeSwitch installation ? > > regards > Dave > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From william.suffill at gmail.com Tue Nov 3 13:10:19 2009 From: william.suffill at gmail.com (William Suffill) Date: Tue, 3 Nov 2009 16:10:19 -0500 Subject: [Freeswitch-users] IP Phones with FreeSwitch In-Reply-To: <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com> <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> Message-ID: <6b65470d0911031310g7f487ff9rfb61368280831471@mail.gmail.com> Cisco 7960 and the like that they push on the enterprise level for call manager also can be flashed with sip based firmware. I've only used the 7960 with the sip firmware. SPA942 and the like that used to be under Linksys/Sipura before that are targeted more toward smaller businesses and run SIP out of the box without any license complications. -- W From rupa at rupa.com Tue Nov 3 13:19:56 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 3 Nov 2009 15:19:56 -0600 Subject: [Freeswitch-users] IP Phones with FreeSwitch In-Reply-To: <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com> <1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> Message-ID: These phones work with FS, come by irc and you can talk to sekil about his use of them. In general, if you haven't invested in a bunch of phones, I'd recommend: Polycom 330,450,550 -- pick your price point snom - again, pick your price point These are generally well supported over the rest. On Tue, Nov 3, 2009 at 2:57 PM, Dave Stevenson wrote: > Hi again, > > sorry to be here again ! > > OK, now that I know that 3Com phones and FreeSwitch don't mix, my next > question is about Cisco ! > > I see that the FreeSwitch Interoperability list includes Cisco phones such > as the 7940 and 7960. > > I believe that these phones need user licenses to work with Cisco Call > Manager. > > What I'd like to confirm is that I would not need any Cisco licenses or > anything else to get a Cisco IP phone working with FreeSwitch. > > Again, I'd really appreciate feedback from anyone using either of these (or > other) Cisco phones with FreeSwitch on whether any additional licenses or > software are required to work with an "out of the box" FreeSwitch > installation ? > > regards > Dave > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From jerry.richards at teotech.com Tue Nov 3 13:22:11 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 3 Nov 2009 13:22:11 -0800 Subject: [Freeswitch-users] Dial Plan Question Message-ID: <0A46BCC1ED4C452CAD31DF64A734C492@greyhawk.tonecommander.com> My understanding of DialPlan/CallRouting is that it can be accomplished via static XML tags, or alternatively, via a DialPlan Application that interfaces with the dptools module. Question: If my above assumption is true, how does one select one approach over the other? What is the criteria/considerations that would govern the decision? Best Regards, Jerry From jerry.richards at teotech.com Tue Nov 3 13:25:58 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 3 Nov 2009 13:25:58 -0800 Subject: [Freeswitch-users] Error checking for PMP [general error] Message-ID: When I start Freeswitch, I see an "Error checking for PMP [general error]" as shown below. Does anyone know what could cause this? [root at TeoProxy bin]# ./freeswitch Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch -waste. auto-adjusting stack size for optimal performance.... 2009-11-02 10:12:27.17579 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-02 10:12:27.18373 [DEBUG] switch_event.c:553 Create event dispatch thread 0 2009-11-02 10:12:27.428749 [INFO] switch_nat.c:392 Scanning for NAT 2009-11-02 10:12:27.428885 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 2009-11-02 10:12:27.678480 [DEBUG] switch_nat.c:152 Checking for PMP 2/5 2009-11-02 10:12:27.679449 [DEBUG] switch_nat.c:152 Checking for PMP 3/5 2009-11-02 10:12:28.179388 [DEBUG] switch_nat.c:152 Checking for PMP 4/5 2009-11-02 10:12:29.179217 [DEBUG] switch_nat.c:152 Checking for PMP 5/5 2009-11-02 10:12:31.178879 [ERR] switch_nat.c:183 Error checking for PMP [general error] 2009-11-02 10:12:31.178902 [DEBUG] switch_nat.c:397 Checking for UPnP 2009-11-02 10:12:43.176881 [INFO] switch_nat.c:411 No PMP or UPnP NAT detected! 2009-11-02 10:12:43.210145 [INFO] switch_core_sqldb.c:538 Opening DB 2009-11-02 10:12:43.919804 [NOTICE] switch_scheduler.c:166 Starting task thread 2009-11-02 10:12:43.937881 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1257185563 2009-11-02 10:12:43.937980 [CONSOLE] switch_core.c:1449 Bringing up environment. 2009-11-02 10:12:43.937994 [CONSOLE] switch_core.c:1450 Loading Modules. 2009-11-02 10:12:43.938319 [INFO] switch_time.c:661 Timezone loaded 530 definitions 2009-11-02 10:12:43.938336 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_SOFTTIMER_MODULE] 2009-11-02 10:12:43.938351 [NOTICE] switch_loadable_module.c:228 Adding Timer 'soft' 2009-11-02 10:12:43.938413 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_PCM_MODULE] Best Regards, Jerry From stevendt at primrosebank.net Tue Nov 3 13:26:26 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 21:26:26 -0000 Subject: [Freeswitch-users] IP Phones with FreeSwitch References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com><65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com><507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com><127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com><1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> Message-ID: Rupa, thanks a lot for the pointers - I'm just about to try to pick up some phones, so the tips are timely. Actually, I have been trying the IRC thing today, but keep getting "connection refused", it's been a few years since I used IRC, but I think I have a Firewall problem that I'm working on. Hopefully, I'll be there soon, regards Dave ----- Original Message ----- From: "Rupa Schomaker" To: Sent: Tuesday, November 03, 2009 9:19 PM Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch > These phones work with FS, come by irc and you can talk to sekil about > his use of them. > > In general, if you haven't invested in a bunch of phones, I'd recommend: > > Polycom 330,450,550 -- pick your price point > snom - again, pick your price point > > These are generally well supported over the rest. > > On Tue, Nov 3, 2009 at 2:57 PM, Dave Stevenson > wrote: >> Hi again, >> >> sorry to be here again ! >> >> OK, now that I know that 3Com phones and FreeSwitch don't mix, my next >> question is about Cisco ! >> >> I see that the FreeSwitch Interoperability list includes Cisco phones >> such >> as the 7940 and 7960. >> >> I believe that these phones need user licenses to work with Cisco Call >> Manager. >> >> What I'd like to confirm is that I would not need any Cisco licenses or >> anything else to get a Cisco IP phone working with FreeSwitch. >> >> Again, I'd really appreciate feedback from anyone using either of these >> (or >> other) Cisco phones with FreeSwitch on whether any additional licenses or >> software are required to work with an "out of the box" FreeSwitch >> installation ? >> >> regards >> Dave >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From stevendt at primrosebank.net Tue Nov 3 13:28:41 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 3 Nov 2009 21:28:41 -0000 Subject: [Freeswitch-users] IP Phones with FreeSwitch References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com><1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> <6b65470d0911031310g7f487ff9rfb61368280831471@mail.gmail.com> Message-ID: Thanks a lot - to both William and Shelby, that makes me more confident about trying out at least one Cisco and Rupa has just given me a few more options, so, hopefully, I won't make the 3Com mistake again ! regards Dave ----- Original Message ----- From: "William Suffill" To: Sent: Tuesday, November 03, 2009 9:10 PM Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch > Cisco 7960 and the like that they push on the enterprise level for > call manager also can be flashed with sip based firmware. I've only > used the 7960 with the sip firmware. > > > SPA942 and the like that used to be under Linksys/Sipura before that > are targeted more toward smaller businesses and run SIP out of the box > without any license complications. > > -- W > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jerry.richards at teotech.com Tue Nov 3 13:35:05 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 3 Nov 2009 13:35:05 -0800 Subject: [Freeswitch-users] WARNING On Inbound Call Question Message-ID: I have my Freeswitch server with an installed Sangoma A101D card. Most everything works okay, however, when I get an inbound call from the PSTN, I see the following warning show up in the log. Additionally, the caller (on the PSTN) does not hear ringback, and if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. Here are the two warnings: [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA Here is the log of the warning upon an inbound call: freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> 2009-11-02 09:06:01.664835 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0 Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176] 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on 1:1 from DOWN to RING 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig [START] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound channel OpenZAP/1:1/5384 2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384) State Change CS_NEW -> CS_INIT 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_INIT 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/5384) State INIT 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384) State Change CS_INIT -> CS_ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/5384) State INIT going to sleep 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/5384) State ROUTING 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384 CHANNEL ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78 OpenZAP/1:1/5384 Standard ROUTING 2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing 4253813176->5384 in context default Dialplan: OpenZAP/1:1/5384 parsing [default->unloop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->tod_example] continue=true Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example] Dialplan: OpenZAP/1:1/5384 Action set(open=true) Dialplan: OpenZAP/1:1/5384 parsing [default->SangomaPRI] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI] destination_number(5384) =~ /^9(\d+)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->global-intercept] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept] destination_number(5384) =~ /^(5380)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->group-intercept] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept] destination_number(5384) =~ /^\*8$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->intercept-ext] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext] destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->redial] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384) =~ /^870$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->global] continue=true Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: OpenZAP/1:1/5384 Absolute Condition [global] Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-2] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-2] destination_number(5384) =~ /^9001$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-1] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-1] destination_number(5384) =~ /^9000$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] destination_number(5384) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] destination_number(5384) =~ /^779$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call_return] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call_return] destination_number(5384) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->del-group] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [del-group] destination_number(5384) =~ /^80(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->add-group] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [add-group] destination_number(5384) =~ /^81(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-simo] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-simo] destination_number(5384) =~ /^82(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-order] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-order] destination_number(5384) =~ /^83(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->extension-intercom] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [extension-intercom] destination_number(5384) =~ /^8(5[34][8901][0-9])$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->Local_Extension] continue=false Dialplan: OpenZAP/1:1/5384 Regex (PASS) [Local_Extension] destination_number(5384) =~ /^(5[34][8901][0-9])$/ break=on-false Dialplan: OpenZAP/1:1/5384 Action set(dialed_extension=5384) Dialplan: OpenZAP/1:1/5384 Action export(dialed_extension=5384) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: OpenZAP/1:1/5384 Action set(ringback=${us-ring}) Dialplan: OpenZAP/1:1/5384 Action set(transfer_ringback=local_stream://moh) Dialplan: OpenZAP/1:1/5384 Action set(call_timeout=30) Dialplan: OpenZAP/1:1/5384 Action set(hangup_after_bridge=true) Dialplan: OpenZAP/1:1/5384 Action set(continue_on_fail=true) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: OpenZAP/1:1/5384 Action answer() Dialplan: OpenZAP/1:1/5384 Action sleep(1000) Dialplan: OpenZAP/1:1/5384 Action voicemail(default ${domain_name} ${dialed_extension}) 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:114 (OpenZAP/1:1/5384) State Change CS_ROUTING -> CS_EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/5384) State ROUTING going to sleep 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:491 (OpenZAP/1:1/5384) State EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] mod_openzap.c:408 OpenZAP/1:1/5384 CHANNEL EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:151 OpenZAP/1:1/5384 Standard EXECUTE EXECUTE OpenZAP/1:1/5384 set(open=true) 2009-11-02 09:06:01.666685 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [open]=[true] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-spymap/4253813176/b678f311-ab74-4cc1-afac-b83d89a 53132) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/4253813176/5384) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/global/b678f311-ab74-4cc1-afac-b83d89a5 3132) EXECUTE OpenZAP/1:1/5384 set(dialed_extension=5384) 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [dialed_extension]=[5384] EXECUTE OpenZAP/1:1/5384 export(dialed_extension=5384) 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:886 EXPORT [dialed_extension]=[5384] EXECUTE OpenZAP/1:1/5384 bind_meta_app(1 b s execute_extension::dx XML features) 2009-11-02 09:06:01.667682 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 execute_extension::dx XML features EXECUTE OpenZAP/1:1/5384 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 -01.wav) 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 -01.wav EXECUTE OpenZAP/1:1/5384 bind_meta_app(3 b s execute_extension::cf XML features) 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 execute_extension::cf XML features EXECUTE OpenZAP/1:1/5384 set(ringback=%(2000,4000,440.0,480.0)) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE OpenZAP/1:1/5384 set(transfer_ringback=local_stream://moh) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [transfer_ringback]=[local_stream://moh] EXECUTE OpenZAP/1:1/5384 set(call_timeout=30) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [call_timeout]=[30] EXECUTE OpenZAP/1:1/5384 set(hangup_after_bridge=true) 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [hangup_after_bridge]=[true] EXECUTE OpenZAP/1:1/5384 set(continue_on_fail=true) 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [continue_on_fail]=[true] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-call_return/5384/4253813176) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial_ext/5384/b678f311-ab74-4cc1-afac-b83d89 a53132) EXECUTE OpenZAP/1:1/5384 set(called_party_callgroup=techsupport) 2009-11-02 09:06:01.670679 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [called_party_callgroup]=[techsupport] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/techsupport/b678f311-ab74-4cc1-afac-b83 d89a53132) EXECUTE OpenZAP/1:1/5384 bridge(user/5384 at 192.168.72.141) 2009-11-02 09:06:01.671683 [DEBUG] switch_ivr_originate.c:1027 variable string 0 = [presence_id=5384 at 192.168.72.141] 2009-11-02 09:06:01.671683 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:5384 at 192.168.72.163:5060 [9e7b8fae-6194-430c-951b-948ebd2c2a3b] 2009-11-02 09:06:01.671683 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_NEW -> CS_INIT 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_INIT 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:83 sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA INIT 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_INIT -> CS_ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT going to sleep 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:130 sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING going to sleep 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:5384 at 192.168.72.163:5060 entering state [calling][0] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:5384 at 192.168.72.163:5060) State CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:1701 OpenZAP/1:1/5384 receive message [PROGRESS] 2009-11-02 09:06:01.673742 [DEBUG] mod_openzap.c:759 Changing state on 1:1 from RING to PROGRESS 2009-11-02 09:06:01.674787 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [PROGRESS] 2009-11-02 09:06:01.675844 [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 2009-11-02 09:06:01.684776 [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA 2009-11-02 09:06:01.684776 [DEBUG] switch_core_session.c:630 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.684776 [NOTICE] switch_ivr_originate.c:1701 Pre-Answer OpenZAP/1:1/5384! 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1777 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-02 09:06:01.693835 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:5384 at 192.168.72.163:5060 entering state [proceeding][180] 2009-11-02 09:06:01.693835 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:5384 at 192.168.72.163:5060! 2009-11-02 09:06:01.705777 [DEBUG] switch_core_io.c:649 OpenZAP/1:1/5384 receive message [TRANSCODING_NECESSARY] freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> Best Regards, Jerry From sicfslist at gmail.com Tue Nov 3 13:44:55 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Tue, 03 Nov 2009 15:44:55 -0600 Subject: [Freeswitch-users] Dial Plan Question In-Reply-To: <0A46BCC1ED4C452CAD31DF64A734C492@greyhawk.tonecommander.com> References: <0A46BCC1ED4C452CAD31DF64A734C492@greyhawk.tonecommander.com> Message-ID: <4AF0A457.5080702@gmail.com> I think the real question is what are you trying to do ... for some things it's very easy to just whip up a static XML file and be done with it. For others you probably want some sort of interaction with a DB. The options here are pretty endless: -- XML curl -- handing off the call to a script call from a static dial plan (use lua if there is going to be any load) -- event_socket -- mod_lcr But ultimately I think it's what you're trying to accomplish that matters. For a PBX install I'd say static files is probably about as easy as it is going to get. For delivering a service you'd probably want interaction with a DB. I've use XML curl a lot and have even starting using direct DB queries from static dialplans using mod_memcache and memcachedb (not memcache ... persistent storage). SDR Jerry Richards wrote: > My understanding of DialPlan/CallRouting is that it can be accomplished via > static XML tags, or alternatively, via a DialPlan Application that > interfaces with the dptools module. > > Question: If my above assumption is true, how does one select one approach > over the other? What is the criteria/considerations that would govern the > decision? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From frank at carmickle.com Tue Nov 3 14:00:22 2009 From: frank at carmickle.com (Frank Carmickle) Date: Tue, 3 Nov 2009 17:00:22 -0500 Subject: [Freeswitch-users] portaudio error In-Reply-To: <20091103172437.GA9418@hijacked.us> References: <20091103170110.GK10757@base.carmickle.com> <20091103172437.GA9418@hijacked.us> Message-ID: <20091103220022.GL10757@base.carmickle.com> On Tue, Nov 03, Andrew Thompson wrote: > On Tue, Nov 03, 2009 at 12:01:10PM -0500, Frank Carmickle wrote: > > Hello > > > > Debian lenny with svn15321 > > > > freeswitch at internal> load mod_portaudio > > -ERR [module load file routine returned an error] > > > > 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input devicefreeswitch at internal> 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 Cannot find an input device > > 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_portaudio.so > > **Module load routine returned an error** > > > Try installing the alsa development headers, it's got some stupid name > on debian like libasound2-devel or something. Then re-build the > portaudio module and library (a couple well placed make cleans should do > it). Hi Libasound2-dev is still installed. I have had PA working in the passed. I think it was as of svn 14000 or so. Thanks for the help. --FC From rupa at rupa.com Tue Nov 3 14:18:35 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 3 Nov 2009 16:18:35 -0600 Subject: [Freeswitch-users] Error checking for PMP [general error] In-Reply-To: References: Message-ID: If you don't have a router with NAT-PMP enabled then it is expected. Same if you don't have upnp. If you are behind a NAT, it is in your best interest to enable one or the other in your router. It will save you a bunch of headaches... On Tue, Nov 3, 2009 at 3:25 PM, Jerry Richards wrote: > > When I start Freeswitch, I see an "Error checking for PMP [general error]" > as shown below. ?Does anyone know what could cause this? > > > [root at TeoProxy bin]# ./freeswitch > Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch > -waste. > auto-adjusting stack size for optimal performance.... > 2009-11-02 10:12:27.17579 [INFO] switch_event.c:565 Activate Eventing > Engine. > 2009-11-02 10:12:27.18373 [DEBUG] switch_event.c:553 Create event dispatch > thread 0 > 2009-11-02 10:12:27.428749 [INFO] switch_nat.c:392 Scanning for NAT > 2009-11-02 10:12:27.428885 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 > 2009-11-02 10:12:27.678480 [DEBUG] switch_nat.c:152 Checking for PMP 2/5 > 2009-11-02 10:12:27.679449 [DEBUG] switch_nat.c:152 Checking for PMP 3/5 > 2009-11-02 10:12:28.179388 [DEBUG] switch_nat.c:152 Checking for PMP 4/5 > 2009-11-02 10:12:29.179217 [DEBUG] switch_nat.c:152 Checking for PMP 5/5 > 2009-11-02 10:12:31.178879 [ERR] switch_nat.c:183 Error checking for PMP > [general error] > 2009-11-02 10:12:31.178902 [DEBUG] switch_nat.c:397 Checking for UPnP > 2009-11-02 10:12:43.176881 [INFO] switch_nat.c:411 No PMP or UPnP NAT > detected! > 2009-11-02 10:12:43.210145 [INFO] switch_core_sqldb.c:538 Opening DB > 2009-11-02 10:12:43.919804 [NOTICE] switch_scheduler.c:166 Starting task > thread > 2009-11-02 10:12:43.937881 [DEBUG] switch_scheduler.c:214 Added task 1 > heartbeat (core) to run at 1257185563 > 2009-11-02 10:12:43.937980 [CONSOLE] switch_core.c:1449 Bringing up > environment. > 2009-11-02 10:12:43.937994 [CONSOLE] switch_core.c:1450 Loading Modules. > 2009-11-02 10:12:43.938319 [INFO] switch_time.c:661 Timezone loaded 530 > definitions > 2009-11-02 10:12:43.938336 [CONSOLE] switch_loadable_module.c:889 > Successfully Loaded [CORE_SOFTTIMER_MODULE] > 2009-11-02 10:12:43.938351 [NOTICE] switch_loadable_module.c:228 Adding > Timer 'soft' > 2009-11-02 10:12:43.938413 [CONSOLE] switch_loadable_module.c:889 > Successfully Loaded [CORE_PCM_MODULE] > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From anthony.minessale at gmail.com Tue Nov 3 14:23:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 16:23:09 -0600 Subject: [Freeswitch-users] WARNING On Inbound Call Question In-Reply-To: References: Message-ID: <191c3a030911031423y58905d44mc00a7d949573ff86@mail.gmail.com> can you try the same thing with the latest trunk or pre-release tarball. On Tue, Nov 3, 2009 at 3:35 PM, Jerry Richards wrote: > > I have my Freeswitch server with an installed Sangoma A101D card. Most > everything works okay, however, when I get an inbound call from the PSTN, I > see the following warning show up in the log. Additionally, the caller (on > the PSTN) does not hear ringback, and if the call is not answered within > about 12 seconds, the call ends (so it doesn't go to voice mail). If I > make > a call from one internal phone to another, then it will go to voice mail > after 30 seconds. > > > Here are the two warnings: > > [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] > Rc=0 CSid=0 Seq=11 > [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to > PROGRESS_MEDIA > > > Here is the log of the warning upon an inbound call: > > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> 2009-11-02 09:06:01.664835 > [WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0 > Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176] > 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on > 1:1 from DOWN to RING > 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING] > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig > [START] > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound > channel OpenZAP/1:1/5384 > 2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel > OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132] > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384) > State Change CS_NEW -> CS_INIT > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal > OpenZAP/1:1/5384 [BREAK] > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 > (OpenZAP/1:1/5384) Running State Change CS_INIT > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 > (OpenZAP/1:1/5384) State INIT > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384) > State Change CS_INIT -> CS_ROUTING > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal > OpenZAP/1:1/5384 [BREAK] > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 > (OpenZAP/1:1/5384) State INIT going to sleep > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 > (OpenZAP/1:1/5384) Running State Change CS_ROUTING > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484 > (OpenZAP/1:1/5384) State ROUTING > 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384 > CHANNEL ROUTING > 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78 > OpenZAP/1:1/5384 Standard ROUTING > 2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing > 4253813176->5384 in context default > Dialplan: OpenZAP/1:1/5384 parsing [default->unloop] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~ > /^true$/ break=on-false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~ > /^true$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->tod_example] continue=true > Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example] > Dialplan: OpenZAP/1:1/5384 Action set(open=true) > Dialplan: OpenZAP/1:1/5384 parsing [default->SangomaPRI] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI] > destination_number(5384) =~ /^9(\d+)$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->global-intercept] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept] > destination_number(5384) =~ /^(5380)$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->group-intercept] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept] > destination_number(5384) =~ /^\*8$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->intercept-ext] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext] > destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->redial] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384) > =~ > /^870$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->global] continue=true > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~ > /^true$/ break=never > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > Dialplan: OpenZAP/1:1/5384 Absolute Condition [global] > Dialplan: OpenZAP/1:1/5384 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: OpenZAP/1:1/5384 Action > > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe > r}) > Dialplan: OpenZAP/1:1/5384 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-2] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-2] > destination_number(5384) =~ /^9001$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-1] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-1] > destination_number(5384) =~ /^9000$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] > destination_number(5384) > =~ /^88(.*)$|^\*0(.*)$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] > destination_number(5384) > =~ /^779$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->call_return] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call_return] > destination_number(5384) =~ /^\*69$|^869$|^lcr$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->del-group] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [del-group] > destination_number(5384) > =~ /^80(\d{2})$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->add-group] continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [add-group] > destination_number(5384) > =~ /^81(\d{2})$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-simo] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-simo] > destination_number(5384) =~ /^82(\d{2})$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-order] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-order] > destination_number(5384) =~ /^83(\d{2})$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->extension-intercom] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [extension-intercom] > destination_number(5384) =~ /^8(5[34][8901][0-9])$/ break=on-false > Dialplan: OpenZAP/1:1/5384 parsing [default->Local_Extension] > continue=false > Dialplan: OpenZAP/1:1/5384 Regex (PASS) [Local_Extension] > destination_number(5384) =~ /^(5[34][8901][0-9])$/ break=on-false > Dialplan: OpenZAP/1:1/5384 Action set(dialed_extension=5384) > Dialplan: OpenZAP/1:1/5384 Action export(dialed_extension=5384) > Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(1 b s execute_extension::dx > XML features) > Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(2 b s > > record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft > ime(%Y-%m-%d-%H-%M-%S)}.wav) > Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(3 b s execute_extension::cf > XML features) > Dialplan: OpenZAP/1:1/5384 Action set(ringback=${us-ring}) > Dialplan: OpenZAP/1:1/5384 Action set(transfer_ringback=local_stream://moh) > Dialplan: OpenZAP/1:1/5384 Action set(call_timeout=30) > Dialplan: OpenZAP/1:1/5384 Action set(hangup_after_bridge=true) > Dialplan: OpenZAP/1:1/5384 Action set(continue_on_fail=true) > Dialplan: OpenZAP/1:1/5384 Action > > hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe > r}) > Dialplan: OpenZAP/1:1/5384 Action > hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) > Dialplan: OpenZAP/1:1/5384 Action > set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}) > Dialplan: OpenZAP/1:1/5384 Action > hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) > Dialplan: OpenZAP/1:1/5384 Action > bridge(user/${dialed_extension}@${domain_name}) > Dialplan: OpenZAP/1:1/5384 Action answer() > Dialplan: OpenZAP/1:1/5384 Action sleep(1000) > Dialplan: OpenZAP/1:1/5384 Action voicemail(default ${domain_name} > ${dialed_extension}) > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:114 > (OpenZAP/1:1/5384) State Change CS_ROUTING -> CS_EXECUTE > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_session.c:932 Send signal > OpenZAP/1:1/5384 [BREAK] > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:484 > (OpenZAP/1:1/5384) State ROUTING going to sleep > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:398 > (OpenZAP/1:1/5384) Running State Change CS_EXECUTE > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:491 > (OpenZAP/1:1/5384) State EXECUTE > 2009-11-02 09:06:01.666685 [DEBUG] mod_openzap.c:408 OpenZAP/1:1/5384 > CHANNEL EXECUTE > 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:151 > OpenZAP/1:1/5384 Standard EXECUTE > EXECUTE OpenZAP/1:1/5384 set(open=true) > 2009-11-02 09:06:01.666685 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [open]=[true] > EXECUTE OpenZAP/1:1/5384 > > hash(insert/192.168.72.141-spymap/4253813176/b678f311-ab74-4cc1-afac-b83d89a > 53132) > EXECUTE OpenZAP/1:1/5384 > hash(insert/192.168.72.141-last_dial/4253813176/5384) > EXECUTE OpenZAP/1:1/5384 > > hash(insert/192.168.72.141-last_dial/global/b678f311-ab74-4cc1-afac-b83d89a5 > 3132) > EXECUTE OpenZAP/1:1/5384 set(dialed_extension=5384) > 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [dialed_extension]=[5384] > EXECUTE OpenZAP/1:1/5384 export(dialed_extension=5384) > 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:886 EXPORT > [dialed_extension]=[5384] > EXECUTE OpenZAP/1:1/5384 bind_meta_app(1 b s execute_extension::dx XML > features) > 2009-11-02 09:06:01.667682 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 > execute_extension::dx XML features > EXECUTE OpenZAP/1:1/5384 bind_meta_app(2 b s > > record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 > -01.wav) > 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 > > record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 > -01.wav > EXECUTE OpenZAP/1:1/5384 bind_meta_app(3 b s execute_extension::cf XML > features) > 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 > execute_extension::cf XML features > EXECUTE OpenZAP/1:1/5384 set(ringback=%(2000,4000,440.0,480.0)) > 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [ringback]=[%(2000,4000,440.0,480.0)] > EXECUTE OpenZAP/1:1/5384 set(transfer_ringback=local_stream://moh) > 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [transfer_ringback]=[local_stream://moh] > EXECUTE OpenZAP/1:1/5384 set(call_timeout=30) > 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [call_timeout]=[30] > EXECUTE OpenZAP/1:1/5384 set(hangup_after_bridge=true) > 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [hangup_after_bridge]=[true] > EXECUTE OpenZAP/1:1/5384 set(continue_on_fail=true) > 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [continue_on_fail]=[true] > EXECUTE OpenZAP/1:1/5384 > hash(insert/192.168.72.141-call_return/5384/4253813176) > EXECUTE OpenZAP/1:1/5384 > > hash(insert/192.168.72.141-last_dial_ext/5384/b678f311-ab74-4cc1-afac-b83d89 > a53132) > EXECUTE OpenZAP/1:1/5384 set(called_party_callgroup=techsupport) > 2009-11-02 09:06:01.670679 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET > [called_party_callgroup]=[techsupport] > EXECUTE OpenZAP/1:1/5384 > > hash(insert/192.168.72.141-last_dial/techsupport/b678f311-ab74-4cc1-afac-b83 > d89a53132) > EXECUTE OpenZAP/1:1/5384 bridge(user/5384 at 192.168.72.141) > 2009-11-02 09:06:01.671683 [DEBUG] switch_ivr_originate.c:1027 variable > string 0 = [presence_id=5384 at 192.168.72.141] > 2009-11-02 09:06:01.671683 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/sip:5384 at 192.168.72.163:5060 > [9e7b8fae-6194-430c-951b-948ebd2c2a3b] > 2009-11-02 09:06:01.671683 [DEBUG] mod_sofia.c:2811 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_NEW -> > CS_INIT > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_INIT > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT > 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:83 > sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA INIT > 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:111 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_INIT -> > CS_ROUTING > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT going to sleep > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change > CS_ROUTING > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING > 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:130 > sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA ROUTING > 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:63 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING going to sleep > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change > CS_CONSUME_MEDIA > 2009-11-02 09:06:01.672688 [DEBUG] sofia.c:3289 Channel > sofia/internal/sip:5384 at 192.168.72.163:5060 entering state [calling][0] > 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:5384 at 192.168.72.163:5060) State CONSUME_MEDIA > 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:1701 > OpenZAP/1:1/5384 receive message [PROGRESS] > 2009-11-02 09:06:01.673742 [DEBUG] mod_openzap.c:759 Changing state on 1:1 > from RING to PROGRESS > 2009-11-02 09:06:01.674787 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE > [PROGRESS] > 2009-11-02 09:06:01.675844 [WARNING] ss7_boost_client.c:218 TX EVENT (N): > CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 > 2009-11-02 09:06:01.684776 [WARNING] mod_openzap.c:761 VETO Changing state > on 1:1 from PROGRESS to PROGRESS_MEDIA > 2009-11-02 09:06:01.684776 [DEBUG] switch_core_session.c:630 Send signal > OpenZAP/1:1/5384 [BREAK] > 2009-11-02 09:06:01.684776 [NOTICE] switch_ivr_originate.c:1701 Pre-Answer > OpenZAP/1:1/5384! > 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1718 Raw Codec > Activation Success L16 at 8000hz 1 channel 20ms > 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1777 Play > Ringback > Tone [%(2000,4000,440.0,480.0)] > 2009-11-02 09:06:01.693835 [DEBUG] sofia.c:3289 Channel > sofia/internal/sip:5384 at 192.168.72.163:5060 entering state > [proceeding][180] > 2009-11-02 09:06:01.693835 [NOTICE] sofia.c:3353 Ring-Ready > sofia/internal/sip:5384 at 192.168.72.163:5060! > 2009-11-02 09:06:01.705777 [DEBUG] switch_core_io.c:649 OpenZAP/1:1/5384 > receive message [TRANSCODING_NECESSARY] > > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > freeswitch at TeoProxy.greyhawk.tonecommander.com> > > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/c9d835e4/attachment-0002.html From mkitchin.public at gmail.com Tue Nov 3 14:19:04 2009 From: mkitchin.public at gmail.com (mkitchin.public at gmail.com) Date: Tue, 03 Nov 2009 16:19:04 -0600 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Message-ID: <4AF0AC58.3010506@gmail.com> I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: 7940-Config.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/cea1d113/attachment-0002.txt -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: FS-Log.log Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/cea1d113/attachment-0002.pl From hjqlopez at hotmail.com Tue Nov 3 16:05:53 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Tue, 3 Nov 2009 19:05:53 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media Message-ID: Hi, I tried r15332 and set in the sofia profile: a) bypass_media_after_bridge=true only b) bypass_media_after_bridge=true, param name="media-option" value="resume-media-on-hold"/> In both cases FS is hanging up the initial call (A to FS) after accepting the REFER to C: A <- reINVITE with FS' SDP <- FS A -> 200 -> FS A <- ACK <- FS A <- BYE <- FS The call to C is not even tried. I found this line is the logs that could give some idea: 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] after sending the ACK for the reINVITE Regards, Humberto >please try r15326 >I think i have it working. > >I recommend for optimal results you set bypass_media_after_bridge=true >either as a global or in your DP in place of bypass_media=true > > >On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana wrote: > >> Hi Mike, >> >> I re-tried with trunk rev 15319 but I got almost the same behavior: There >> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But >> still there is no reINVITE for A (with C's SDP) after the call from FS to C >> is established. >> >> Anyway, we decided for now to do a different implementation but if you want >> to explore more in this issue count me in ;-) >> >> >> Thank you very much! >> >> Humberto _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://go.microsoft.com/?linkid=9691817 From msc at freeswitch.org Tue Nov 3 16:16:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Nov 2009 16:16:10 -0800 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> References: <24251951.post@talk.nabble.com> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> Message-ID: <87f2f3b90911031616t2c731372i192f514d302522e9@mail.gmail.com> On Tue, Nov 3, 2009 at 11:11 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > It appears that Tony has already added an option (amazing) BUT you > should really be setup for central provisioning with an installed base > that large... You'll eventually have issues that *NO* amount of > Tony/FreeSWITCH magic can fix. > > Kristian is correct. Listen to him because he's familiar with having lots and lots of units out in the field. The bandage Tony applied will eventually wear off. The long-term solution is to treat the malady and not the symptom. I'm certain that members of the FS community could point you toward some resources to assist with central provisioning. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/31fc9d54/attachment-0002.html From msc at freeswitch.org Tue Nov 3 16:27:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Nov 2009 16:27:30 -0800 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <4AF0AC58.3010506@gmail.com> References: <4AF0AC58.3010506@gmail.com> Message-ID: <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public at gmail.com < mkitchin.public at gmail.com> wrote: > I'm working on an alternative to a $120,000 Cisco phone system that my > > company is looking at. I got Freeswitch installed on CentOS last week > using the Quick and Dirty instructions. That part was painless. We had a > few 7940s laying around. After some wrestling with it, I got the latest > SIP firmware installed and what I hoped was a functional config > (attached). X-Lite phones can call each other no problem. 7940s can call > X-Lite no problem. Anytime I try and call a 7940, it goes straight to > voicemail. I attached a log file that shows the activity when trying to > call a7940 from X-Lite. > X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is > nshplpbx1.unix/10.85.0.53. Everything is on the same LAN. Different > subnets, but no firewalls. > I didn't see anything that said posting attachments was frowned upon. I > apologize if it isn't appropriate. I'm guessing this is something simple > and I'm just clueless on how to diagnose the issue. > I'm not tied to using this model for good, but it is what we had laying > around. Any help would be greatly appreciated. Next step is configuring > it to talk to Verizon VOIP over a DS3. > > Thanks, > Matthew Kitchin > > Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now. Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd say first thing to do is capture the SIP traffic to see if there are any clues. A "normal temporary failure" doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org) You can also join the IRC channel #freeswitch on irc.freenode.net and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/c583af89/attachment-0002.html From brian at freeswitch.org Tue Nov 3 16:31:46 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 18:31:46 -0600 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <338A8EA0-956E-4B49-8234-AC534244FDFE@freeswitch.org> Do you have ANY nat involved? /b On Nov 3, 2009, at 6:05 PM, Humberto Quintana wrote: > 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/ > external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] > after sending the ACK for the reINVITE From anthony.minessale at gmail.com Tue Nov 3 16:38:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Nov 2009 18:38:20 -0600 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <191c3a030911031638h340bc54dm83ac9c3a888e5ba0@mail.gmail.com> I don't know what you are talking about anymore. The scenario I had tested is when a call is bridged in bypass_media=true bridge and you blind transfer that call back to the dialplan as soon as it hits the routing state it will resume media. it has been confirmed to not work and confirmed to have been fixed several time and if you are still having a problem you must have something blocking some of your packets or something . You have to understand that sip is a protocol and your description is completely non-standard. Perhaps you should get a console trace and attach it to a jira. The trace probably makes more sense to me. sofia profile internal siptrace on console loglevel debug reproduce and attach the whole capture. On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > > Hi, > > I tried r15332 and set in the sofia profile: > > a) bypass_media_after_bridge=true only > b) bypass_media_after_bridge=true, param name="media-option" > value="resume-media-on-hold"/> > > > In both cases FS is hanging up the initial call (A to FS) after accepting > the REFER to C: > > A <- reINVITE with FS' SDP <- FS > A -> 200 -> FS > A <- ACK <- FS > A <- BYE <- FS > > The call to C is not even tried. > > I found this line is the logs that could give some idea: > > 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup > sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] > after sending the ACK for the reINVITE > > > Regards, > > > Humberto > > >please try r15326 > >I think i have it working. > > > >I recommend for optimal results you set bypass_media_after_bridge=true > >either as a global or in your DP in place of bypass_media=true > > > > > >On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana hotmail.com>wrote: > > > >> Hi Mike, > >> > >> I re-tried with trunk rev 15319 but I got almost the same behavior: > There > >> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. > But > >> still there is no reINVITE for A (with C's SDP) after the call from FS > to C > >> is established. > >> > >> Anyway, we decided for now to do a different implementation but if you > want > >> to explore more in this issue count me in ;-) > >> > >> > >> Thank you very much! > >> > >> Humberto > > > _________________________________________________________________ > Windows Live: Friends get your Flickr, Yelp, and Digg updates when they > e-mail you. > http://go.microsoft.com/?linkid=9691817 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/29ef0f9b/attachment-0002.html From peder at networkoblivion.com Tue Nov 3 18:12:54 2009 From: peder at networkoblivion.com (Peder) Date: Tue, 3 Nov 2009 20:12:54 -0600 Subject: [Freeswitch-users] IP Phones with FreeSwitch In-Reply-To: References: <6FF5B673AB13485EB0DE1C05C2E7FF70@bp1.ad.bp.com> <65d96fc80911031153q2dca4834wb547bd4682269520@mail.gmail.com> <507898380911031218x2a63c14cgdc07d8f80dc230f6@mail.gmail.com> <127BA5C26D55406A97556953CAA85336@bp1.ad.bp.com><1D962668589942B880D8FE0B05CE50E0@bp1.ad.bp.com> <6b65470d0911031310g7f487ff9rfb61368280831471@mail.gmail.com> Message-ID: <037301ca5cf4$5193d960$f4bb8c20$@com> FYI, you can't do "presence" with the Cisco phones, so you can't see if someone is on the phone. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dave Stevenson Sent: Tuesday, November 03, 2009 3:29 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch Thanks a lot - to both William and Shelby, that makes me more confident about trying out at least one Cisco and Rupa has just given me a few more options, so, hopefully, I won't make the 3Com mistake again ! regards Dave ----- Original Message ----- From: "William Suffill" To: Sent: Tuesday, November 03, 2009 9:10 PM Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch > Cisco 7960 and the like that they push on the enterprise level for > call manager also can be flashed with sip based firmware. I've only > used the 7960 with the sip firmware. > > > SPA942 and the like that used to be under Linksys/Sipura before that > are targeted more toward smaller businesses and run SIP out of the box > without any license complications. > > -- W > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ujjval at simplesignal.com Tue Nov 3 18:27:59 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Tue, 3 Nov 2009 18:27:59 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> , <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7170307767D@EXMBXCLUS01.citservers.local> Was that sarcasm or you really mean it? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, November 02, 2009 9:08 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator you know I have heard this before... It seems to ONLY be AT&T /b On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote: > Yes, I think I did. However here is what furthur testing revelas. If > I dial in from AT&T cell phone, I do not see any DTMF using Don's > IVR.xml.conf to call my conf app. But when I dial the same number > using a Verizon Cell, it works. > > When I dial a number that is provisioned to call the Conf App > directly from the public.xml dialplan...it works even with the same > AT&T cell phone... > > Strange behaviour _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gromovd at gmail.com Tue Nov 3 19:16:43 2009 From: gromovd at gmail.com (Dmitry Gromov) Date: Tue, 3 Nov 2009 22:16:43 -0500 Subject: [Freeswitch-users] Wiki typo Message-ID: Hi! Was just reading wiki here: http://wiki.freeswitch.org/wiki/Home_PBX_Example It lists sample sofia.conf.xml which has this parameter: I think it should read inbound-*bypass*-media and not inbound-*no*-media... I know, it says "outdated" but still, can be confusing. Anyone here who can edit wiki and correct? Thanks, Dmitry -- DG NJ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/e889dcd2/attachment-0002.html From brian at freeswitch.org Tue Nov 3 19:34:24 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Nov 2009 21:34:24 -0600 Subject: [Freeswitch-users] Wiki typo In-Reply-To: References: Message-ID: <26432E7B-B9E7-4F4E-921F-E4B6C9AD4F6C@freeswitch.org> Yes you can login and edit the wiki yourself. Thanks, /b On Nov 3, 2009, at 9:16 PM, Dmitry Gromov wrote: > Hi! > > Was just reading wiki here: http://wiki.freeswitch.org/wiki/Home_PBX_Example > It lists sample sofia.conf.xml which has this parameter: > > > I think it should read inbound-bypass-media and not inbound-no- > media... > > I know, it says "outdated" but still, can be confusing. > > Anyone here who can edit wiki and correct? > > Thanks, > Dmitry > > -- > DG > NJ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/2f14b1f4/attachment-0002.html From gromovd at gmail.com Tue Nov 3 19:45:22 2009 From: gromovd at gmail.com (Dmitry Gromov) Date: Tue, 3 Nov 2009 22:45:22 -0500 Subject: [Freeswitch-users] Wiki typo In-Reply-To: <26432E7B-B9E7-4F4E-921F-E4B6C9AD4F6C@freeswitch.org> References: <26432E7B-B9E7-4F4E-921F-E4B6C9AD4F6C@freeswitch.org> Message-ID: Thanks, done - page has been corrected! On Tue, Nov 3, 2009 at 22:34, Brian West wrote: > Yes you can login and edit the wiki yourself. > > > You know... I actually spent some time looking for login/create account link when I noticed this typo. No idea why I did not see it then :) -- DG NJ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/2d7dbdd5/attachment-0002.html From mkitchin.public at gmail.com Tue Nov 3 20:10:14 2009 From: mkitchin.public at gmail.com (mkitchin.public at gmail.com) Date: Tue, 03 Nov 2009 22:10:14 -0600 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> Message-ID: <4AF0FEA6.7070308@gmail.com> Michael Collins wrote: > > > On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public at gmail.com > > wrote: > > I'm working on an alternative to a $120,000 Cisco phone system that my > > company is looking at. I got Freeswitch installed on CentOS last week > using the Quick and Dirty instructions. That part was painless. We > had a > few 7940s laying around. After some wrestling with it, I got the > latest > SIP firmware installed and what I hoped was a functional config > (attached). X-Lite phones can call each other no problem. 7940s > can call > X-Lite no problem. Anytime I try and call a 7940, it goes straight to > voicemail. I attached a log file that shows the activity when > trying to > call a7940 from X-Lite. > X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is > nshplpbx1.unix/10.85.0.53 . Everything is on > the same LAN. Different > subnets, but no firewalls. > I didn't see anything that said posting attachments was frowned > upon. I > apologize if it isn't appropriate. I'm guessing this is something > simple > and I'm just clueless on how to diagnose the issue. > I'm not tied to using this model for good, but it is what we had > laying > around. Any help would be greatly appreciated. Next step is > configuring > it to talk to Verizon VOIP over a DS3. > > Thanks, > Matthew Kitchin > > > Matthew, > Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We > think you'll find FS is as powerful as any software out there right now. > > Here's a handy wiki page that will help you get the diagnosing skills > you need: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > I'd say first thing to do is capture the SIP traffic to see if there > are any clues. A "normal temporary failure" doesn't give you a lot of > detail. :) If you're new to SIP debugging then the best thing to do is > to capture the SIP trace and put it in the pastebin. > (http://pastebin.freeswitch.org) > > You can also join the IRC channel #freeswitch on irc.freenode.net > and get some real-time help. There are some > sharp folks in there, not the least of which are the three main > FreeSWITCH developers. > > -MC Thank you. I think I did what you are looking for. I stopped FS and launched this command. TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch and captured all output to http://pastebin.freeswitch.org/10965 Does this tell you anything? I'm definitely new to SIP and phone system admin in general. I have plenty of network and Linux experience. With that in mind, someone on this mailing list emailed me directly and said SipX would be a better fit for me. Is that blasphemy for me to even mention? I went through the documentation and the provisioning aspect and web interface do look tempting to a novice. I apologize if this is like trying to buy a chevy at a ford dealership. I'm looking to deploy about 150 handsets at a corporate office and then 10 to 12 handsets at 120 remote locations. We are moving from an old key system, so our current features are very limited. We just need a few ACD groups, call history, and the other general basics. I first found Asterisk and read about some of the shortcomings. FS looks like the most robust solution. I have no idea where SipX would fit in. The people here are obviously a very knowledgeable group and I would gladly accept any thoughts, comments, etc. From msc at freeswitch.org Tue Nov 3 20:39:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Nov 2009 20:39:14 -0800 Subject: [Freeswitch-users] Wiki typo In-Reply-To: References: <26432E7B-B9E7-4F4E-921F-E4B6C9AD4F6C@freeswitch.org> Message-ID: <87f2f3b90911032039j3a21c1f6p9bc068fa5391766d@mail.gmail.com> On Tue, Nov 3, 2009 at 7:45 PM, Dmitry Gromov wrote: > Thanks, done - page has been corrected! > > On Tue, Nov 3, 2009 at 22:34, Brian West wrote: > >> Yes you can login and edit the wiki yourself. >> >> >> > You know... I actually spent some time looking for login/create account > link when I noticed this typo. No idea why I did not see it then :) > > > Thank you for not giving up! :) We appreciate it when the community helps out. Nicely done. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091103/4478c6ef/attachment-0002.html From peter at cindyandpeter.com Tue Nov 3 21:37:59 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Wed, 4 Nov 2009 00:37:59 -0500 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <4AF0FEA6.7070308@gmail.com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> Message-ID: <025101ca5d10$f81228c0$e8367a40$@com> Matthew, I'm about in the same boat as you are, just on a smaller scale. We have a ton of Nortel telephony gear, but it's time to move out of the 90's and enter this millennium. My Cisco quote was in the same ballpark as yours. The Cisco stuff is mature, rock solid, meshes very well with their network gear and is actually relatively easy to set up and maintain if you know your way around IOS. I just refuse to pay that kind of money for yet another semi-proprietary solution. After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The developers and the community are great and available, but just starting out with SIP and voip in general, this may not be the best platform. So let the blasphemy begin :) SipX was a breeze to install (insert CD, boot, next next next...) and looks pretty solid. I believe they actually use FreeSWITCH for their voicemail and conferencing, internally. I just couldn't get my head around their GUI, ACD was too basic and had all kinds of issues getting stuff to "just work". 3CX (Windows Only) was completely painless. It just worked. But I'm still not convinced that I want to run all my voice on a single windows box. Plus it's not free/open/etc and I don't want to lock myself in again. Although it's an asterisk based solution, I found trixbox to be very easy. Setup is automatic and everything "just worked". The GUI is simple and logical enough that I can let somebody else handle the day-to-day phone setup and basic admin. I have my doubts about it scaling to 250 users, though. This may be a completely flawed strategy and I may very well be shooting myself in the foot by doing this, but I plan on piloting a trixbox install with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH box next to it for the more advanced stuff. Once I get more comfortable with the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH, I have a feeling that that trixbox is going to get phased out... Peter -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mkitchin.public at gmail.com Sent: Tuesday, November 03, 2009 11:10 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Michael Collins wrote: > > > On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public at gmail.com > > wrote: > > I'm working on an alternative to a $120,000 Cisco phone system that my > > company is looking at. I got Freeswitch installed on CentOS last week > using the Quick and Dirty instructions. That part was painless. We > had a > few 7940s laying around. After some wrestling with it, I got the > latest > SIP firmware installed and what I hoped was a functional config > (attached). X-Lite phones can call each other no problem. 7940s > can call > X-Lite no problem. Anytime I try and call a 7940, it goes straight to > voicemail. I attached a log file that shows the activity when > trying to > call a7940 from X-Lite. > X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is > nshplpbx1.unix/10.85.0.53 . Everything is on > the same LAN. Different > subnets, but no firewalls. > I didn't see anything that said posting attachments was frowned > upon. I > apologize if it isn't appropriate. I'm guessing this is something > simple > and I'm just clueless on how to diagnose the issue. > I'm not tied to using this model for good, but it is what we had > laying > around. Any help would be greatly appreciated. Next step is > configuring > it to talk to Verizon VOIP over a DS3. > > Thanks, > Matthew Kitchin > > > Matthew, > Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We > think you'll find FS is as powerful as any software out there right now. > > Here's a handy wiki page that will help you get the diagnosing skills > you need: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > I'd say first thing to do is capture the SIP traffic to see if there > are any clues. A "normal temporary failure" doesn't give you a lot of > detail. :) If you're new to SIP debugging then the best thing to do is > to capture the SIP trace and put it in the pastebin. > (http://pastebin.freeswitch.org) > > You can also join the IRC channel #freeswitch on irc.freenode.net > and get some real-time help. There are some > sharp folks in there, not the least of which are the three main > FreeSWITCH developers. > > -MC Thank you. I think I did what you are looking for. I stopped FS and launched this command. TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch and captured all output to http://pastebin.freeswitch.org/10965 Does this tell you anything? I'm definitely new to SIP and phone system admin in general. I have plenty of network and Linux experience. With that in mind, someone on this mailing list emailed me directly and said SipX would be a better fit for me. Is that blasphemy for me to even mention? I went through the documentation and the provisioning aspect and web interface do look tempting to a novice. I apologize if this is like trying to buy a chevy at a ford dealership. I'm looking to deploy about 150 handsets at a corporate office and then 10 to 12 handsets at 120 remote locations. We are moving from an old key system, so our current features are very limited. We just need a few ACD groups, call history, and the other general basics. I first found Asterisk and read about some of the shortcomings. FS looks like the most robust solution. I have no idea where SipX would fit in. The people here are obviously a very knowledgeable group and I would gladly accept any thoughts, comments, etc. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jason at jasonjgw.net Tue Nov 3 22:42:01 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 4 Nov 2009 17:42:01 +1100 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <025101ca5d10$f81228c0$e8367a40$@com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> <025101ca5d10$f81228c0$e8367a40$@com> Message-ID: <20091104064201.GA15804@jdc.jasonjgw.net> Peter J. Zandvoort wrote: > After looking at various asterisk distributions, SipX, 3CX and > what-have-you, I've come to the conclusion that FreeSWITCH is by far the > most advanced platform out there. Its architecture and performance is > literally light years ahead of the rest and I have yet to come up with > something that it can't do. But all that comes at a price: The learning > curve is like scaling a brick wall. The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki. From tculjaga at gmail.com Tue Nov 3 23:15:58 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 4 Nov 2009 08:15:58 +0100 Subject: [Freeswitch-users] Sipura Codec Problem In-Reply-To: <87f2f3b90911031616t2c731372i192f514d302522e9@mail.gmail.com> References: <24251951.post@talk.nabble.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <3A54C554-CD82-493B-8A8B-F9E1237B9963@freeswitch.org> <99E8604F-FC7B-41CF-A513-C9E9E6AC5E9A@gmail.com> <2d9149cd0911031111i6d2358a4ic80cab77a6836cc5@mail.gmail.com> <87f2f3b90911031616t2c731372i192f514d302522e9@mail.gmail.com> Message-ID: <65d96fc80911032315n4e5c5474kf68a60964a73320d@mail.gmail.com> just an off-topic question but it concenns mass provissioning ... does anyone know if there is an open TR069 platform we can work on? T. On Wed, Nov 4, 2009 at 1:16 AM, Michael Collins wrote: > > > On Tue, Nov 3, 2009 at 11:11 AM, Kristian Kielhofner < > kristian.kielhofner at gmail.com> wrote: > >> It appears that Tony has already added an option (amazing) BUT you >> should really be setup for central provisioning with an installed base >> that large... You'll eventually have issues that *NO* amount of >> Tony/FreeSWITCH magic can fix. >> >> Kristian is correct. Listen to him because he's familiar with having lots > and lots of units out in the field. The bandage Tony applied will eventually > wear off. The long-term solution is to treat the malady and not the symptom. > I'm certain that members of the FS community could point you toward some > resources to assist with central provisioning. > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/e0d0270e/attachment-0002.html From odermann at googlemail.com Wed Nov 4 00:23:01 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 4 Nov 2009 09:23:01 +0100 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com> <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> Message-ID: <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive? so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits. 2009/11/3 Anthony Minessale : > The patch was it's ability to accept subsequent invites. > Your problem is that in sip each new attempt to send an invite is another > call. > > 484 is a final response so the call with too few digits is terminated. From brian.stafford at lattice-voice.com Wed Nov 4 01:44:43 2009 From: brian.stafford at lattice-voice.com (Brian Stafford) Date: Wed, 04 Nov 2009 09:44:43 +0000 Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg of call In-Reply-To: <191c3a030911031223p23835d6ev4c3c3ddd98193f50@mail.gmail.com> References: <4AEB0A9C.7010907@lattice-voice.com> <4AEB1166.80002@lattice-voice.com> <4AEFF7B2.3080607@lattice-voice.com> <191c3a030911031223p23835d6ev4c3c3ddd98193f50@mail.gmail.com> Message-ID: <4AF14D0B.2060704@lattice-voice.com> Anthony Minessale wrote: > There are 2 ways to use the auto in > > one is to attended transfer the call into the extension with auto in > the other is to bind_meta_app a call to valet_park + auto in > > blind transfer to auto in only has one leg so the guy you transferred > is the only one who can hear it because when you press the blind xfer > key you hangup the call on your side. The penny drops - pretty obvious in hindsight. I've set it up with bind_meta_app and it's working very nicely now. Many thanks. Brian From sicfslist at gmail.com Wed Nov 4 04:08:06 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 04 Nov 2009 06:08:06 -0600 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <025101ca5d10$f81228c0$e8367a40$@com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> <025101ca5d10$f81228c0$e8367a40$@com> Message-ID: <4AF16EA6.7040708@gmail.com> Peter, Did you look at ? Probably just what you are looking for. GUI goodness based on FS. SDR Peter J. Zandvoort wrote: > Matthew, > > I'm about in the same boat as you are, just on a smaller scale. We have a > ton of Nortel telephony gear, but it's time to move out of the 90's and > enter this millennium. My Cisco quote was in the same ballpark as yours. > > The Cisco stuff is mature, rock solid, meshes very well with their network > gear and is actually relatively easy to set up and maintain if you know your > way around IOS. I just refuse to pay that kind of money for yet another > semi-proprietary solution. > > After looking at various asterisk distributions, SipX, 3CX and > what-have-you, I've come to the conclusion that FreeSWITCH is by far the > most advanced platform out there. Its architecture and performance is > literally light years ahead of the rest and I have yet to come up with > something that it can't do. But all that comes at a price: The learning > curve is like scaling a brick wall. The developers and the community are > great and available, but just starting out with SIP and voip in general, > this may not be the best platform. So let the blasphemy begin :) > > SipX was a breeze to install (insert CD, boot, next next next...) and looks > pretty solid. I believe they actually use FreeSWITCH for their voicemail and > conferencing, internally. I just couldn't get my head around their GUI, ACD > was too basic and had all kinds of issues getting stuff to "just work". > > 3CX (Windows Only) was completely painless. It just worked. But I'm still > not convinced that I want to run all my voice on a single windows box. Plus > it's not free/open/etc and I don't want to lock myself in again. > > Although it's an asterisk based solution, I found trixbox to be very easy. > Setup is automatic and everything "just worked". The GUI is simple and > logical enough that I can let somebody else handle the day-to-day phone > setup and basic admin. I have my doubts about it scaling to 250 users, > though. > > This may be a completely flawed strategy and I may very well be shooting > myself in the foot by doing this, but I plan on piloting a trixbox install > with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH > box next to it for the more advanced stuff. Once I get more comfortable with > the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH, > I have a feeling that that trixbox is going to get phased out... > > Peter > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > mkitchin.public at gmail.com > Sent: Tuesday, November 03, 2009 11:10 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones > > Michael Collins wrote: > >> On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public at gmail.com >> > > wrote: >> >> I'm working on an alternative to a $120,000 Cisco phone system that my >> >> company is looking at. I got Freeswitch installed on CentOS last week >> using the Quick and Dirty instructions. That part was painless. We >> had a >> few 7940s laying around. After some wrestling with it, I got the >> latest >> SIP firmware installed and what I hoped was a functional config >> (attached). X-Lite phones can call each other no problem. 7940s >> can call >> X-Lite no problem. Anytime I try and call a 7940, it goes straight to >> voicemail. I attached a log file that shows the activity when >> trying to >> call a7940 from X-Lite. >> X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is >> nshplpbx1.unix/10.85.0.53 . Everything is on >> the same LAN. Different >> subnets, but no firewalls. >> I didn't see anything that said posting attachments was frowned >> upon. I >> apologize if it isn't appropriate. I'm guessing this is something >> simple >> and I'm just clueless on how to diagnose the issue. >> I'm not tied to using this model for good, but it is what we had >> laying >> around. Any help would be greatly appreciated. Next step is >> configuring >> it to talk to Verizon VOIP over a DS3. >> >> Thanks, >> Matthew Kitchin >> >> >> Matthew, >> Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We >> think you'll find FS is as powerful as any software out there right now. >> >> Here's a handy wiki page that will help you get the diagnosing skills >> you need: >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> I'd say first thing to do is capture the SIP traffic to see if there >> are any clues. A "normal temporary failure" doesn't give you a lot of >> detail. :) If you're new to SIP debugging then the best thing to do is >> to capture the SIP trace and put it in the pastebin. >> (http://pastebin.freeswitch.org) >> >> You can also join the IRC channel #freeswitch on irc.freenode.net >> and get some real-time help. There are some >> sharp folks in there, not the least of which are the three main >> FreeSWITCH developers. >> >> -MC >> > Thank you. I think I did what you are looking for. I stopped FS and > launched this command. > TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch > and captured all output to http://pastebin.freeswitch.org/10965 > Does this tell you anything? > I'm definitely new to SIP and phone system admin in general. I have > plenty of network and Linux experience. With that in mind, someone on > this mailing list emailed me directly and said SipX would be a better > fit for me. Is that blasphemy for me to even mention? I went through the > documentation and the provisioning aspect and web interface do look > tempting to a novice. I apologize if this is like trying to buy a chevy > at a ford dealership. I'm looking to deploy about 150 handsets at a > corporate office and then 10 to 12 handsets at 120 remote locations. We > are moving from an old key system, so our current features are very > limited. We just need a few ACD groups, call history, and the other > general basics. I first found Asterisk and read about some of the > shortcomings. FS looks like the most robust solution. I have no idea > where SipX would fit in. The people here are obviously a very > knowledgeable group and I would gladly accept any thoughts, comments, etc. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Wed Nov 4 06:03:25 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Nov 2009 08:03:25 -0600 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com> <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> Message-ID: <9948F343-9230-42B8-AB83-C6DE0DA9886D@freeswitch.org> I'm going to say No! /b On Nov 4, 2009, at 2:23 AM, Dennis wrote: > is there a way to send something like 484 (or something else), which > does not make it a final answer and keep the call/socket alive? > > so we can ask the cirpack for further digits and decide what to do, if > the cirpack does not send any digits. From roy at net-vantage.com Tue Nov 3 21:41:05 2009 From: roy at net-vantage.com (RA Cohen) Date: Wed, 04 Nov 2009 00:41:05 -0500 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Message-ID: <4AF113F1.3090300@net-vantage.com> Here's what's in switch.conf.xml: Yet this message: SIP/2.0 503 Maximum Calls In Progress This is a small medical practice, 5-6 extensions, 3000 outbound minutes per month and at least the same inbound. We did fsctl shutdown restart and it flushed the sessions. What is going on? Thank you for your help! -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations" From carlos.talbot at gmail.com Wed Nov 4 06:51:37 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 4 Nov 2009 10:51:37 -0400 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> Message-ID: <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson wrote: > Jeff, > > thanks a lot for the reply. I was a little confused by the fact that the > "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried > that I might lose something. As you say though, think that I'll cross my > fingers and try the updated release. I am running FreeSwitch on a test > machine at the moment until the target hardware arrives - hopefully > tomorrow, so I can afford to have a little play. > I usually try to update the svn file at least once a month. I have a new version ready that was compiled last night but am ironing out login issues with the FS dudes for upload access. Also, the SVN snapshot now includes binaries for 32 and 64 bit. It no longer includes flite though as the install file was approaching 80MB in size. I will revisit this later if others feel it important to include flite. > > You mentioned FreePBX V3. I had been fumbling around trying to work out > what > this is and from what I've read, it seems to provide a GUI Front End for > configuring FreeSwitch ? > Yes, it's still in development phase and as such not ready for production use. > > I am guessing that while it has been installed with FreeSwitch, I then need > to run the FreePBX Installer to update the FreePBX/FreeSwitch configuration > on my hardware ? > > > When I start FreeSwitch, it does not automatically load the WAMPServer. > > Freeswitch and WAMPServer are independant of each other. WAMPServer is bundled in this install for the purpose of FreePBX as MySQL, Apache and PHP are all required components of FreePBX. When I start WAMPServer manually, and open up localhost (127.0.0.1) in a web > browser, I can see the WampServer logo and various tools such as phpinfo() > and phpmyadmin. FreePBX is there under Your Projects. > > If you want to configure FreePBX you need to click on the FreePBX.url shortcut that gets created on your desktop. > When I opened this up the first time, it appeared to want to install > FreePBX > over FreeSwitch, I tried to abort this when it was going to overwrite some > FreeSwitch conf files and I thought I'd better not go on until I had a > better idea what was happening. I backed out of the FreePBX install and now > I can't get the FreePBX or phpmyadmin pages up again (missing files) so it > looks like I'm going to have to reinstall anyway. > > So, for next time,am I right in thinking that I should proceed with running > the FreePBX install from the WAMPServer menu ? > No, launch it from the shortcut as stated above. Unfortunately, at this time there is very little user documentation on configuring FreePBX. Here is the link to the developer's info: http://www.freepbx.org/v3 regards, Carlos > > > ----- Original Message ----- > From: "Jeff Lenk" > To: > Sent: Tuesday, November 03, 2009 2:48 PM > Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > > > > > Hi Dave, > > > > These are supported by "Carlos Talbot" . They also include Freepbx v3 > > > > Just as you said freeswitch-1.0.4.exe is the tagged release and > > freeswitch.exe is a newer svn snapshot. > > > > There should be no problems installing the new version allthough best to > > just try and see! > > > > Not sure why the newest one is from October 7th. > > > > Jeff > > > > > > Dave Stevenson wrote: > >> > >> Hi, > >> > >> I have read the Docs on the Wiki > >> ( > http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) > >> but am still not sure of what the different Windows install files are. > >> Currently, the Windows Installer directory contains :- > >> > >> LATEST_SVN_15106 - 6 Bytes > >> > >> freeswitch-1.0.4.exe - 42 Megabytes > >> > >> freeswitch.exe - 32 Megabytes > >> > >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd > >> September. The freeswitch.exe file is dated 7th October and think that > it > >> contains the minor updates since 3rd September ? > >> > >> Could someone who knows FreeSwitch under windows help me understand the > >> two files please ? > >> > >> I chickened out of running the later exe in case it did something to the > >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe > with > >> the old one already installed ? > >> What will it actually do ? > >> > >> regards > >> Dave > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > -- > > View this message in context: > > http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/52173352/attachment-0002.html From diego.viola at gmail.com Wed Nov 4 07:25:55 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 4 Nov 2009 15:25:55 +0000 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress In-Reply-To: <4AF113F1.3090300@net-vantage.com> References: <4AF113F1.3090300@net-vantage.com> Message-ID: <86a32abc0911040725t621c287ax53b992fbcefd8691@mail.gmail.com> Hello, I tried to help Roy with this issue yesterday, I saw that calls couldn't go through and then I made a sofia profile internal siptrace on. Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and saw he had like 800 sessions. I thought it was an ACL issue but it wasn't, it seems like he reached a session limit, when I restarted his FS the problem went away. Best Regards, Diego On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen wrote: > Here's what's in switch.conf.xml: > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > per month and at least the same inbound. We did fsctl shutdown restart > and it flushed the sessions. What is going on? > > Thank you for your help! > > -- > Roy A Cohen > Network Advantage LLC > www.net-vantage.com > 413.223.9007 option 1 > -------------------------------------------------- > "Bringing Cost-Saving, State-of-the-Art Technology > Solutions to Small and Mid-Size Organizations" > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/a737b2ef/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 4 07:39:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Nov 2009 09:39:21 -0600 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> Message-ID: <191c3a030911040739k5e24a2ferdb56a3419196b581@mail.gmail.com> You cannot. This is how the sip spec works. Every new invite is a new call and a new trip to the dialplan. You will probably need to design your code to send the appropriate 484 and be prepared to exit and be called again with the new digits. On Wed, Nov 4, 2009 at 2:23 AM, Dennis wrote: > is there a way to send something like 484 (or something else), which > does not make it a final answer and keep the call/socket alive? > > so we can ask the cirpack for further digits and decide what to do, if > the cirpack does not send any digits. > > > > 2009/11/3 Anthony Minessale : > > The patch was it's ability to accept subsequent invites. > > Your problem is that in sip each new attempt to send an invite is another > > call. > > > > 484 is a final response so the call with too few digits is terminated. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/3c245f8f/attachment-0002.html From mike at jerris.com Wed Nov 4 07:43:10 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 4 Nov 2009 10:43:10 -0500 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress In-Reply-To: <86a32abc0911040725t621c287ax53b992fbcefd8691@mail.gmail.com> References: <4AF113F1.3090300@net-vantage.com> <86a32abc0911040725t621c287ax53b992fbcefd8691@mail.gmail.com> Message-ID: Call loop? On Nov 4, 2009, at 10:25 AM, Diego Viola wrote: > Hello, > > I tried to help Roy with this issue yesterday, I saw that calls > couldn't go through and then I made a sofia profile internal > siptrace on. > > Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" > and saw he had like 800 sessions. > > I thought it was an ACL issue but it wasn't, it seems like he > reached a session limit, when I restarted his FS the problem went > away. > > Best Regards, > > Diego From jlenk at frontiernet.net Wed Nov 4 07:43:54 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Wed, 4 Nov 2009 07:43:54 -0800 (PST) Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> Message-ID: <1257349434463-3946039.post@n2.nabble.com> Hi Carlos, very cool that the x64 version is included now! Hopefully this will get more people using the x64 version under Windows! Regards, Jeff Carlos Talbot wrote: > > On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson > wrote: > >> Jeff, >> >> thanks a lot for the reply. I was a little confused by the fact that the >> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >> that I might lose something. As you say though, think that I'll cross my >> fingers and try the updated release. I am running FreeSwitch on a test >> machine at the moment until the target hardware arrives - hopefully >> tomorrow, so I can afford to have a little play. >> > > I usually try to update the svn file at least once a month. I have a new > version ready that was compiled last night but am ironing out login issues > with the FS dudes for upload access. Also, the SVN snapshot now includes > binaries for 32 and 64 bit. It no longer includes flite though as the > install file was approaching 80MB in size. I will revisit this later if > others feel it important to include flite. > >> >> You mentioned FreePBX V3. I had been fumbling around trying to work out >> what >> this is and from what I've read, it seems to provide a GUI Front End for >> configuring FreeSwitch ? >> > Yes, it's still in development phase and as such not ready for production > use. > >> >> I am guessing that while it has been installed with FreeSwitch, I then >> need >> to run the FreePBX Installer to update the FreePBX/FreeSwitch >> configuration >> on my hardware ? >> >> >> When I start FreeSwitch, it does not automatically load the WAMPServer. >> >> Freeswitch and WAMPServer are independant of each other. WAMPServer is > bundled in this install for the purpose of FreePBX as MySQL, Apache and > PHP > are all required components of FreePBX. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a > web >> browser, I can see the WampServer logo and various tools such as >> phpinfo() >> and phpmyadmin. FreePBX is there under Your Projects. >> >> If you want to configure FreePBX you need to click on the FreePBX.url > shortcut that gets created on your desktop. > > >> When I opened this up the first time, it appeared to want to install >> FreePBX >> over FreeSwitch, I tried to abort this when it was going to overwrite >> some >> FreeSwitch conf files and I thought I'd better not go on until I had a >> better idea what was happening. I backed out of the FreePBX install and >> now >> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >> it >> looks like I'm going to have to reinstall anyway. >> >> So, for next time,am I right in thinking that I should proceed with >> running >> the FreePBX install from the WAMPServer menu ? >> > > No, launch it from the shortcut as stated above. Unfortunately, at this > time > there is very little user documentation on configuring FreePBX. Here is > the > link to the developer's info: http://www.freepbx.org/v3 > > regards, > > Carlos > >> >> >> ----- Original Message ----- >> From: "Jeff Lenk" >> To: >> Sent: Tuesday, November 03, 2009 2:48 PM >> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >> >> >> > >> > Hi Dave, >> > >> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >> > >> > Just as you said freeswitch-1.0.4.exe is the tagged release and >> > freeswitch.exe is a newer svn snapshot. >> > >> > There should be no problems installing the new version allthough best >> to >> > just try and see! >> > >> > Not sure why the newest one is from October 7th. >> > >> > Jeff >> > >> > >> > Dave Stevenson wrote: >> >> >> >> Hi, >> >> >> >> I have read the Docs on the Wiki >> >> ( >> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> >> but am still not sure of what the different Windows install files are. >> >> Currently, the Windows Installer directory contains :- >> >> >> >> LATEST_SVN_15106 - 6 Bytes >> >> >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> >> >> freeswitch.exe - 32 Megabytes >> >> >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> >> September. The freeswitch.exe file is dated 7th October and think that >> it >> >> contains the minor updates since 3rd September ? >> >> >> >> Could someone who knows FreeSwitch under windows help me understand >> the >> >> two files please ? >> >> >> >> I chickened out of running the later exe in case it did something to >> the >> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >> with >> >> the old one already installed ? >> >> What will it actually do ? >> >> >> >> regards >> >> Dave >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > -- >> > View this message in context: >> > >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html Sent from the freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Wed Nov 4 07:44:08 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 4 Nov 2009 16:44:08 +0100 Subject: [Freeswitch-users] SIP Overlap support? In-Reply-To: <9948F343-9230-42B8-AB83-C6DE0DA9886D@freeswitch.org> References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com> <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com> <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com> <4AEC5C65.6050800@puzzled.xs4all.nl> <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org> <5e414ed0911030757p11110b6bmb64e88070796aad3@mail.gmail.com> <191c3a030911030919n7f125890qf169b2f484ce721@mail.gmail.com> <5e414ed0911040023m4a4e25e1le33a7d1dc8cc52c1@mail.gmail.com> <9948F343-9230-42B8-AB83-C6DE0DA9886D@freeswitch.org> Message-ID: <65d96fc80911040744v3bc8c604w557ebbf68ebaddf4@mail.gmail.com> Brian is right, pls, lets stop with exceptions and get stick to RFCs... otherwise it will be a big mess ... T. On Wed, Nov 4, 2009 at 3:03 PM, Brian West wrote: > I'm going to say No! > > /b > > On Nov 4, 2009, at 2:23 AM, Dennis wrote: > > > is there a way to send something like 484 (or something else), which > > does not make it a final answer and keep the call/socket alive? > > > > so we can ask the cirpack for further digits and decide what to do, if > > the cirpack does not send any digits. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/6109c3c0/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 4 07:46:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Nov 2009 09:46:11 -0600 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress In-Reply-To: <4AF113F1.3090300@net-vantage.com> References: <4AF113F1.3090300@net-vantage.com> Message-ID: <191c3a030911040746g516f8579u5f641c30c7a54b8a@mail.gmail.com> Which revision of FreeSWITCH are you using? On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen wrote: > Here's what's in switch.conf.xml: > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > per month and at least the same inbound. We did fsctl shutdown restart > and it flushed the sessions. What is going on? > > Thank you for your help! > > -- > Roy A Cohen > Network Advantage LLC > www.net-vantage.com > 413.223.9007 option 1 > -------------------------------------------------- > "Bringing Cost-Saving, State-of-the-Art Technology > Solutions to Small and Mid-Size Organizations" > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/dfb1a0e7/attachment-0002.html From roy at net-vantage.com Wed Nov 4 08:00:37 2009 From: roy at net-vantage.com (RA Cohen) Date: Wed, 04 Nov 2009 11:00:37 -0500 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Message-ID: <4AF1A525.5090909@net-vantage.com> FreeSWITCH Version 1.0.trunk (15321) -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations" From peter at cindyandpeter.com Wed Nov 4 08:33:57 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Wed, 4 Nov 2009 11:33:57 -0500 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <20091104064201.GA15804@jdc.jasonjgw.net> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> <025101ca5d10$f81228c0$e8367a40$@com> <20091104064201.GA15804@jdc.jasonjgw.net> Message-ID: <027f01ca5d6c$9c1603a0$d4420ae0$@com> Absolutely agreed. To use Matthew's original car metaphor: When you just got your learner's permit, the old Chevy may be a better choice than the Ferrari. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, November 04, 2009 1:42 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Peter J. Zandvoort wrote: > After looking at various asterisk distributions, SipX, 3CX and > what-have-you, I've come to the conclusion that FreeSWITCH is by far the > most advanced platform out there. Its architecture and performance is > literally light years ahead of the rest and I have yet to come up with > something that it can't do. But all that comes at a price: The learning > curve is like scaling a brick wall. The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jbarou at sqli.com Wed Nov 4 08:56:29 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Wed, 4 Nov 2009 17:56:29 +0100 Subject: [Freeswitch-users] Question about jingle_profiles Message-ID: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> Hi everybody, I actually working on mod_dingaling (gtalk). I can make call from FS to Gtalk, and from Gtalk to FS. But I have a problem, in jingle_profile I have a file like this : here when I put an user account like john or bob its doesn't work whereas I put something like 1000 or 8400 it works. When I tried to put a real phone number It doesn't work too (I have a gateway with my PBX). Somebody know, why it doesn't work with name and work with number ? Thanks. -- Jonathan BAROU SQLI LYON - CRCI 0472405368 jbarou at sqli.com lyon.crci at sqli.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/6d60edb4/attachment-0002.html From qinglan_zeng at hotmail.com Wed Nov 4 08:54:09 2009 From: qinglan_zeng at hotmail.com (=?gb2312?B?tPPE4MjL?=) Date: Wed, 4 Nov 2009 16:54:09 +0000 Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: Hello All, Newbie to FS and I installed FS using Windows precompiled binaries. I want to set up some skype trunks with FS and so I followed the instructions while get some errors: (1). Launch Skype by clicking the skype.exe. (2). Launch FS (3) In FS I enter the cmd as: load mod_skypiax and then come to error: module load file routine retured an error. I had saved a screenshot for your referrence. Any idea on this? Thanks Daniel Zeng From: freeswitch-users-request at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 41, Issue 27 To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 07:46:45 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --??????-- From: diego.viola at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 15:25:55 +0000 Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Hello, I tried to help Roy with this issue yesterday, I saw that calls couldn't go through and then I made a sofia profile internal siptrace on. Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and saw he had like 800 sessions. I thought it was an ACL issue but it wasn't, it seems like he reached a session limit, when I restarted his FS the problem went away. Best Regards, Diego On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen wrote: Here's what's in switch.conf.xml: Yet this message: SIP/2.0 503 Maximum Calls In Progress This is a small medical practice, 5-6 extensions, 3000 outbound minutes per month and at least the same inbound. We did fsctl shutdown restart and it flushed the sessions. What is going on? Thank you for your help! -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations" _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --??????-- From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 09:39:21 -0600 Subject: Re: [Freeswitch-users] SIP Overlap support? You cannot. This is how the sip spec works. Every new invite is a new call and a new trip to the dialplan. You will probably need to design your code to send the appropriate 484 and be prepared to exit and be called again with the new digits. On Wed, Nov 4, 2009 at 2:23 AM, Dennis wrote: is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive? so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits. 2009/11/3 Anthony Minessale : > The patch was it's ability to accept subsequent invites. > Your problem is that in sip each new attempt to send an invite is another > call. > > 484 is a final response so the call with too few digits is terminated. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 --??????-- From: mike at jerris.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 10:43:10 -0500 Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Call loop? On Nov 4, 2009, at 10:25 AM, Diego Viola wrote: > Hello, > > I tried to help Roy with this issue yesterday, I saw that calls > couldn't go through and then I made a sofia profile internal > siptrace on. > > Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" > and saw he had like 800 sessions. > > I thought it was an ACL issue but it wasn't, it seems like he > reached a session limit, when I restarted his FS the problem went > away. > > Best Regards, > > Diego --??????-- From: jlenk at frontiernet.net To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 07:43:54 -0800 Subject: Re: [Freeswitch-users] Precompiled Windows Binaries Hi Carlos, very cool that the x64 version is included now! Hopefully this will get more people using the x64 version under Windows! Regards, Jeff Carlos Talbot wrote: > > On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson > wrote: > >> Jeff, >> >> thanks a lot for the reply. I was a little confused by the fact that the >> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >> that I might lose something. As you say though, think that I'll cross my >> fingers and try the updated release. I am running FreeSwitch on a test >> machine at the moment until the target hardware arrives - hopefully >> tomorrow, so I can afford to have a little play. >> > > I usually try to update the svn file at least once a month. I have a new > version ready that was compiled last night but am ironing out login issues > with the FS dudes for upload access. Also, the SVN snapshot now includes > binaries for 32 and 64 bit. It no longer includes flite though as the > install file was approaching 80MB in size. I will revisit this later if > others feel it important to include flite. > >> >> You mentioned FreePBX V3. I had been fumbling around trying to work out >> what >> this is and from what I've read, it seems to provide a GUI Front End for >> configuring FreeSwitch ? >> > Yes, it's still in development phase and as such not ready for production > use. > >> >> I am guessing that while it has been installed with FreeSwitch, I then >> need >> to run the FreePBX Installer to update the FreePBX/FreeSwitch >> configuration >> on my hardware ? >> >> >> When I start FreeSwitch, it does not automatically load the WAMPServer. >> >> Freeswitch and WAMPServer are independant of each other. WAMPServer is > bundled in this install for the purpose of FreePBX as MySQL, Apache and > PHP > are all required components of FreePBX. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a > web >> browser, I can see the WampServer logo and various tools such as >> phpinfo() >> and phpmyadmin. FreePBX is there under Your Projects. >> >> If you want to configure FreePBX you need to click on the FreePBX.url > shortcut that gets created on your desktop. > > >> When I opened this up the first time, it appeared to want to install >> FreePBX >> over FreeSwitch, I tried to abort this when it was going to overwrite >> some >> FreeSwitch conf files and I thought I'd better not go on until I had a >> better idea what was happening. I backed out of the FreePBX install and >> now >> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >> it >> looks like I'm going to have to reinstall anyway. >> >> So, for next time,am I right in thinking that I should proceed with >> running >> the FreePBX install from the WAMPServer menu ? >> > > No, launch it from the shortcut as stated above. Unfortunately, at this > time > there is very little user documentation on configuring FreePBX. Here is > the > link to the developer's info: http://www.freepbx.org/v3 > > regards, > > Carlos > >> >> >> ----- Original Message ----- >> From: "Jeff Lenk" >> To: >> Sent: Tuesday, November 03, 2009 2:48 PM >> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >> >> >> > >> > Hi Dave, >> > >> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >> > >> > Just as you said freeswitch-1.0.4.exe is the tagged release and >> > freeswitch.exe is a newer svn snapshot. >> > >> > There should be no problems installing the new version allthough best >> to >> > just try and see! >> > >> > Not sure why the newest one is from October 7th. >> > >> > Jeff >> > >> > >> > Dave Stevenson wrote: >> >> >> >> Hi, >> >> >> >> I have read the Docs on the Wiki >> >> ( >> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> >> but am still not sure of what the different Windows install files are. >> >> Currently, the Windows Installer directory contains :- >> >> >> >> LATEST_SVN_15106 - 6 Bytes >> >> >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> >> >> freeswitch.exe - 32 Megabytes >> >> >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> >> September. The freeswitch.exe file is dated 7th October and think that >> it >> >> contains the minor updates since 3rd September ? >> >> >> >> Could someone who knows FreeSwitch under windows help me understand >> the >> >> two files please ? >> >> >> >> I chickened out of running the later exe in case it did something to >> the >> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >> with >> >> the old one already installed ? >> >> What will it actually do ? >> >> >> >> regards >> >> Dave >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > -- >> > View this message in context: >> > >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html Sent from the freeswitch-users mailing list archive at Nabble.com. --??????-- From: tculjaga at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 16:44:08 +0100 Subject: Re: [Freeswitch-users] SIP Overlap support? Brian is right, pls, lets stop with exceptions and get stick to RFCs... otherwise it will be a big mess ... T. On Wed, Nov 4, 2009 at 3:03 PM, Brian West wrote: I'm going to say No! /b On Nov 4, 2009, at 2:23 AM, Dennis wrote: > is there a way to send something like 484 (or something else), which > does not make it a final answer and keep the call/socket alive? > > so we can ask the cirpack for further digits and decide what to do, if > the cirpack does not send any digits. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --??????-- From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 4 Nov 2009 09:46:11 -0600 Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress Which revision of FreeSWITCH are you using? On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen wrote: Here's what's in switch.conf.xml: Yet this message: SIP/2.0 503 Maximum Calls In Progress This is a small medical practice, 5-6 extensions, 3000 outbound minutes per month and at least the same inbound. We did fsctl shutdown restart and it flushed the sessions. What is going on? Thank you for your help! -- Roy A Cohen Network Advantage LLC www.net-vantage.com 413.223.9007 option 1 -------------------------------------------------- "Bringing Cost-Saving, State-of-the-Art Technology Solutions to Small and Mid-Size Organizations" _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _________________________________________________________________ ?????????????????msn????? http://ditu.live.com/?form=TL&swm=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/4556646d/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: skypiax load error.doc Type: application/msword Size: 113152 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/4556646d/attachment-0002.doc From gmaruzz at celliax.org Wed Nov 4 09:18:47 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 4 Nov 2009 18:18:47 +0100 Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: <7b197bef0911040918u6cfd7f46s157e90fceacd5ddd@mail.gmail.com> 2009/11/4 ??? : > Newbie to FS and I installed FS using Windows precompiled binaries. I want > to set up some skype trunks with FS and so I followed the instructions while > get some errors: > (1). Launch Skype by clicking the skype.exe. > (2). Launch FS > (3) In FS I enter the cmd as: load mod_skypiax and then come to error: > module load file routine retured an error. I had saved a screenshot for your > referrence. > Please Daniel, do not send mail both to me personally and to the mailing list. Send only to the mailing list. As you can see in the screenshot you attach, mod_skypiax cannot find its configuration file. For what I can understand, you have not the skills needed for administering FS, so it would be better if you find someone (a friend, etc) that can help you. -giovanni > Any idea on this? > > Thanks > Daniel Zeng > > From: freeswitch-users-request at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 41, Issue 27 > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 07:46:45 -0800 > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > --??????-- > From: diego.viola at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date : Wed, 4 Nov 2009 15:25:55 +0000 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Hello, > > I tried to help Roy with this issue yesterday, I saw that calls couldn't go > through and then I made a sofia profile internal siptrace on. > > Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and saw > he had like 800 sessions. > > > I thought it was an ACL issue but it wasn't, it seems like he reached a > session limit, when I restarted his FS the problem went away. > > Best Regards, > > Diego > > On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen wrote: > > Here's what's in switch.conf.xml: > > > > > > > > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > > per month and at least the same inbound. We did fsctl shutdown restart > > and it flushed the sessions. What is going on? > > > > Thank you for your help! > > > > -- > > Roy A Cohen > > Network Advantage LLC > > www.net-vantage.com > > 413.223.9007 option 1 > > -------------------------------------------------- > > "Bringing Cost-Saving, State-of-the-Art Technology > > Solutions to Small and M id-Size Organizations" > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > --??????-- > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 09:39:21 -0600 > Subject: Re: [Freeswitch-users] SIP Overlap support? > > You cannot. > This is how the sip spec works. > Every new invite is a new call and a new trip to the dialplan. > > You will probabl y need to design your code to send the appropriate 484 and > be prepared to exit and be called again with the new digits. > > > > On Wed, Nov 4, 2009 at 2:23 AM, Dennis wrote: > > is there a way to send something like 484 (or something else), which > > does not make it a final answer and keep the call/socket alive? > > > > so we can ask the cirpack for further digits and decide what to do, if > > the cirpack does not send any digits. > > > > > > > > 2009/11/3 Anthony Minessale : > >> The patch was it's ability to accept subsequent invites. > >> Your problem is that in sip each new attempt to send an invite is another > >> call. > >> > >> 484 is a final response so the call with too few digits is terminated. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > --??????-- > From: mike at jerris.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 10:43:10 -0500 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Call loop? > > On Nov 4, 2009, at 10:25 AM, Diego Viola wrote: > >> Hello, >> >> I tried to help Roy with this issue yesterday, I saw that ca > lls >> couldn't go through and then I made a sofia profile internal >> siptrace on. >> >> Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" >> and saw he had like 800 sessions. >> >> I thought it was an ACL issue but it wasn't, it seems like he >> reached a session limit, when I restarted his FS the problem went >> away. >> >> Best Regards, >> >> Diego > > > > > > --??????-- > From: jlenk at frontiernet.net > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 07:43:54 -0800 > Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > > Hi Carlos, > > very cool that the x64 version is included now! Hopefully this will get more > people using the x64 version under Windows! > > Regards, > Jeff > > > Carlos Talbot wrote: >> >> On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson >> ebank.net>wrote: >> >>> Jeff, >>> >>> thanks a lot for the reply. I was a little confused by the fact that the >>> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >>> that I might lose something. As you say though, think that I'll cross my >>> fingers and try the updated release. I am running FreeSwitch on a test >>> machine at the moment until the target hardware arrives - hopefully >>> tomorrow, so I can afford to have a little play. >>> >> >> I usually try to update the svn file at least once a month. I have a new >> version ready that was compiled last night but am ironing out login issues >> with the FS dudes for upload access. Also, the SVN snapshot now includes >> binaries for 32 and 64 bit. It no longer includes flite though as the >> install file was approaching 80MB in size. I will revisit this later if >> others feel it impo > rtant to include flite. >> >>> >>> You mentioned FreePBX V3. I had been fumbling around trying to work out >>> what >>> this is and from what I've read, it seems to provide a GUI Front End for >>> configuring FreeSwitch ? >>> >> Yes, it's still in development phase and as such not ready for production >> use. >> >>> >>> I am guessing that while it has been installed with FreeSwitch, I then >>> need >>> to run the FreePBX Installer to update the FreePBX/FreeSwitch >>> configuration >>> on my hardware ? >>> >>> >>> When I start FreeSwitch, it does not automatically load the WAMPServer. >>> >>> Freeswitch and WAMPServer are independant of each other. WAMPServer is >> bundled in this install for the purpose of FreePBX as MySQL, Apache and >> PHP >> are all required components of FreePBX. >> >> When > I start WAMPServer manually, and open up localhost (127.0.0.1) in a >> web >>> browser, I can see the WampServer logo and various tools such as >>> phpinfo() >>> and phpmyadmin. FreePBX is there under Your Projects. >>> >>> If you want to configure FreePBX you need to click on the FreePBX.url >> shortcut that gets created on your desktop. >> >> >>> When I opened this up the first time, it appeared to want to install >>> FreePBX >>> over FreeSwitch, I tried to abort this when it was going to overwrite >>> some >>> FreeSwitch conf files and I thought I'd better not go on until I had a >>> better idea what was happening. I backed out of the FreePBX install and >>> now >>> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >>> it >>> looks like I'm going to have to reinstall anyway. >>> >>> So, for nex > t time,am I right in thinking that I should proceed with >>> running >>> the FreePBX install from the WAMPServer menu ? >>> >> >> No, launch it from the shortcut as stated above. Unfortunately, at this >> time >> there is very little user documentation on configuring FreePBX. Here is >> the >> link to the developer's info: http://www.freepbx.org/v3 >> >> regards, >> >> Carlos >> >>> >>> >>> ----- Original Message ----- >>> From: "Jeff Lenk" >>> To: >>> Sent: Tuesday, November 03, 2009 2:48 PM >>> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >>> >>> >>> > >>> > Hi Dave, >>> > >>> > These are supported by "Carlos Talbot" . They also include Freepbx v3< > BR>>> > >>> > Just as you said freeswitch-1.0.4.exe is the tagged release and >>> > freeswitch.exe is a newer svn snapshot. >>> > >>> > There should be no problems installing the new version allthough best >>> to >>> > just try and see! >>> > >>> > Not sure why the newest one is from October 7th. >>> > >>> > Jeff >>> > >>> > >>> > Dave Stevenson wrote: >>> >> >>> >> Hi, >>> >> >>> >> I have read the Docs on the Wiki >>> >> ( >>> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >>> >> but am still not sure of what the different Windows install files are. >>> >> Currently, the Windows Installer directory contains :- >>> >> >>> >> LATEST_SVN_15106 - 6 Bytes >>> >> >>> >> freeswitch-1.0.4.exe - 42 Megabytes >>> >> >>> >> freeswitch.exe - 32 Megabytes >>> >> >>> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >>> >> September. The freeswitch.exe file is dated 7th October and think that >>> it >>> >> contains the minor updates since 3rd September ? >>> >> >>> >> Could someone who knows FreeSwitch under windows help me understand >>> the >>> >> two files please ? >>> >> >>> >> I chickened out of running the later exe in case it did something to >>> the >>> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >>> with >>> >> the old one already installed ? >>> >> What will it > actually do ? >>> >> >>> >> regards >>> >> Dave >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> > >>> > -- >>> > View this message in context: >>> > >>> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3 > 937943p3938887.html >>> > Sent from the freeswitch-users mailing list archive at Nabble.com. >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http > ://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http > ://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > --??????-- > From: tculjaga at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 16:44:08 +0100 > Subject: Re: [Freeswitch-users] SIP Overlap support? > > Brian is right, > > pls, lets stop with exceptions and get stick to RFCs... otherwise it will be > a big mess ... > > T. > > On Wed, Nov 4, 2009 at 3:03 PM, Brian West wrote: > > I'm going to say No! > > > > /b > > > > On Nov 4, 2009, at 2:23 AM, Dennis wrote: > > > >> is there a way to send something like 484 (or something else), which > >> does not make it a final answer and keep the call/socket alive? > >> > >> so we can ask the cirpack for further digits and decide what to do, if > >> the cirpack does not send any digits. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > --??????-- > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 09:46:11 -0600 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Which revision of FreeSWITCH are you using? > > > On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen wrote: > > Here's what's in switch.conf.xml: > > > > > > > > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > > per month and at least the same inbound. We did fsctl shutdown restart > > and it flushed the sessions. What is going on? > > > > Thank you for your help! > > > > -- > > Roy A Cohen > > Network Advantage LLC > > www.net-vantage.com > > 413.223.9007 option 1 > > -------------------------------------------------- > > "Bringing Cost-Saving, State-of-the-Art Technology > > Solutions to Small and M id-Size Organizations" > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > ________________________________ > ??????????MSN??? ????? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at celliax.org Wed Nov 4 09:22:29 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 4 Nov 2009 18:22:29 +0100 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <1257349434463-3946039.post@n2.nabble.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <1257349434463-3946039.post@n2.nabble.com> Message-ID: <7b197bef0911040922y21aa10c0l53909b9f3ea07c5@mail.gmail.com> On Wed, Nov 4, 2009 at 4:43 PM, Jeff Lenk wrote: > > very cool that the x64 version is included now! Hopefully this will get more > people using the x64 version under Windows! ...and will get more people using the x64 version of Windows! ;) -gm > > Regards, > Jeff > > > Carlos Talbot wrote: >> >> On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson >> wrote: >> >>> Jeff, >>> >>> thanks a lot for the reply. I was a little confused by the fact that the >>> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >>> that I might lose something. As you say though, think that I'll cross my >>> fingers and try the updated release. I am running FreeSwitch on a test >>> machine at the moment until the target hardware arrives - hopefully >>> tomorrow, so I can afford to have a little play. >>> >> >> I usually try to update the svn file at least once a month. I have a new >> version ready that was compiled last night but am ironing out login issues >> with the FS dudes for upload access. Also, the SVN snapshot now includes >> binaries for 32 and 64 bit. It no longer includes flite though as the >> install file was approaching 80MB in size. I will revisit this later if >> others feel it important to include flite. >> >>> >>> You mentioned FreePBX V3. I had been fumbling around trying to work out >>> what >>> this is and from what I've read, it seems to provide a GUI Front End for >>> configuring FreeSwitch ? >>> >> Yes, it's still in development phase and as such not ready for production >> use. >> >>> >>> I am guessing that while it has been installed with FreeSwitch, I then >>> need >>> to run the FreePBX Installer to update the FreePBX/FreeSwitch >>> configuration >>> on my hardware ? >>> >>> >>> When I start FreeSwitch, it does not automatically load the WAMPServer. >>> >>> Freeswitch and WAMPServer are independant of each other. WAMPServer is >> bundled in this install for the purpose of FreePBX as MySQL, Apache and >> PHP >> are all required components of FreePBX. >> >> When I start WAMPServer manually, and open up localhost (127.0.0.1) in a >> web >>> browser, I can see the WampServer logo and various tools such as >>> phpinfo() >>> and phpmyadmin. FreePBX is there under Your Projects. >>> >>> If you want to configure FreePBX you need to click on the FreePBX.url >> shortcut that gets created on your desktop. >> >> >>> When I opened this up the first time, it appeared to want to install >>> FreePBX >>> over FreeSwitch, I tried to abort this when it was going to overwrite >>> some >>> FreeSwitch conf files and I thought I'd better not go on until I had a >>> better idea what was happening. I backed out of the FreePBX install and >>> now >>> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >>> it >>> looks like I'm going to have to reinstall anyway. >>> >>> So, for next time,am I right in thinking that I should proceed with >>> running >>> the FreePBX install from the WAMPServer menu ? >>> >> >> No, launch it from the shortcut as stated above. Unfortunately, at this >> time >> there is very little user documentation on configuring FreePBX. Here is >> the >> link to the developer's info: http://www.freepbx.org/v3 >> >> regards, >> >> Carlos >> >>> >>> >>> ----- Original Message ----- >>> From: "Jeff Lenk" >>> To: >>> Sent: Tuesday, November 03, 2009 2:48 PM >>> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >>> >>> >>> > >>> > Hi Dave, >>> > >>> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >>> > >>> > Just as you said freeswitch-1.0.4.exe is the tagged release and >>> > freeswitch.exe is a newer svn snapshot. >>> > >>> > There should be no problems installing the new version allthough best >>> to >>> > just try and see! >>> > >>> > Not sure why the newest one is from October 7th. >>> > >>> > Jeff >>> > >>> > >>> > Dave Stevenson wrote: >>> >> >>> >> Hi, >>> >> >>> >> I have read the Docs on the Wiki >>> >> ( >>> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >>> >> but am still not sure of what the different Windows install files are. >>> >> Currently, the Windows Installer directory contains :- >>> >> >>> >> LATEST_SVN_15106 - 6 Bytes >>> >> >>> >> freeswitch-1.0.4.exe - 42 Megabytes >>> >> >>> >> freeswitch.exe - 32 Megabytes >>> >> >>> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >>> >> September. The freeswitch.exe file is dated 7th October and think that >>> it >>> >> contains the minor updates since 3rd September ? >>> >> >>> >> Could someone who knows FreeSwitch under windows help me understand >>> the >>> >> two files please ? >>> >> >>> >> I chickened out of running the later exe in case it did something to >>> the >>> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >>> with >>> >> the old one already installed ? >>> >> What will it actually do ? >>> >> >>> >> regards >>> >> Dave >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> > >>> > -- >>> > View this message in context: >>> > >>> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >>> > Sent from the freeswitch-users mailing list archive at Nabble.com. >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From jlenk at frontiernet.net Wed Nov 4 09:28:07 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Wed, 4 Nov 2009 09:28:07 -0800 (PST) Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: <1257355687424-3946759.post@n2.nabble.com> follow these intructions -> http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances ??? wrote: > > > Hello All, > > > > Newbie to FS and I installed FS using Windows precompiled binaries. I want > to set up some skype trunks with FS and so I followed the instructions > while get some errors: > > (1). Launch Skype by clicking the skype.exe. > > (2). Launch FS > > (3) In FS I enter the cmd as: load mod_skypiax and then come to error: > module load file routine retured an error. I had saved a screenshot for > your referrence. > > > > Any idea on this? > > > > Thanks > > Daniel Zeng > > From: freeswitch-users-request at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 41, Issue 27 > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 07:46:45 -0800 > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > --??????-- > From: diego.viola at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 15:25:55 +0000 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Hello, > > I tried to help Roy with this issue yesterday, I saw that calls couldn't > go through and then I made a sofia profile internal siptrace on. > > Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and > saw he had like 800 sessions. > > > I thought it was an ACL issue but it wasn't, it seems like he reached a > session limit, when I restarted his FS the problem went away. > > Best Regards, > > Diego > > > On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen wrote: > > > Here's what's in switch.conf.xml: > > > > > > > > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > > per month and at least the same inbound. We did fsctl shutdown restart > > and it flushed the sessions. What is going on? > > > > Thank you for your help! > > > > -- > > Roy A Cohen > > Network Advantage LLC > > www.net-vantage.com > > 413.223.9007 option 1 > > -------------------------------------------------- > > "Bringing Cost-Saving, State-of-the-Art Technology > > Solutions to Small and Mid-Size Organizations" > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > --??????-- > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 09:39:21 -0600 > Subject: Re: [Freeswitch-users] SIP Overlap support? > > You cannot. > This is how the sip spec works. > Every new invite is a new call and a new trip to the dialplan. > > You will probably need to design your code to send the appropriate 484 and > be prepared to exit and be called again with the new digits. > > > > > On Wed, Nov 4, 2009 at 2:23 AM, Dennis wrote: > > > is there a way to send something like 484 (or something else), which > > does not make it a final answer and keep the call/socket alive? > > > > so we can ask the cirpack for further digits and decide what to do, if > > the cirpack does not send any digits. > > > > > > > > 2009/11/3 Anthony Minessale : > > >> The patch was it's ability to accept subsequent invites. > >> Your problem is that in sip each new attempt to send an invite is another > >> call. > >> > >> 484 is a final response so the call with too few digits is terminated. > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > --??????-- > From: mike at jerris.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 10:43:10 -0500 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Call loop? > > On Nov 4, 2009, at 10:25 AM, Diego Viola wrote: > >> Hello, >> >> I tried to help Roy with this issue yesterday, I saw that calls >> couldn't go through and then I made a sofia profile internal >> siptrace on. >> >> Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" >> and saw he had like 800 sessions. >> >> I thought it was an ACL issue but it wasn't, it seems like he >> reached a session limit, when I restarted his FS the problem went >> away. >> >> Best Regards, >> >> Diego > > > > > > --??????-- > From: jlenk at frontiernet.net > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 07:43:54 -0800 > Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > > Hi Carlos, > > very cool that the x64 version is included now! Hopefully this will get > more > people using the x64 version under Windows! > > Regards, > Jeff > > > Carlos Talbot wrote: >> >> On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson >> wrote: >> >>> Jeff, >>> >>> thanks a lot for the reply. I was a little confused by the fact that the >>> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >>> that I might lose something. As you say though, think that I'll cross my >>> fingers and try the updated release. I am running FreeSwitch on a test >>> machine at the moment until the target hardware arrives - hopefully >>> tomorrow, so I can afford to have a little play. >>> >> >> I usually try to update the svn file at least once a month. I have a new >> version ready that was compiled last night but am ironing out login >> issues >> with the FS dudes for upload access. Also, the SVN snapshot now includes >> binaries for 32 and 64 bit. It no longer includes flite though as the >> install file was approaching 80MB in size. I will revisit this later if >> others feel it important to include flite. >> >>> >>> You mentioned FreePBX V3. I had been fumbling around trying to work out >>> what >>> this is and from what I've read, it seems to provide a GUI Front End for >>> configuring FreeSwitch ? >>> >> Yes, it's still in development phase and as such not ready for production >> use. >> >>> >>> I am guessing that while it has been installed with FreeSwitch, I then >>> need >>> to run the FreePBX Installer to update the FreePBX/FreeSwitch >>> configuration >>> on my hardware ? >>> >>> >>> When I start FreeSwitch, it does not automatically load the WAMPServer. >>> >>> Freeswitch and WAMPServer are independant of each other. WAMPServer is >> bundled in this install for the purpose of FreePBX as MySQL, Apache and >> PHP >> are all required components of FreePBX. >> >> When I start WAMPServer manually, and open up localhost (127.0.0.1) in a >> web >>> browser, I can see the WampServer logo and various tools such as >>> phpinfo() >>> and phpmyadmin. FreePBX is there under Your Projects. >>> >>> If you want to configure FreePBX you need to click on the FreePBX.url >> shortcut that gets created on your desktop. >> >> >>> When I opened this up the first time, it appeared to want to install >>> FreePBX >>> over FreeSwitch, I tried to abort this when it was going to overwrite >>> some >>> FreeSwitch conf files and I thought I'd better not go on until I had a >>> better idea what was happening. I backed out of the FreePBX install and >>> now >>> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >>> it >>> looks like I'm going to have to reinstall anyway. >>> >>> So, for next time,am I right in thinking that I should proceed with >>> running >>> the FreePBX install from the WAMPServer menu ? >>> >> >> No, launch it from the shortcut as stated above. Unfortunately, at this >> time >> there is very little user documentation on configuring FreePBX. Here is >> the >> link to the developer's info: http://www.freepbx.org/v3 >> >> regards, >> >> Carlos >> >>> >>> >>> ----- Original Message ----- >>> From: "Jeff Lenk" >>> To: >>> Sent: Tuesday, November 03, 2009 2:48 PM >>> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >>> >>> >>> > >>> > Hi Dave, >>> > >>> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >>> > >>> > Just as you said freeswitch-1.0.4.exe is the tagged release and >>> > freeswitch.exe is a newer svn snapshot. >>> > >>> > There should be no problems installing the new version allthough best >>> to >>> > just try and see! >>> > >>> > Not sure why the newest one is from October 7th. >>> > >>> > Jeff >>> > >>> > >>> > Dave Stevenson wrote: >>> >> >>> >> Hi, >>> >> >>> >> I have read the Docs on the Wiki >>> >> ( >>> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >>> >> but am still not sure of what the different Windows install files >>> are. >>> >> Currently, the Windows Installer directory contains :- >>> >> >>> >> LATEST_SVN_15106 - 6 Bytes >>> >> >>> >> freeswitch-1.0.4.exe - 42 Megabytes >>> >> >>> >> freeswitch.exe - 32 Megabytes >>> >> >>> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >>> >> September. The freeswitch.exe file is dated 7th October and think >>> that >>> it >>> >> contains the minor updates since 3rd September ? >>> >> >>> >> Could someone who knows FreeSwitch under windows help me understand >>> the >>> >> two files please ? >>> >> >>> >> I chickened out of running the later exe in case it did something to >>> the >>> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >>> with >>> >> the old one already installed ? >>> >> What will it actually do ? >>> >> >>> >> regards >>> >> Dave >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> > >>> > -- >>> > View this message in context: >>> > >>> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >>> > Sent from the freeswitch-users mailing list archive at Nabble.com. >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > --??????-- > From: tculjaga at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 16:44:08 +0100 > Subject: Re: [Freeswitch-users] SIP Overlap support? > > Brian is right, > > pls, lets stop with exceptions and get stick to RFCs... otherwise it will > be a big mess ... > > T. > > > On Wed, Nov 4, 2009 at 3:03 PM, Brian West wrote: > > > I'm going to say No! > > > > /b > > > > > On Nov 4, 2009, at 2:23 AM, Dennis wrote: > > > >> is there a way to send something like 484 (or something else), which > >> does not make it a final answer and keep the call/socket alive? > >> > >> so we can ask the cirpack for further digits and decide what to do, if > >> the cirpack does not send any digits. > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > --??????-- > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 4 Nov 2009 09:46:11 -0600 > Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress > > Which revision of FreeSWITCH are you using? > > > > On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen wrote: > > > Here's what's in switch.conf.xml: > > > > > > > > > > > > > > Yet this message: SIP/2.0 503 Maximum Calls In Progress > > > > This is a small medical practice, 5-6 extensions, 3000 outbound minutes > > per month and at least the same inbound. We did fsctl shutdown restart > > and it flushed the sessions. What is going on? > > > > Thank you for your help! > > > > -- > > Roy A Cohen > > Network Advantage LLC > > www.net-vantage.com > > 413.223.9007 option 1 > > -------------------------------------------------- > > "Bringing Cost-Saving, State-of-the-Art Technology > > Solutions to Small and Mid-Size Organizations" > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _________________________________________________________________ > ?????????????????msn????? > http://ditu.live.com/?form=TL&swm=1 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Skypiax-load-error-tp3946656p3946759.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Nov 4 09:33:15 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Nov 2009 11:33:15 -0600 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <7b197bef0911040922y21aa10c0l53909b9f3ea07c5@mail.gmail.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <1257349434463-3946039.post@n2.nabble.com> <7b197bef0911040922y21aa10c0l53909b9f3ea07c5@mail.gmail.com> Message-ID: I looked out my window... but I didn't see pigs flying... did I miss something! :P /b On Nov 4, 2009, at 11:22 AM, Giovanni Maruzzelli wrote: > ...and will get more people using the x64 version of Windows! ;) > > -gm From msc at freeswitch.org Wed Nov 4 09:36:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Nov 2009 09:36:38 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7170307767D@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> <3C04B27FC880044F8FCD735D0D952FF7170307767D@EXMBXCLUS01.citservers.local> Message-ID: <87f2f3b90911040936j12094470ld8e4d5328cc109f3@mail.gmail.com> On Tue, Nov 3, 2009 at 6:27 PM, Ujjval Karihaloo wrote: > Was that sarcasm or you really mean it? > No, he's serious. There are some issues that are quite endemic to AT&T. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/8ce0d1b3/attachment-0002.html From stevendt at primrosebank.net Wed Nov 4 09:49:42 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 17:49:42 -0000 Subject: [Freeswitch-users] Precompiled Windows Binaries References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> Message-ID: <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> Hi Carlos, thanks a lot for the reply on the Windows stuff - and for doing the pre-compile ! It saves me from having to do that myself while I'm busy just trying to learn FreeSwitch - not an easy task in itself, well, not for me anyway !. As much as a GUI would make working with FS easier, I think I'll keep away from FreePBX at the moment, I'm liable just to get myself more confused and break what I have working. I'll wait for a "Production Ready" version before I mess with this again I think. Just one clarification then, am I right in thinking that, after I have installed a release version of FS, if I then install one of the SVNs over it, it will keep all configuration etc. in tact, i.e., I won't lose anything that I've changed in the conf files etc? Regards Dave ----- Original Message ----- From: Carlos Talbot To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, November 04, 2009 2:51 PM Subject: Re: [Freeswitch-users] Precompiled Windows Binaries On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson wrote: Jeff, thanks a lot for the reply. I was a little confused by the fact that the "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried that I might lose something. As you say though, think that I'll cross my fingers and try the updated release. I am running FreeSwitch on a test machine at the moment until the target hardware arrives - hopefully tomorrow, so I can afford to have a little play. I usually try to update the svn file at least once a month. I have a new version ready that was compiled last night but am ironing out login issues with the FS dudes for upload access. Also, the SVN snapshot now includes binaries for 32 and 64 bit. It no longer includes flite though as the install file was approaching 80MB in size. I will revisit this later if others feel it important to include flite. You mentioned FreePBX V3. I had been fumbling around trying to work out what this is and from what I've read, it seems to provide a GUI Front End for configuring FreeSwitch ? Yes, it's still in development phase and as such not ready for production use. I am guessing that while it has been installed with FreeSwitch, I then need to run the FreePBX Installer to update the FreePBX/FreeSwitch configuration on my hardware ? When I start FreeSwitch, it does not automatically load the WAMPServer. Freeswitch and WAMPServer are independant of each other. WAMPServer is bundled in this install for the purpose of FreePBX as MySQL, Apache and PHP are all required components of FreePBX. When I start WAMPServer manually, and open up localhost (127.0.0.1) in a web browser, I can see the WampServer logo and various tools such as phpinfo() and phpmyadmin. FreePBX is there under Your Projects. If you want to configure FreePBX you need to click on the FreePBX.url shortcut that gets created on your desktop. When I opened this up the first time, it appeared to want to install FreePBX over FreeSwitch, I tried to abort this when it was going to overwrite some FreeSwitch conf files and I thought I'd better not go on until I had a better idea what was happening. I backed out of the FreePBX install and now I can't get the FreePBX or phpmyadmin pages up again (missing files) so it looks like I'm going to have to reinstall anyway. So, for next time,am I right in thinking that I should proceed with running the FreePBX install from the WAMPServer menu ? No, launch it from the shortcut as stated above. Unfortunately, at this time there is very little user documentation on configuring FreePBX. Here is the link to the developer's info: http://www.freepbx.org/v3 regards, Carlos ----- Original Message ----- From: "Jeff Lenk" To: Sent: Tuesday, November 03, 2009 2:48 PM Subject: Re: [Freeswitch-users] Precompiled Windows Binaries > > Hi Dave, > > These are supported by "Carlos Talbot" . They also include Freepbx v3 > > Just as you said freeswitch-1.0.4.exe is the tagged release and > freeswitch.exe is a newer svn snapshot. > > There should be no problems installing the new version allthough best to > just try and see! > > Not sure why the newest one is from October 7th. > > Jeff > > > Dave Stevenson wrote: >> >> Hi, >> >> I have read the Docs on the Wiki >> (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> but am still not sure of what the different Windows install files are. >> Currently, the Windows Installer directory contains :- >> >> LATEST_SVN_15106 - 6 Bytes >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> freeswitch.exe - 32 Megabytes >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> September. The freeswitch.exe file is dated 7th October and think that it >> contains the minor updates since 3rd September ? >> >> Could someone who knows FreeSwitch under windows help me understand the >> two files please ? >> >> I chickened out of running the later exe in case it did something to the >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe with >> the old one already installed ? >> What will it actually do ? >> >> regards >> Dave >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/5eeb37ef/attachment-0002.html From jmillican at sentinelcommunications.com Wed Nov 4 09:52:18 2009 From: jmillican at sentinelcommunications.com (John Millican) Date: Wed, 04 Nov 2009 12:52:18 -0500 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <1257349434463-3946039.post@n2.nabble.com> <7b197bef0911040922y21aa10c0l53909b9f3ea07c5@mail.gmail.com> Message-ID: <4AF1BF52.7010000@sentinelcommunications.com> Brian West wrote: > I looked out my window... but I didn't see pigs flying... did I miss > something! :P > > /b > > On Nov 4, 2009, at 11:22 AM, Giovanni Maruzzelli wrote: > >> ...and will get more people using the x64 version of Windows! ;) >> >> -gm When their own commercials say that there old software is prone to crashing and hangs, why should I trust there "new" software? -- JohnM From stevendt at primrosebank.net Wed Nov 4 10:03:26 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 18:03:26 -0000 Subject: [Freeswitch-users] Gateway Error References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> Message-ID: Hi, I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway and am seeing the following error :- "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sigura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who know what will happen.." Having just bought the Gateway specifically for FS, that was a bit of a "rude awakening" ! Does anyone know of a fix in the pipeline, or am I sc***ed already ? Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/df99f265/attachment-0002.html From ujjval at simplesignal.com Wed Nov 4 10:11:43 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Wed, 4 Nov 2009 10:11:43 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <87f2f3b90911040936j12094470ld8e4d5328cc109f3@mail.gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> <3C04B27FC880044F8FCD735D0D952FF7170307767D@EXMBXCLUS01.citservers.local> <87f2f3b90911040936j12094470ld8e4d5328cc109f3@mail.gmail.com> Message-ID: <3C04B27FC880044F8FCD735D0D952FF717030777DE@EXMBXCLUS01.citservers.local> Interesting...Thx for the heads up. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, November 04, 2009 10:37 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator On Tue, Nov 3, 2009 at 6:27 PM, Ujjval Karihaloo > wrote: Was that sarcasm or you really mean it? No, he's serious. There are some issues that are quite endemic to AT&T. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/62658188/attachment-0002.html From msc at freeswitch.org Wed Nov 4 10:16:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Nov 2009 10:16:43 -0800 Subject: [Freeswitch-users] Users hanged up for unknown reason In-Reply-To: <1257244093831-3937601.post@n2.nabble.com> References: <1257244093831-3937601.post@n2.nabble.com> Message-ID: <87f2f3b90911041016u620ca88bk4f0d6a4ceb339b4b@mail.gmail.com> On Tue, Nov 3, 2009 at 2:28 AM, Maciej Aniserowicz < maciej.aniserowicz at gmail.com> wrote: > > Hi, > I have a strange problem. I control FS with commands sent by tcp in > response > to events published via tcp. I do something like: > 1) call 1st user > 2) call 2nd user > 3) 1st and 2nd talk > 4) call another user > 5) 1st and another talk > etc... > > Sometimes (quite regularly) users are hanged up (with cause > NORMAL_CLEARING) > even if they do not hangup manually. > > I pasted one such scenario in pastebin > (http://pastebin.freeswitch.org/10955), it includes logs from commands > sent > by me and events received from FS. Could someone take a look and see what > am > I doing wrong? > Seeing only the events it is difficult to see what triggered them. Can you repeat these tests and capture the debug output from the CLI? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/afc91d6c/attachment-0002.html From msc at freeswitch.org Wed Nov 4 10:23:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Nov 2009 10:23:01 -0800 Subject: [Freeswitch-users] Gateway Error In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> Message-ID: <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> On Wed, Nov 4, 2009 at 10:03 AM, Dave Stevenson wrote: > Hi, > > I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway > and am seeing the following error :- > > "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they meant > to say was 20 > This issue has so far been identified to happen on the following broken > platforms/devices: > Linksys/Sigura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so broken > who know what will happen.." > > Having just bought the Gateway specifically for FS, that was a bit of a > "rude awakening" ! > > Does anyone know of a fix in the pipeline, or am I sc***ed already ? > The cynical among us will say that you were hosed the moment you paid for a Linksys device. :) It's very sad but the FS devs find this kind of thing all the time. They've literally got all sorts of checks in the code to make sure that devices aren't saying one thing and doing something else. Cisco is not the only one to do stupid things like this. In any case, just be aware of it. If you want suggestions then list to the others here who can offer their experiences with various devices they have in production. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/ced0a94c/attachment-0002.html From stevendt at primrosebank.net Wed Nov 4 10:48:28 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 18:48:28 -0000 Subject: [Freeswitch-users] Gateway Error References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com><6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> Message-ID: <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> Michael et al - and specifically, the FS Developers, this is all the more annoying given the fact that the SPA-3102 was bought specifically to run with FreeSwitch following a recommendation here in the UK. It was just unwrapped this afternoon :-( (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). I am setting up a VOIP system at home, and this device sounded like the ideal gateway to the PSTN. What does the error message actually mean - is this device a non-starter or are there work-arounds or fixes to the code in progress ? Surely the device can't be as "broken" as the message - or am I just being too hopeful ? Regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, November 04, 2009 6:23 PM Subject: Re: [Freeswitch-users] Gateway Error On Wed, Nov 4, 2009 at 10:03 AM, Dave Stevenson wrote: Hi, I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway and am seeing the following error :- "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sigura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who know what will happen.." Having just bought the Gateway specifically for FS, that was a bit of a "rude awakening" ! Does anyone know of a fix in the pipeline, or am I sc***ed already ? The cynical among us will say that you were hosed the moment you paid for a Linksys device. :) It's very sad but the FS devs find this kind of thing all the time. They've literally got all sorts of checks in the code to make sure that devices aren't saying one thing and doing something else. Cisco is not the only one to do stupid things like this. In any case, just be aware of it. If you want suggestions then list to the others here who can offer their experiences with various devices they have in production. -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/4535e099/attachment-0002.html From mastermind202 at gmail.com Wed Nov 4 10:57:29 2009 From: mastermind202 at gmail.com (mm_202) Date: Wed, 4 Nov 2009 13:57:29 -0500 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <027f01ca5d6c$9c1603a0$d4420ae0$@com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> <025101ca5d10$f81228c0$e8367a40$@com> <20091104064201.GA15804@jdc.jasonjgw.net> <027f01ca5d6c$9c1603a0$d4420ae0$@com> Message-ID: <63de75710911041057x472d44aj9bc52bb460a8c8cd@mail.gmail.com> I had the exact same problem with the Cisco phones not being able to receive calls. I fixed it by messing around with the NAT settings in the internal sofia profile. From what I remember, I just removed the line and everything worked fine. -- mm_202. On Wed, Nov 4, 2009 at 11:33 AM, Peter J. Zandvoort wrote: > Absolutely agreed. To use Matthew's original car metaphor: When you just got > your learner's permit, the old Chevy may be a better choice than the > Ferrari. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason > White > Sent: Wednesday, November 04, 2009 1:42 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones > > Peter J. Zandvoort wrote: >> After looking at various asterisk distributions, SipX, 3CX and >> what-have-you, I've come to the conclusion that FreeSWITCH is by far the >> most advanced platform out there. Its architecture and performance is >> literally light years ahead of the rest and I have yet to come up with >> something that it can't do. But all that comes at a price: The learning >> curve is like scaling a brick wall. > > The most flexible and sophisticated tools tend to have this characteristic, > the best solution to which is a supportive community and good documentation. > FreeSWITCH has the community; the documentation is improving thanks to > ongoing > efforts to extend, clarify and enhance the wiki. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From testeador01 at gmail.com Wed Nov 4 11:34:09 2009 From: testeador01 at gmail.com (Milena) Date: Wed, 4 Nov 2009 14:34:09 -0500 Subject: [Freeswitch-users] Question about jingle_profiles In-Reply-To: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> References: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> Message-ID: Hello, A question to clarify: do you have an extension on your dialplan that matches "john" or "bob" and bridges the call? If you don't, you need to create it, if you do have it but it doesn't work, post the cli output for when you try to call so we can see what is going on. 2009/11/4 Jonathan Barou > Hi everybody, > > I actually working on mod_dingaling (gtalk). I can make call from FS to > Gtalk, and from Gtalk to FS. > But I have a problem, in jingle_profile I have a file like this : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > here when I put an user account like > john or bob its doesn't work whereas I put something like 1000 or 8400 it > works. > > When I tried to put a real phone number It doesn't work too (I have a > gateway with my PBX). > > Somebody know, why it doesn't work with name and work with number ? > > Thanks. > > > -- > Jonathan BAROU > SQLI LYON - CRCI > 0472405368 > jbarou at sqli.com > lyon.crci at sqli.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/c83f8580/attachment-0002.html From mike at jerris.com Wed Nov 4 11:36:49 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 4 Nov 2009 14:36:49 -0500 Subject: [Freeswitch-users] Gateway Error In-Reply-To: <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com><6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> Message-ID: <33167DE5-D670-46F0-BECA-4802B917E206@jerris.com> It means you need to go change the setting from the broken defaults, thats all. Mike On Nov 4, 2009, at 1:48 PM, Dave Stevenson wrote: > Michael et al - and specifically, the FS Developers, > > this is all the more annoying given the fact that the SPA-3102 was > bought specifically to run with FreeSwitch following a > recommendation here in the UK. It was just unwrapped this > afternoon :-( > > (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). > > I am setting up a VOIP system at home, and this device sounded like > the ideal gateway to the PSTN. > > What does the error message actually mean - is this device a non- > starter or are there work-arounds or fixes to the code in progress ? > > Surely the device can't be as "broken" as the message - or am I just > being too hopeful ? > > Regards > Dave > > > ----- Original Message ----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, November 04, 2009 6:23 PM > Subject: Re: [Freeswitch-users] Gateway Error > > > > On Wed, Nov 4, 2009 at 10:03 AM, Dave Stevenson > wrote: > Hi, > > I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice > Gateway and am seeing the following error :- > > "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what > they meant to say was 20 > This issue has so far been identified to happen on the following > broken platforms/devices: > Linksys/Sigura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so > broken who know what will happen.." > > Having just bought the Gateway specifically for FS, that was a bit > of a "rude awakening" ! > > Does anyone know of a fix in the pipeline, or am I sc***ed already ? > > The cynical among us will say that you were hosed the moment you > paid for a Linksys device. :) It's very sad but the FS devs find > this kind of thing all the time. They've literally got all sorts of > checks in the code to make sure that devices aren't saying one thing > and doing something else. Cisco is not the only one to do stupid > things like this. In any case, just be aware of it. > > If you want suggestions then list to the others here who can offer > their experiences with various devices they have in production. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/7a7ecbe7/attachment-0002.html From chris.chen2004 at gmail.com Wed Nov 4 11:45:02 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Wed, 4 Nov 2009 14:45:02 -0500 Subject: [Freeswitch-users] Question about jingle_profiles In-Reply-To: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> References: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> Message-ID: <507898380911041145u431865f8uc8877fce3c2e3778@mail.gmail.com> you have to define the extension "john" or "bob" or whatever number you want in the dialplan for the context "public". Just follow your jingle profile you define. Simple, no other tricks. Thanks, Chris On Wed, Nov 4, 2009 at 11:56 AM, Jonathan Barou wrote: > Hi everybody, > > I actually working on mod_dingaling (gtalk). I can make call from FS to > Gtalk, and from Gtalk to FS. > But I have a problem, in jingle_profile I have a file like this : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > here when I put an user account like > john or bob its doesn't work whereas I put something like 1000 or 8400 it > works. > > When I tried to put a real phone number It doesn't work too (I have a > gateway with my PBX). > > Somebody know, why it doesn't work with name and work with number ? > > Thanks. > > > -- > Jonathan BAROU > SQLI LYON - CRCI > 0472405368 > jbarou at sqli.com > lyon.crci at sqli.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/46d81714/attachment-0002.html From stevendt at primrosebank.net Wed Nov 4 11:49:11 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 19:49:11 -0000 Subject: [Freeswitch-users] Gateway Error References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com><6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> <33167DE5-D670-46F0-BECA-4802B917E206@jerris.com> Message-ID: <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> Phew ! Thanks Mike, I was very worried there. Now, if I just knew which were the "broken defaults", I'd know where to go next :-) Regards Dave ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, November 04, 2009 7:36 PM Subject: Re: [Freeswitch-users] Gateway Error It means you need to go change the setting from the broken defaults, thats all. Mike On Nov 4, 2009, at 1:48 PM, Dave Stevenson wrote: Michael et al - and specifically, the FS Developers, this is all the more annoying given the fact that the SPA-3102 was bought specifically to run with FreeSwitch following a recommendation here in the UK. It was just unwrapped this afternoon :-( (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). I am setting up a VOIP system at home, and this device sounded like the ideal gateway to the PSTN. What does the error message actually mean - is this device a non-starter or are there work-arounds or fixes to the code in progress ? Surely the device can't be as "broken" as the message - or am I just being too hopeful ? Regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, November 04, 2009 6:23 PM Subject: Re: [Freeswitch-users] Gateway Error On Wed, Nov 4, 2009 at 10:03 AM, Dave Stevenson wrote: Hi, I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway and am seeing the following error :- "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sigura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who know what will happen.." Having just bought the Gateway specifically for FS, that was a bit of a "rude awakening" ! Does anyone know of a fix in the pipeline, or am I sc***ed already ? The cynical among us will say that you were hosed the moment you paid for a Linksys device. :) It's very sad but the FS devs find this kind of thing all the time. They've literally got all sorts of checks in the code to make sure that devices aren't saying one thing and doing something else. Cisco is not the only one to do stupid things like this. In any case, just be aware of it. If you want suggestions then list to the others here who can offer their experiences with various devices they have in production. -MC -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/359364bb/attachment-0002.html From info at daccii.it Wed Nov 4 11:55:39 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Wed, 04 Nov 2009 20:55:39 +0100 Subject: [Freeswitch-users] Question about set/export applications Message-ID: <4AF1DC3B.6070607@daccii.it> Hi to all, i'm trying to setup a simple after hours ivr, without using lua/javascript, but only xml. What i do is to catch weekdays, set some vars, catch working hours based on weekdays, and, in the end, catch if it is working time or not. If not, just set another var. Actually the code is really bad, i'll reorganize it later, the biggest problem is that i can't read setted variables! This code --- --- should set variable working_day_a to true, but if, in the following extension, i check ${working_day_a} field i get it empty. From logs --- Dialplan: OpenZAP/1:1/03 Action set(working_day_a=true) . . . Dialplan: OpenZAP/1:1/03 Regex (FAIL) [working_day_a_hours] ${working_day_a}() =~ /^true$/ break=on-false --- Here extensions http://pastebin.freeswitch.org/10977 while here relevant parts of log http://pastebin.freeswitch.org/10978 Thank for your help! Best Regards, Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/7f0712ee/attachment-0002.vcf From info at daccii.it Wed Nov 4 12:04:36 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Wed, 04 Nov 2009 21:04:36 +0100 Subject: [Freeswitch-users] Question about set/export applications In-Reply-To: <4AF1DC3B.6070607@daccii.it> References: <4AF1DC3B.6070607@daccii.it> Message-ID: <4AF1DE54.4050300@daccii.it> Ehm, sorry, i just fixed it using inline attribute applying it on action/application/set tags: i missed it on dialplan wiki page. Sorry for the mail Albano Daniele Salvatore - Lavoro ha scritto: > Hi to all, > > i'm trying to setup a simple after hours ivr, without using > lua/javascript, but only xml. > > What i do is to catch weekdays, set some vars, catch working hours based > on weekdays, and, in the end, catch if it is working time or not. If > not, just set another var. > > Actually the code is really bad, i'll reorganize it later, the biggest > problem is that i can't read setted variables! > > . > . > . -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/a750e8bb/attachment-0002.vcf From larclap at yahoo.com Wed Nov 4 12:20:29 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 4 Nov 2009 12:20:29 -0800 Subject: [Freeswitch-users] Copy voicemail greeting Message-ID: <011501ca5d8c$415fc340$c41f49c0$@com> Is it possible to copy an existing wav greeting from one extension to another? I think something has to be added to db/voicemail_default.db, but it's not a text file. Is it just easier to re-record the message from the 2nd extension? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/9c68c2ea/attachment-0002.html From brian at freeswitch.org Wed Nov 4 12:24:32 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Nov 2009 14:24:32 -0600 Subject: [Freeswitch-users] Copy voicemail greeting In-Reply-To: <011501ca5d8c$415fc340$c41f49c0$@com> References: <011501ca5d8c$415fc340$c41f49c0$@com> Message-ID: copy the wav file and insert the record. /b On Nov 4, 2009, at 2:20 PM, Lars Zeb wrote: > Is it possible to copy an existing wav greeting from one extension > to another? I think something has to be added to db/ > voicemail_default.db, but it?s not a text file. > > Is it just easier to re-record the message from the 2nd extension? > > Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/6eb72a1f/attachment-0002.html From kristian.kielhofner at gmail.com Wed Nov 4 12:57:05 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 4 Nov 2009 15:57:05 -0500 Subject: [Freeswitch-users] Gateway Error In-Reply-To: <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> <33167DE5-D670-46F0-BECA-4802B917E206@jerris.com> <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> Message-ID: <2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo On Wed, Nov 4, 2009 at 2:49 PM, Dave Stevenson wrote: > Phew ! > > Thanks Mike, I was very worried there. > > Now, if I just knew which were the "broken defaults", I'd know where to go > next :-) > > Regards > Dave > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, November 04, 2009 7:36 PM > Subject: Re: [Freeswitch-users] Gateway Error > It means you need to go change the setting from the broken defaults, thats > all. > Mike > On Nov 4, 2009, at 1:48 PM, Dave Stevenson wrote: > > Michael et al - and specifically, the FS Developers, > > this is all the more annoying given the fact that the SPA-3102 was bought > specifically to run with FreeSwitch following a recommendation here in the > UK. It?was just unwrapped this afternoon :-( > > (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). > > I am setting up a VOIP system at home, and this device sounded like the > ideal gateway to the PSTN. > > What does the error message actually mean - is this device a non-starter or > are there work-arounds or?fixes to the code in progress ? > > Surely the device can't be as "broken" as the message - or am I just being > too hopeful ? > > Regards > Dave > > > > ----- Original Message ----- > From:?Michael Collins > To:?freeswitch-users at lists.freeswitch.org > Sent:?Wednesday, November 04, 2009 6:23 PM > Subject:?Re: [Freeswitch-users] Gateway Error > > > On Wed, Nov 4, 2009 at 10:03 AM, Dave > Stevenson??wrote: >> >> Hi, >> >> I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway >> and am seeing the following error :- >> >> "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they >> meant to say was 20 >> This issue has so far been identified to happen on the following broken >> platforms/devices: >> Linksys/Sigura aka Cisco >> ShoreTel >> Sonus/L3 >> We will try to fix it but some of the devices on this list are so broken >> who know what will happen.." >> >> Having just bought the Gateway specifically for FS, that was a bit of a >> "rude awakening" ! >> >> Does anyone know of a fix in the pipeline, or am I sc***ed already ? > > The cynical among us will say that you were hosed the moment you paid for a > Linksys device. :) It's very sad but the FS devs find this kind of thing all > the time. They've literally got all sorts of checks in the code to make sure > that devices aren't saying one thing and doing something else. Cisco is not > the only one to do stupid things like this. In any case, just be aware of > it. > > If you want suggestions then list to the others here who can offer their > experiences with various devices they have in production. > -MC > > > ________________________________ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ________________________________ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From carlos.talbot at gmail.com Wed Nov 4 12:58:20 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 4 Nov 2009 16:58:20 -0400 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> Message-ID: <5800526b0911041258w262f9277o9eba45b05ebbfc8c@mail.gmail.com> On Wed, Nov 4, 2009 at 1:49 PM, Dave Stevenson wrote: > Hi Carlos, > > Just one clarification then, am I right in thinking that, after I have > installed a release version of FS, if I then install one of the SVNs over > it, it will keep all configuration etc. in tact, i.e., I won't lose anything > that I've changed in the conf files etc? > > Regards > Dave > Hi Dave, Your best bet would be to keep 1.0.4 and SVN in separate installation directories since the SVN version will install default copies of the conf folder and overwrite your existing config. This is no different than your typical Windows App install file. You could just install SVN in a temp location and copy out all but the conf folder to your current 1.0.4 location. FYI, I just uploaded SVN 15355. regards, Carlos > > > > ----- Original Message ----- > *From:* Carlos Talbot > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Wednesday, November 04, 2009 2:51 PM > *Subject:* Re: [Freeswitch-users] Precompiled Windows Binaries > > > > On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson > wrote: > >> Jeff, >> >> thanks a lot for the reply. I was a little confused by the fact that the >> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried >> that I might lose something. As you say though, think that I'll cross my >> fingers and try the updated release. I am running FreeSwitch on a test >> machine at the moment until the target hardware arrives - hopefully >> tomorrow, so I can afford to have a little play. >> > > I usually try to update the svn file at least once a month. I have a new > version ready that was compiled last night but am ironing out login issues > with the FS dudes for upload access. Also, the SVN snapshot now includes > binaries for 32 and 64 bit. It no longer includes flite though as the > install file was approaching 80MB in size. I will revisit this later if > others feel it important to include flite. > >> >> You mentioned FreePBX V3. I had been fumbling around trying to work out >> what >> this is and from what I've read, it seems to provide a GUI Front End for >> configuring FreeSwitch ? >> > Yes, it's still in development phase and as such not ready for production > use. > >> >> I am guessing that while it has been installed with FreeSwitch, I then >> need >> to run the FreePBX Installer to update the FreePBX/FreeSwitch >> configuration >> on my hardware ? >> >> >> When I start FreeSwitch, it does not automatically load the WAMPServer. >> >> Freeswitch and WAMPServer are independant of each other. WAMPServer is > bundled in this install for the purpose of FreePBX as MySQL, Apache and PHP > are all required components of FreePBX. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a >> web >> browser, I can see the WampServer logo and various tools such as phpinfo() >> and phpmyadmin. FreePBX is there under Your Projects. >> >> If you want to configure FreePBX you need to click on the FreePBX.url > shortcut that gets created on your desktop. > > >> When I opened this up the first time, it appeared to want to install >> FreePBX >> over FreeSwitch, I tried to abort this when it was going to overwrite some >> FreeSwitch conf files and I thought I'd better not go on until I had a >> better idea what was happening. I backed out of the FreePBX install and >> now >> I can't get the FreePBX or phpmyadmin pages up again (missing files) so it >> looks like I'm going to have to reinstall anyway. >> >> So, for next time,am I right in thinking that I should proceed with >> running >> the FreePBX install from the WAMPServer menu ? >> > > No, launch it from the shortcut as stated above. Unfortunately, at this > time there is very little user documentation on configuring FreePBX. Here is > the link to the developer's info: http://www.freepbx.org/v3 > > regards, > > Carlos > >> >> >> ----- Original Message ----- >> From: "Jeff Lenk" >> To: >> Sent: Tuesday, November 03, 2009 2:48 PM >> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >> >> >> > >> > Hi Dave, >> > >> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >> > >> > Just as you said freeswitch-1.0.4.exe is the tagged release and >> > freeswitch.exe is a newer svn snapshot. >> > >> > There should be no problems installing the new version allthough best to >> > just try and see! >> > >> > Not sure why the newest one is from October 7th. >> > >> > Jeff >> > >> > >> > Dave Stevenson wrote: >> >> >> >> Hi, >> >> >> >> I have read the Docs on the Wiki >> >> ( >> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> >> but am still not sure of what the different Windows install files are. >> >> Currently, the Windows Installer directory contains :- >> >> >> >> LATEST_SVN_15106 - 6 Bytes >> >> >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> >> >> freeswitch.exe - 32 Megabytes >> >> >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> >> September. The freeswitch.exe file is dated 7th October and think that >> it >> >> contains the minor updates since 3rd September ? >> >> >> >> Could someone who knows FreeSwitch under windows help me understand the >> >> two files please ? >> >> >> >> I chickened out of running the later exe in case it did something to >> the >> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >> with >> >> the old one already installed ? >> >> What will it actually do ? >> >> >> >> regards >> >> Dave >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > -- >> > View this message in context: >> > >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/29d2ebce/attachment-0002.html From stevendt at primrosebank.net Wed Nov 4 13:07:27 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 21:07:27 -0000 Subject: [Freeswitch-users] Gateway Error References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com><6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> <2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> Message-ID: <1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com> Kristian, thanks very much ! After trawling the internet, I had just found the Linksys "RTP Packet Size" and worked out that it was the "ptime" parameter that FreeSwitch flagged as a problem. I had literally just changed that the very minute your post came in ! I'm off not to read that Wiki that looks like it will tell me everything else that I need to know on the SPA-3012, thanks and best regards Dave ----- Original Message ----- From: "Kristian Kielhofner" To: Sent: Wednesday, November 04, 2009 8:57 PM Subject: Re: [Freeswitch-users] Gateway Error http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo On Wed, Nov 4, 2009 at 2:49 PM, Dave Stevenson wrote: > Phew ! > > Thanks Mike, I was very worried there. > > Now, if I just knew which were the "broken defaults", I'd know where to go > next :-) > > Regards > Dave > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, November 04, 2009 7:36 PM > Subject: Re: [Freeswitch-users] Gateway Error > It means you need to go change the setting from the broken defaults, thats > all. > Mike > On Nov 4, 2009, at 1:48 PM, Dave Stevenson wrote: > > Michael et al - and specifically, the FS Developers, > > this is all the more annoying given the fact that the SPA-3102 was bought > specifically to run with FreeSwitch following a recommendation here in the > UK. It was just unwrapped this afternoon :-( > > (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). > > I am setting up a VOIP system at home, and this device sounded like the > ideal gateway to the PSTN. > > What does the error message actually mean - is this device a non-starter > or > are there work-arounds or fixes to the code in progress ? > > Surely the device can't be as "broken" as the message - or am I just being > too hopeful ? > > Regards > Dave > > > > ----- Original Message ----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, November 04, 2009 6:23 PM > Subject: Re: [Freeswitch-users] Gateway Error > > > On Wed, Nov 4, 2009 at 10:03 AM, Dave > Stevenson wrote: >> >> Hi, >> >> I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice >> Gateway >> and am seeing the following error :- >> >> "[WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they >> meant to say was 20 >> This issue has so far been identified to happen on the following broken >> platforms/devices: >> Linksys/Sigura aka Cisco >> ShoreTel >> Sonus/L3 >> We will try to fix it but some of the devices on this list are so broken >> who know what will happen.." >> >> Having just bought the Gateway specifically for FS, that was a bit of a >> "rude awakening" ! >> >> Does anyone know of a fix in the pipeline, or am I sc***ed already ? > > The cynical among us will say that you were hosed the moment you paid for > a > Linksys device. :) It's very sad but the FS devs find this kind of thing > all > the time. They've literally got all sorts of checks in the code to make > sure > that devices aren't saying one thing and doing something else. Cisco is > not > the only one to do stupid things like this. In any case, just be aware of > it. > > If you want suggestions then list to the others here who can offer their > experiences with various devices they have in production. > -MC > > > ________________________________ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ________________________________ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From hjqlopez at hotmail.com Wed Nov 4 13:39:28 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Wed, 4 Nov 2009 16:39:28 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: Thanks for your time, -The scenario is still the same: Always bypass media. Environment 100% NAT free :-) Call established from A to B through FS. Then...? Blind transfer from B to C (Refer-to: C) RTP should go directly between A and C. -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the REFER-to:C, BUT there is no 2-way audio.? Only RTP from C to A (due to the lack of reINVITE to A, after C answers). Please check SIP diagram here: http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html -What it's wrong with r15332 is there is not such call to C. For sure I know SIP is a protocol, may be my description was not clear but this SIP diagram speaks by itself ;-) http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html -You could check the sofia debug for r15332 here: http://pastebin.com/m6f2b3836 Best regards, Humberto > > I don't know what you are talking about anymore. > > The scenario I had tested is when a call is bridged in bypass_media=true > bridge > and you blind transfer that call back to the dialplan > > as soon as it hits the routing state it will resume media. > > > it has been confirmed to not work and confirmed to have been fixed several > time and if you are still having a problem you must have something blocking > some of your packets or something . > > You have to understand that sip is a protocol and your description is > completely non-standard. > Perhaps you should get a console trace and attach it to a jira. The trace > probably makes more sense to me. > > sofia profile internal siptrace on > console loglevel debug > > reproduce and attach the whole capture. > > > > On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > >> >> Hi, >> >> I tried r15332 and set in the sofia profile: >> >> a) bypass_media_after_bridge=true only >> b) bypass_media_after_bridge=true, param name="media-option" >> value="resume-media-on-hold"/> >> >> >> In both cases FS is hanging up the initial call (A to FS) after accepting >> the REFER to C: >> >> A <- reINVITE with FS' SDP <- FS >> A -> 200 -> FS >> A <- ACK <- FS >> A <- BYE <- FS >> >> The call to C is not even tried. >> >> I found this line is the logs that could give some idea: >> >> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] >> after sending the ACK for the reINVITE >> >> >> Regards, >> >> >> Humberto >> >>>please try r15326 >>>I think i have it working. >>> >>>I recommend for optimal results you set bypass_media_after_bridge=true >>>either as a global or in your DP in place of bypass_media=true >>> >>> >>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >> hotmail.com>wrote: >>> >>>> Hi Mike, >>>> >>>> I re-tried with trunk rev 15319 but I got almost the same behavior: >> There >>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. >> But >>>> still there is no reINVITE for A (with C's SDP) after the call from FS >> to C >>>> is established. >>>> >>>> Anyway, we decided for now to do a different implementation but if you >> want >>>> to explore more in this issue count me in ;-) >>>> >>>> >>>> Thank you very much! >>>> >>>> Humberto >> >> >> _________________________________________________________________ >> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they >> e-mail you. >> http://go.microsoft.com/?linkid=9691817 >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > _________________________________________________________________ > Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now > http://go.microsoft.com/?linkid=9691818 _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook. http://go.microsoft.com/?linkid=9691816 From jerry.richards at teotech.com Wed Nov 4 13:48:51 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 4 Nov 2009 13:48:51 -0800 Subject: [Freeswitch-users] Dial Plan Question In-Reply-To: <4AF0A457.5080702@gmail.com> References: <0A46BCC1ED4C452CAD31DF64A734C492@greyhawk.tonecommander.com> <4AF0A457.5080702@gmail.com> Message-ID: Okay. Say we want 1000 internal user extensions and want them to be configured with individual dial plans that route the call based on the extension's callgroup, time-of-day, and presence. Would be okay to create a static XML dialplan file for each extension, so calls to/from each extension would be routed uniquely based upon these parameters? This approach sounds straightforward to us. Best Regards, Jerry -----Original Message----- From: Shelby Ramsey [mailto:sicfslist at gmail.com] Sent: Tuesday, November 03, 2009 1:45 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dial Plan Question I think the real question is what are you trying to do ... for some things it's very easy to just whip up a static XML file and be done with it. For others you probably want some sort of interaction with a DB. The options here are pretty endless: -- XML curl -- handing off the call to a script call from a static dial plan (use lua if there is going to be any load) -- event_socket -- mod_lcr But ultimately I think it's what you're trying to accomplish that matters. For a PBX install I'd say static files is probably about as easy as it is going to get. For delivering a service you'd probably want interaction with a DB. I've use XML curl a lot and have even starting using direct DB queries from static dialplans using mod_memcache and memcachedb (not memcache ... persistent storage). SDR Jerry Richards wrote: > My understanding of DialPlan/CallRouting is that it can be > accomplished via static XML tags, or alternatively, via a DialPlan > Application that interfaces with the dptools module. > > Question: If my above assumption is true, how does one select one > approach over the other? What is the criteria/considerations that > would govern the decision? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > From brian at freeswitch.org Wed Nov 4 13:59:11 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Nov 2009 15:59:11 -0600 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <1E83D0AE-9587-4CC3-8FBD-073217A839A0@freeswitch.org> What phones are you using? I do this exact same scenario without a problem over and over testing with anthm. So I would like to know what phones you're using. /b On Nov 4, 2009, at 3:39 PM, Humberto Quintana wrote: > > Thanks for your time, > > -The scenario is still the same: > > Always bypass media. > Environment 100% NAT free :-) > Call established from A to B through FS. Then... > Blind transfer from B to C (Refer-to: C) > RTP should go directly between A and C. > > > -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the > REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due > to the lack of reINVITE to A, after C answers). > > Please check SIP diagram here: > > http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html > > > -What it's wrong with r15332 is there is not such call to C. For > sure I know SIP is a protocol, may be my description was not clear > but this SIP diagram speaks by itself ;-) > > http://provision.netcelerate.net/ngrep/ > blindxfer2009-11-04rev15332.html > > > -You could check the sofia debug for r15332 here: > http://pastebin.com/m6f2b3836 > > > Best regards, > > Humberto > >> >> I don't know what you are talking about anymore. >> >> The scenario I had tested is when a call is bridged in >> bypass_media=true >> bridge >> and you blind transfer that call back to the dialplan >> >> as soon as it hits the routing state it will resume media. >> >> >> it has been confirmed to not work and confirmed to have been fixed >> several >> time and if you are still having a problem you must have something >> blocking >> some of your packets or something . >> >> You have to understand that sip is a protocol and your description is >> completely non-standard. >> Perhaps you should get a console trace and attach it to a jira. The >> trace >> probably makes more sense to me. >> >> sofia profile internal siptrace on >> console loglevel debug >> >> reproduce and attach the whole capture. >> >> >> >> On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: >> >>> >>> Hi, >>> >>> I tried r15332 and set in the sofia profile: >>> >>> a) bypass_media_after_bridge=true only >>> b) bypass_media_after_bridge=true, param name="media-option" >>> value="resume-media-on-hold"/> >>> >>> >>> In both cases FS is hanging up the initial call (A to FS) after >>> accepting >>> the REFER to C: >>> >>> A <- reINVITE with FS' SDP <- FS >>> A -> 200 -> FS >>> A <- ACK <- FS >>> A <- BYE <- FS >>> >>> The call to C is not even tried. >>> >>> I found this line is the logs that could give some idea: >>> >>> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >>> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] >>> [RECOVERY_ON_TIMER_EXPIRE] >>> after sending the ACK for the reINVITE >>> >>> >>> Regards, >>> >>> >>> Humberto >>> >>>> please try r15326 >>>> I think i have it working. >>>> >>>> I recommend for optimal results you set >>>> bypass_media_after_bridge=true >>>> either as a global or in your DP in place of bypass_media=true >>>> >>>> >>>> On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >>> hotmail.com>wrote: >>>> >>>>> Hi Mike, >>>>> >>>>> I re-tried with trunk rev 15319 but I got almost the same >>>>> behavior: >>> There >>>>> is now a reINVITE (with FS' SDP) going to A when the REFER is >>>>> accepted. >>> But >>>>> still there is no reINVITE for A (with C's SDP) after the call >>>>> from FS >>> to C >>>>> is established. >>>>> >>>>> Anyway, we decided for now to do a different implementation but >>>>> if you >>> want >>>>> to explore more in this issue count me in ;-) >>>>> >>>>> >>>>> Thank you very much! >>>>> >>>>> Humberto >>> >>> >>> _________________________________________________________________ >>> Windows Live: Friends get your Flickr, Yelp, and Digg updates when >>> they >>> e-mail you. >>> http://go.microsoft.com/?linkid=9691817 >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> _________________________________________________________________ >> Ready. Set. Get a great deal on Windows 7. See fantastic deals on >> Windows 7 now >> http://go.microsoft.com/?linkid=9691818 > > _________________________________________________________________ > Windows Live: Make it easier for your friends to see what you?re up > to on Facebook. > http://go.microsoft.com/?linkid=9691816 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From oseslija at gmail.com Wed Nov 4 14:38:13 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Wed, 4 Nov 2009 23:38:13 +0100 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> Message-ID: <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> Hello, I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS. Phones are registering to CCME and FS simultaneously. So far, everything is working just fine. I think SCCP will be obsolete in the future since even Cisco is working more and more on SIP. OTOH, I really hate SIP images for 79x1 (that's why I put CCME in the first place), where as ones for 79x0 are behaving much better. Regards, Ognjen On Tue, Nov 3, 2009 at 6:52 PM, mm_202 wrote: > FS doesnt support SCCP (from what I gathered, just because no one has > bothered coding it). > > Are there other users out there has use SCCP and FS? (with some > middleware in between) > > If enough people would find a use for it, I'd be willing to actually > code it (esp if someone offered a bounty). > So, would anyone besides me want/use a SCCP endpoint in FS? > > -- mm_202. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/22dd30a9/attachment-0002.html From msc at freeswitch.org Wed Nov 4 14:58:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Nov 2009 14:58:44 -0800 Subject: [Freeswitch-users] Dial Plan Question In-Reply-To: References: <0A46BCC1ED4C452CAD31DF64A734C492@greyhawk.tonecommander.com> <4AF0A457.5080702@gmail.com> Message-ID: <87f2f3b90911041458p789ed510h75205688c0f23e99@mail.gmail.com> On Wed, Nov 4, 2009 at 1:48 PM, Jerry Richards wrote: > > Okay. Say we want 1000 internal user extensions and want them to be > configured with individual dial plans that route the call based on the > extension's callgroup, time-of-day, and presence. Would be okay to create > a > static XML dialplan file for each extension, so calls to/from each > extension > would be routed uniquely based upon these parameters? This approach sounds > straightforward to us. > > Best Regards, > Jerry > > By "static" do you mean "doesn't change very often"? :) I don't see why you couldn't do this, although I'd be interested in knowing how easy/hard it is for you to maintain something like this. My guess is that those who have dialplans this large and complex are probably using mod_xml_curl and serving up their dialplans from another server. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/7749df54/attachment-0002.html From stevendt at primrosebank.net Wed Nov 4 15:59:31 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 4 Nov 2009 23:59:31 -0000 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><1257259714704-3938887.post@n2.nabble.com><5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com><6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com><2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> <1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com> Message-ID: <665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> I am trying to follow the configuration give in the "SPA3102 FreeSwitch HowTo". When I create the 00_spa3102.xml file, FreeSwitch won't load. If I rename the file (to, say .txt) then rename it to an xml once FreeSwitch is up and do a "reloadxml" command, I get an error flagged :- +OK [[error near line 3379]: missing >] I'm pretty sure the error must be in the 00_spa3102.xml file, but I can't see where the error might be - it looks identical to that on the Wiki page ? regards Dave From jmesquita at freeswitch.org Wed Nov 4 16:09:04 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 4 Nov 2009 22:09:04 -0200 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo In-Reply-To: <665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <6516A202CEE6464E9E74050A60E17894@bp1.ad.bp.com> <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> <33167DE5-D670-46F0-BECA-4802B917E206@jerris.com> <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> <2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> <1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com> <665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> Message-ID: Look at this line on the freeswitch.fsxml and it will tell you exactly where the problem is. Beware that nested comments are not allowed in XML. -- JM On Wed, Nov 4, 2009 at 9:59 PM, Dave Stevenson wrote: > I am trying to follow the configuration give in the "SPA3102 FreeSwitch > HowTo". > > When I create the 00_spa3102.xml file, FreeSwitch won't load. > If I rename the file (to, say .txt) then rename it to an xml once > FreeSwitch > is up and do a "reloadxml" command, I get an error flagged :- > > +OK [[error near line 3379]: missing >] > > I'm pretty sure the error must be in the 00_spa3102.xml file, but I can't > see where the error might be - it looks identical to that on the Wiki page > ? > > regards > Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/a0f7594f/attachment-0002.html From msc at freeswitch.org Wed Nov 4 16:27:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Nov 2009 16:27:36 -0800 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com> <688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com> <33167DE5-D670-46F0-BECA-4802B917E206@jerris.com> <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> <2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> <1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com> <665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> Message-ID: <87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com> FYI, the file JM mentioned is actually in freeswitch/log directory. :) The file "freeswitch.fsxml" is the monster file that has all of the individual XML files included. When you got find line 3379 you'll most likely see a missing ">" char on line 3378 or near there. You'll have a few more of these before you're an expert, I guarantee it. :D It's frequently just a typo. -MC 2009/11/4 Jo?o Mesquita > Look at this line on the freeswitch.fsxml and it will tell you exactly > where the problem is. > > Beware that nested comments are not allowed in XML. > > -- JM > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/81f1315d/attachment-0002.html From stevendt at primrosebank.net Wed Nov 4 16:49:12 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 5 Nov 2009 00:49:12 -0000 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com><2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com><1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com><665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> <87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com> Message-ID: <97FBB4B6002848BCA4F2D89F13626754@bp1.ad.bp.com> Joao & Michael, thanks a lot for the pointers to the combined xml file. Nothing as forgivable as a typo - I'm just dumb ! I had not changed the entry for destination number from the sample file :- To use the actual destination number - duh ! - I had changed the IP address, but not noticed the other required field :-( Still, with your help, I've got it now Regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Thursday, November 05, 2009 12:27 AM Subject: Re: [Freeswitch-users] SPA3102 FreeSwitch HowTo FYI, the file JM mentioned is actually in freeswitch/log directory. :) The file "freeswitch.fsxml" is the monster file that has all of the individual XML files included. When you got find line 3379 you'll most likely see a missing ">" char on line 3378 or near there. You'll have a few more of these before you're an expert, I guarantee it. :D It's frequently just a typo. -MC 2009/11/4 Jo?o Mesquita Look at this line on the freeswitch.fsxml and it will tell you exactly where the problem is. Beware that nested comments are not allowed in XML. -- JM ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/08421800/attachment-0002.html From larclap at yahoo.com Wed Nov 4 17:10:46 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 4 Nov 2009 17:10:46 -0800 Subject: [Freeswitch-users] Copy voicemail greeting In-Reply-To: References: <011501ca5d8c$415fc340$c41f49c0$@com> Message-ID: <01a501ca5db4$cf0024b0$6d006e10$@com> What tool/GUI do you use to edit the db contents? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, November 04, 2009 12:25 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Copy voicemail greeting copy the wav file and insert the record. /b On Nov 4, 2009, at 2:20 PM, Lars Zeb wrote: Is it possible to copy an existing wav greeting from one extension to another? I think something has to be added to db/voicemail_default.db, but it's not a text file. Is it just easier to re-record the message from the 2nd extension? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/f472204e/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 4 17:12:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Nov 2009 19:12:43 -0600 Subject: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress In-Reply-To: <4AF1A525.5090909@net-vantage.com> References: <4AF1A525.5090909@net-vantage.com> Message-ID: <191c3a030911041712l7f6362fbw93d1d3695ed433e5@mail.gmail.com> how often? what platform is the machine hardware do you have a console trace by entering console loglevel debug and looking for anything odd? The only causes are Being unable to launch threads from process limits or limitations of the os The profile is restarting. maybe your ip or default gateway is changing edit /usr/local/freeswitch/conf/autoload_configs/sofia.conf.xml and uncomment On Wed, Nov 4, 2009 at 10:00 AM, RA Cohen wrote: > > FreeSWITCH Version 1.0.trunk (15321) > > -- > Roy A Cohen > Network Advantage LLC > www.net-vantage.com > 413.223.9007 option 1 > -------------------------------------------------- > "Bringing Cost-Saving, State-of-the-Art Technology > Solutions to Small and Mid-Size Organizations" > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/130a6aec/attachment-0002.html From ujjval at simplesignal.com Wed Nov 4 18:09:01 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Wed, 4 Nov 2009 18:09:01 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> Message-ID: <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> Any ideas on the below...has anyone implemented the below: Once I have the Moderator and Participants logged on, how do I invoke the moderator previlidges, LIk esay muting everyone/someone or kicking someone out of the Conf and the like? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo Sent: Monday, November 02, 2009 12:52 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Rob: Once I have the Moderator and Participants logged on, how do I invoke the moderator previlidges, LIk esay muting everyone/someone or kicking someone out of the Conf and the like? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Friday, October 30, 2009 9:34 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hm, strange. I haven't seen that before. Can you pastebin your logs at debug level? On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > It's strange... a tcpdump tells me that there is no DTMF from my > provider when using IVR, but when I call into a TN that goes > directly into the Conference App, I see DTMF from the provider. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 7:23 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > I've never had any problem with that. Is your logging at debug level > so you can see the RECV DTFM in the log/fs_cli? Are you calling from > a SIP phone on the pbx, or via a PSTN provider? Maybe your provider > isn't passing them through. > > Make sure your logging is turned up then try something simpler, like > calling the echo application, and see if DTFM comes through. > > Rob > > On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: > >> Rob: >> >> For some reason, I don't see the DTMF appear on the fs_CLI when >> using the below configuration....so it basically timesout. >> >> UK >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, October 26, 2009 9:21 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Thx a lot Rob, reading the wiki your way or using IVR seems correct.. >> =============== >> The wiki also says that the wait-mod might be "used in conjunction >> with an IVR where the moderators are authenticated with an extra >> pass- >> code", which is what I did. I guess that's why I didn't understand >> the point of the +pin. >> ====================== >> >> I will try it out. >> >> Again thx a lot for your help. Will keep everyone posted. >> >> ________________________________________ >> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >> Sent: Friday, October 23, 2009 12:22 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I just re-tested with the pin in my dial plan: >> >> >> >> And it doesn't challenge me for the pin. I just drop right in. I >> figured this is how it was intended, since the wiki says the pin is >> set initially and only challenged in later attempts [by future >> callers]: >> >> "The first time a conference name (confname) is used, it will be >> created on demand, and the pin will be set to what ever is specified >> at that time: the pin in the data string if specified, or if not, the >> "pin" setting in the conference profile, and if that is also >> unspecified, then there is no pin protection. Any later attempt to >> join the conference must specify the same pin number, if one existed >> when it was created. " >> >> >> The wiki also says that the wait-mod might be "used in conjunction >> with an IVR where the moderators are authenticated with an extra >> pass- >> code", which is what I did. I guess that's why I didn't understand >> the point of the +pin. >> >> I'm sure there's a scenario where its used and useful, the wiki just >> doesn't explain it. >> >> Rob >> >> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >> >>> Well first off you're not defining a pine here... >>> >>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>> pin >>> number] >>> >>> That might be why its not asking for a pin. >>> >>> /b >>> >>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>> >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Wed Nov 4 21:55:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Nov 2009 23:55:38 -0600 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> Message-ID: <191c3a030911042155i64faeedmbd72d4fc5f51ca45@mail.gmail.com> but if he wants to code it we wouldn't mind it right? =p On Wed, Nov 4, 2009 at 4:38 PM, Ognjen Seslija wrote: > Hello, > > I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS. > Phones are registering to CCME and FS simultaneously. So far, everything is > working just fine. > > I think SCCP will be obsolete in the future since even Cisco is working > more and more on SIP. OTOH, I really hate SIP images for 79x1 (that's why I > put CCME in the first place), where as ones for 79x0 are behaving much > better. > > Regards, > Ognjen > > > > On Tue, Nov 3, 2009 at 6:52 PM, mm_202 wrote: > >> FS doesnt support SCCP (from what I gathered, just because no one has >> bothered coding it). >> >> Are there other users out there has use SCCP and FS? (with some >> middleware in between) >> >> If enough people would find a use for it, I'd be willing to actually >> code it (esp if someone offered a bounty). >> So, would anyone besides me want/use a SCCP endpoint in FS? >> >> -- mm_202. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091104/3b315101/attachment-0002.html From ahmedmunir007 at gmail.com Wed Nov 4 22:22:08 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Thu, 5 Nov 2009 11:22:08 +0500 Subject: [Freeswitch-users] Calling more than 1 variable in condition Message-ID: Hi, In my dial plan I've created a variable named SIP_CALL, PSTN_CALL. If SIP_CALL = true, it dials out to sip call, when PSTN_CALL=true, it dials out to landline call, as I provide sample below; The problem I'm facing is how can I apply condition when I've to call more than 1 variables? Like if there are no records in SIP numbering plan table and PSTN numbering plan table so it get the digits and dial out the to carrier (how to apply AND operation in condition?) i.e. Kindly advise for this issue. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/78020414/attachment-0002.html From tculjaga at gmail.com Wed Nov 4 23:28:48 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 5 Nov 2009 08:28:48 +0100 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <191c3a030911042155i64faeedmbd72d4fc5f51ca45@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> <191c3a030911042155i64faeedmbd72d4fc5f51ca45@mail.gmail.com> Message-ID: <65d96fc80911042328g5c25ed51o2aa90f2618ba9638@mail.gmail.com> On Thu, Nov 5, 2009 at 6:55 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > but if he wants to code it we wouldn't mind it right? =p > > > > On Wed, Nov 4, 2009 at 4:38 PM, Ognjen Seslija wrote: > >> Hello, >> >> I would just say, a mod_skinny is more than welcome and i will the 1st one willing to use it. T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/04d03ef1/attachment-0002.html From math.parent at gmail.com Wed Nov 4 23:33:20 2009 From: math.parent at gmail.com (Mathieu Parent) Date: Thu, 5 Nov 2009 08:33:20 +0100 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> Message-ID: <960738410911042333x1e49719fu34f353bf015c9d95@mail.gmail.com> Hello, On Wed, Nov 4, 2009 at 11:38 PM, Ognjen Seslija wrote: > Hello, > > I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS. > Phones are registering to CCME and FS simultaneously. So far, everything is > working just fine. Great. I'm interrested by this simultaneous registering. Do you havve some docs? Mathieu Parent From d_hound at ymail.com Wed Nov 4 23:44:44 2009 From: d_hound at ymail.com (Hound Dog) Date: Wed, 4 Nov 2009 23:44:44 -0800 (PST) Subject: [Freeswitch-users] problem with failover routes for LCR A-Z scenario Message-ID: <727374.16142.qm@web111917.mail.gq1.yahoo.com> I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) it came out a little long , sorry for that :) I found that it difficult/impossible to make LCR use only carriers that I choose scenario is as follows , taking the UK as example for a destination ( prices are not real , just an example ) I have 2 carriers offering routes to the UK , landline and mobile my buying prices Destination carrier1 Price carrier2 price 44 (all UK) $0.01 $0.01 447 (UK mobile) $0.15 $0.19 my selling prices Destination price 44 (all UK) $0.015 447 (UKmobile) $0.17 so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable for UK mobile I can ** only ** make a profit if I use carrier 1 ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) translating it to MOD_LCR information digits rate carrier_id ( other columns ignored ) 44 0.01 1 44 0.01 2 447 0.015 1 this looks good : 44 prefix will be shared between carrier 1 and 2 447 prefix will only go to carreir 1 so it fits perfectly - BUT testing this I get - API CALL [lcr(447965404547)] output: | Digit Match | Carrier | Rate | Codec | CID Regexp | Dialstring | | 447 | carr1 | 0.15 | G711 | | [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.1 | | 44 | carr2 | 0.01 | G711 | | [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.2 | Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) - it all makes sense ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail so far I didnt find a solution for that , so if there is one I love bo pointed there I did think it over a little and came up with 2 options that could be used , and I am also planning to code them and propose patch to maintainers , I would love to get comments on those ( in case there are no existing solution ) option 1 - setting some routes as last option , add another param to the LCR table called last_route , when hitting a route with last_route=1, stop processing additional routes and return your routing decision so far so in our case the route entry with 44 to carrier1 will have last_route=1 , the other 44 routes will have last_route=0 to allow for failover option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. in other words dont failover from a finer route to a wider route. it will need to be a global option and I will be quite simple to use, but will require entering mutiple entries of the same length prefix for each carrier you would like to use its intutive and relatively simple to manage , but requires more lcr entries to get you where you want From jbarou at sqli.com Thu Nov 5 00:40:20 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Thu, 5 Nov 2009 09:40:20 +0100 Subject: [Freeswitch-users] Question about jingle_profiles In-Reply-To: <507898380911041145u431865f8uc8877fce3c2e3778@mail.gmail.com> References: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> <507898380911041145u431865f8uc8877fce3c2e3778@mail.gmail.com> Message-ID: <8048ff7f0911050040n791b59efp33dcc4a4236c71ca@mail.gmail.com> Hi, In the dialplan I have the extension "Local_extension" with "john" and it's working when I call john from the account 1000 with softphone. When I try to make a call from Gtalk to FS I have that in the console : 09-11-05 09:25:58.370659 [DEBUG] switch_rtp.c:2780 Activate VAD codec PCMU 20ms 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1184 (DingaLing/new) State Change CS_INIT -> CS_ROUTING 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal DingaLing/new [BREAK] 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:330 (DingaLing/new) State INIT going to sleep 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 (DingaLing/new) Running State Change CS_ROUTING 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 (DingaLing/new) State ROUTING 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1198 DingaLing/new CHANNEL ROUTING 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:78 DingaLing/new Standard ROUTING 2009-11-05 09:25:58.370659 [INFO] mod_dialplan_xml.c:391 Processing support.voip at gmail.com/gmail.B8861D13->john in context public Dialplan: DingaLing/new parsing [public->unloop] continue=false Dialplan: DingaLing/new Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: DingaLing/new Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: DingaLing/new parsing [public->outside_call] continue=true Dialplan: DingaLing/new Absolute Condition [outside_call] Dialplan: DingaLing/new Action set(outside_call=true) Dialplan: DingaLing/new parsing [public->call_debug] continue=true Dialplan: DingaLing/new Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: DingaLing/new parsing [public->public_extensions] continue=false Dialplan: DingaLing/new Regex (FAIL) [public_extensions] destination_number(john) =~ /^(10[01][0-9])$/ break=on-false Dialplan: DingaLing/new parsing [public->public_did] continue=false Dialplan: DingaLing/new Regex (FAIL) [public_did] destination_number(john) =~ /^(5551212)$/ break=on-false 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:114 (DingaLing/new) State Change CS_ROUTING -> CS_EXECUTE 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal DingaLing/new [BREAK] 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 (DingaLing/new) State ROUTING going to sleep 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 (DingaLing/new) Running State Change CS_EXECUTE 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:340 (DingaLing/new) State EXECUTE 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1215 DingaLing/new CHANNEL EXECUTE 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:151 DingaLing/new Standard EXECUTE EXECUTE DingaLing/new set(outside_call=true) 2009-11-05 09:25:58.381289 [DEBUG] mod_dptools.c:752 DingaLing/new SET [outside_call]=[true] 2009-11-05 09:25:58.381289 [NOTICE] switch_core_state_machine.c:179 Hangup DingaLing/new [CS_EXECUTE] [NORMAL_CLEARING] 2009-11-05 09:25:58.381289 [DEBUG] switch_channel.c:1837 Send signal DingaLing/new [KILL] 2009-11-05 09:25:58.381289 [DEBUG] libdingaling.c:298 Destroyed Session c1722311748 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.381289 [DEBUG] switch_core_session.c:969 Send signal DingaLing/new [BREAK] 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 (DingaLing/new) State HANGUP 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1293 DingaLing/new CHANNEL HANGUP 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:46 DingaLing/new Standard HANGUP, cause: NORMAL_CLEARING 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 (DingaLing/new) State HANGUP going to sleep 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:340 (DingaLing/new) State EXECUTE going to sleep 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 (DingaLing/new) Running State Change CS_HANGUP 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:454 handler already called, skipping state handler. 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:325 (DingaLing/new) State Change CS_HANGUP -> CS_REPORTING 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal DingaLing/new [BREAK] 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 (DingaLing/new) Running State Change CS_REPORTING 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 (DingaLing/new) State REPORTING 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:53 DingaLing/new Standard REPORTING, cause: NORMAL_CLEARING 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 (DingaLing/new) State REPORTING going to sleep 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:319 (DingaLing/new) State Change CS_REPORTING -> CS_DESTROY 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal DingaLing/new [BREAK] 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:1106 Session 1 (DingaLing/new) Locked, Waiting on external entities 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1124 Session 1 (DingaLing/new) Ended 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1126 Close Channel DingaLing/new [CS_DESTROY] 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:413 (DingaLing/new) Running State Change CS_DESTROY 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 (DingaLing/new) State DESTROY 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1231 NUKE RTP 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:60 DingaLing/new Standard DESTROY 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 (DingaLing/new) State DESTROY going to sleep 2009-11-05 09:25:58.459764 [DEBUG] libdingaling.c:1389 Processing 3 packets in retry queue Thanks 2009/11/4 Chris Chen > you have to define the extension "john" or "bob" or whatever number you > want in the dialplan for the context "public". > > Just follow your jingle profile you define. Simple, no other tricks. > > Thanks, > Chris > > On Wed, Nov 4, 2009 at 11:56 AM, Jonathan Barou wrote: > >> Hi everybody, >> >> I actually working on mod_dingaling (gtalk). I can make call from FS to >> Gtalk, and from Gtalk to FS. >> But I have a problem, in jingle_profile I have a file like this : >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> here when I put an user account like >> john or bob its doesn't work whereas I put something like 1000 or 8400 it >> works. >> >> When I tried to put a real phone number It doesn't work too (I have a >> gateway with my PBX). >> >> Somebody know, why it doesn't work with name and work with number ? >> >> Thanks. >> >> >> -- >> Jonathan BAROU >> SQLI LYON - CRCI >> 0472405368 >> jbarou at sqli.com >> lyon.crci at sqli.com >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jonathan BAROU SQLI LYON - CRCI 0472405368 jbarou at sqli.com lyon.crci at sqli.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/fc933f09/attachment-0002.html From oseslija at gmail.com Thu Nov 5 01:33:02 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Thu, 5 Nov 2009 10:33:02 +0100 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <960738410911042333x1e49719fu34f353bf015c9d95@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> <960738410911042333x1e49719fu34f353bf015c9d95@mail.gmail.com> Message-ID: <4468a6770911050133s1f3f1596n1b0f3e17b291080a@mail.gmail.com> Hello, you need to enable Both Reg option for E.164 Registration for extensions (you can do that via CCME's web interface in Configure->Extensions->Extension ). This will make the phone register to SCCP server and Cisco sending SIP REGISTERs to the proxy/trunk configured in the same time phone regs to SCCP. This is valid for subsequent registrations, also. Regards, Ognjen On Thu, Nov 5, 2009 at 8:33 AM, Mathieu Parent wrote: > Hello, > > > On Wed, Nov 4, 2009 at 11:38 PM, Ognjen Seslija > wrote: > > Hello, > > > > I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS. > > Phones are registering to CCME and FS simultaneously. So far, everything > is > > working just fine. > > Great. I'm interrested by this simultaneous registering. Do you havve some > docs? > > Mathieu Parent > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/a13f86ac/attachment-0002.html From oseslija at gmail.com Thu Nov 5 01:27:29 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Thu, 5 Nov 2009 10:27:29 +0100 Subject: [Freeswitch-users] FS and Skinny (SCCP) In-Reply-To: <191c3a030911042155i64faeedmbd72d4fc5f51ca45@mail.gmail.com> References: <63de75710911030952n2141e584idc60ea74056a9d4b@mail.gmail.com> <4468a6770911041438v168ce3a6g799562c32628a8c1@mail.gmail.com> <191c3a030911042155i64faeedmbd72d4fc5f51ca45@mail.gmail.com> Message-ID: <4468a6770911050127p69f8b633i45e72ab9741b3b2c@mail.gmail.com> Exactly right, Tony. Regards, Ognjen On Thu, Nov 5, 2009 at 6:55 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > but if he wants to code it we wouldn't mind it right? =p > > > > On Wed, Nov 4, 2009 at 4:38 PM, Ognjen Seslija wrote: > >> Hello, >> >> I have CCME 4.1 on 2691 doing SCCP to five 7941 phones with SIP to FS. >> Phones are registering to CCME and FS simultaneously. So far, everything is >> working just fine. >> >> I think SCCP will be obsolete in the future since even Cisco is working >> more and more on SIP. OTOH, I really hate SIP images for 79x1 (that's why I >> put CCME in the first place), where as ones for 79x0 are behaving much >> better. >> >> Regards, >> Ognjen >> >> >> >> On Tue, Nov 3, 2009 at 6:52 PM, mm_202 wrote: >> >>> FS doesnt support SCCP (from what I gathered, just because no one has >>> bothered coding it). >>> >>> Are there other users out there has use SCCP and FS? (with some >>> middleware in between) >>> >>> If enough people would find a use for it, I'd be willing to actually >>> code it (esp if someone offered a bounty). >>> So, would anyone besides me want/use a SCCP endpoint in FS? >>> >>> -- mm_202. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/dce49285/attachment-0002.html From rupa at rupa.com Thu Nov 5 05:17:58 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 5 Nov 2009 07:17:58 -0600 Subject: [Freeswitch-users] problem with failover routes for LCR A-Z scenario In-Reply-To: <727374.16142.qm@web111917.mail.gq1.yahoo.com> References: <727374.16142.qm@web111917.mail.gq1.yahoo.com> Message-ID: Now that user rates are supported in mod_lcr, how about an option that says to drop the route if the user_rate is < rate ? This 1) requires you to use custom sql and 2) be able to represent your user rates in that sql (join to user rate table perhaps?) On Thu, Nov 5, 2009 at 1:44 AM, Hound Dog wrote: > I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) > > it came out a little long , sorry for that :) > > > I found that it difficult/impossible to make LCR use only carriers that I choose > > scenario is as follows , taking the UK as example for a destination ?( prices are not real , just an example ) > > I have 2 carriers offering routes to the UK , landline and mobile > > my buying prices > > Destination ? ? ? carrier1 Price ? ?carrier2 price > 44 ?(all UK) ? ? ?$0.01 ? ? ? ? ? ? $0.01 > 447 (UK mobile) ? $0.15 ? ? ? ? ? ? $0.19 > > my selling prices > > Destination ? ? ? ? price > 44 ?(all UK) ? ? ? ?$0.015 > 447 (UKmobile) ? ? ?$0.17 > > so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable > > for UK mobile I can ** only ** make a profit if I use carrier 1 ? ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) > > > translating it to MOD_LCR information > > digits ? ? rate ? ? ? ?carrier_id ? ? ? ( other columns ignored ) > 44 ? ? ? ? 0.01 ? ? ? ?1 > 44 ? ? ? ? 0.01 ? ? ? ?2 > 447 ? ? ? ?0.015 ? ? ? 1 > > this looks good : > ? ? 44 prefix will be shared between carrier 1 and 2 > ? ? 447 prefix will only go to carreir 1 > > so it fits perfectly - BUT > > testing this I get - > > API CALL [lcr(447965404547)] output: > ?| Digit Match | Carrier | Rate ? ? | Codec | CID Regexp | Dialstring | > ?| 447 ? ? ? ? | carr1 ? | 0.15 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.1 | > ?| 44 ? ? ? ? ?| carr2 ? | 0.01 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.2 | > > Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) ?- it all makes sense > > > ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail > > > so far I didnt find a solution for that , so if there is one I love bo pointed there > > > > > > > I did think it over a little and came up with 2 options that could be used , > ?and I am also planning to code them and propose patch to maintainers , > ?I would love to get comments on those ( in case there are no existing solution ) > > > option 1 - setting some routes as last option , add another param to the LCR table called ?last_route , > ?when hitting a route with last_route=1, ?stop processing additional routes and return your routing decision so far > ?so in our case the route entry with 44 to carrier1 will have last_route=1 ?, the other 44 routes will have last_route=0 to allow for failover > > option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. > in other words dont failover from a finer route to a wider route. > it will need to be a global option and I will be quite simple to use, > ?but will require entering mutiple entries of the same length prefix for each carrier you would like to use > ?its intutive and relatively simple to manage , but requires more lcr entries to get you where you want > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From testeador01 at gmail.com Thu Nov 5 05:18:48 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 5 Nov 2009 08:18:48 -0500 Subject: [Freeswitch-users] Question about jingle_profiles In-Reply-To: <8048ff7f0911050040n791b59efp33dcc4a4236c71ca@mail.gmail.com> References: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> <507898380911041145u431865f8uc8877fce3c2e3778@mail.gmail.com> <8048ff7f0911050040n791b59efp33dcc4a4236c71ca@mail.gmail.com> Message-ID: Hello :) if you look at this line 2009-11-05 09:25:58.370659 [INFO] mod_dialplan_xml.c:391 Processing > support.voip at gmail.com/gmail.B8861D13->john in context public you will see that it is looking for john in the public extensions (freeswitch/conf/dialplan/public.xml). The reason why it finds 1000 is because of this portion in public.xml: You need to create a new extension on the public context for "john" or "bob" or whatever other name you want to be able to contact from the public context. I hope this answers your question. 2009/11/5 Jonathan Barou > Hi, > > In the dialplan I have the extension "Local_extension" with "john" and it's > working when I call john from the account 1000 with softphone. > > When I try to make a call from Gtalk to FS I have that in the console : > > > > 09-11-05 09:25:58.370659 [DEBUG] switch_rtp.c:2780 Activate VAD codec PCMU > 20ms > > 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1184 (DingaLing/new) > State Change CS_INIT -> CS_ROUTING > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal > DingaLing/new [BREAK] > > 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:330 > (DingaLing/new) State INIT going to sleep > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 > (DingaLing/new) Running State Change CS_ROUTING > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 > (DingaLing/new) State ROUTING > > 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1198 DingaLing/new > CHANNEL ROUTING > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:78 > DingaLing/new Standard ROUTING > > 2009-11-05 09:25:58.370659 [INFO] mod_dialplan_xml.c:391 Processing > support.voip at gmail.com/gmail.B8861D13->john in context public > > Dialplan: DingaLing/new parsing [public->unloop] continue=false > > Dialplan: DingaLing/new Regex (PASS) [unloop] ${unroll_loops}(true) =~ > /^true$/ break=on-false > > Dialplan: DingaLing/new Regex (FAIL) [unloop] ${sip_looped_call}() =~ > /^true$/ break=on-false > > Dialplan: DingaLing/new parsing [public->outside_call] continue=true > > Dialplan: DingaLing/new Absolute Condition [outside_call] > > Dialplan: DingaLing/new Action set(outside_call=true) > > Dialplan: DingaLing/new parsing [public->call_debug] continue=true > > Dialplan: DingaLing/new Regex (FAIL) [call_debug] ${call_debug}(false) =~ > /^true$/ break=never > > Dialplan: DingaLing/new parsing [public->public_extensions] continue=false > > Dialplan: DingaLing/new Regex (FAIL) [public_extensions] > destination_number(john) =~ /^(10[01][0-9])$/ break=on-false > > Dialplan: DingaLing/new parsing [public->public_did] continue=false > > Dialplan: DingaLing/new Regex (FAIL) [public_did] destination_number(john) > =~ /^(5551212)$/ break=on-false > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:114 > (DingaLing/new) State Change CS_ROUTING -> CS_EXECUTE > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal > DingaLing/new [BREAK] > > 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 > (DingaLing/new) State ROUTING going to sleep > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 > (DingaLing/new) Running State Change CS_EXECUTE > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:340 > (DingaLing/new) State EXECUTE > > 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1215 DingaLing/new > CHANNEL EXECUTE > > 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:151 > DingaLing/new Standard EXECUTE > > EXECUTE DingaLing/new set(outside_call=true) > > 2009-11-05 09:25:58.381289 [DEBUG] mod_dptools.c:752 DingaLing/new SET > [outside_call]=[true] > > 2009-11-05 09:25:58.381289 [NOTICE] switch_core_state_machine.c:179 Hangup > DingaLing/new [CS_EXECUTE] [NORMAL_CLEARING] > > 2009-11-05 09:25:58.381289 [DEBUG] switch_channel.c:1837 Send signal > DingaLing/new [KILL] > > 2009-11-05 09:25:58.381289 [DEBUG] libdingaling.c:298 Destroyed Session > c1722311748 > > > 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.381289 [DEBUG] switch_core_session.c:969 Send signal > DingaLing/new [BREAK] > > 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 > (DingaLing/new) State HANGUP > > 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1293 DingaLing/new > CHANNEL HANGUP > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:46 > DingaLing/new Standard HANGUP, cause: NORMAL_CLEARING > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 > (DingaLing/new) State HANGUP going to sleep > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:340 > (DingaLing/new) State EXECUTE going to sleep > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 > (DingaLing/new) Running State Change CS_HANGUP > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:454 handler > already called, skipping state handler. > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:325 > (DingaLing/new) State Change CS_HANGUP -> CS_REPORTING > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal > DingaLing/new [BREAK] > > 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 > (DingaLing/new) Running State Change CS_REPORTING > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 > (DingaLing/new) State REPORTING > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:53 > DingaLing/new Standard REPORTING, cause: NORMAL_CLEARING > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 > (DingaLing/new) State REPORTING going to sleep > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:319 > (DingaLing/new) State Change CS_REPORTING -> CS_DESTROY > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal > DingaLing/new [BREAK] > > 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new > CHANNEL KILL > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:1106 Session 1 > (DingaLing/new) Locked, Waiting on external entities > > 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1124 Session 1 > (DingaLing/new) Ended > > 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1126 Close > Channel DingaLing/new [CS_DESTROY] > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:413 > (DingaLing/new) Running State Change CS_DESTROY > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 > (DingaLing/new) State DESTROY > > 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1231 NUKE RTP > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:60 > DingaLing/new Standard DESTROY > > 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 > (DingaLing/new) State DESTROY going to sleep > > 2009-11-05 09:25:58.459764 [DEBUG] libdingaling.c:1389 Processing 3 packets > in retry queue > > Thanks > > > > 2009/11/4 Chris Chen > > you have to define the extension "john" or "bob" or whatever number you >> want in the dialplan for the context "public". >> >> Just follow your jingle profile you define. Simple, no other tricks. >> >> Thanks, >> Chris >> >> On Wed, Nov 4, 2009 at 11:56 AM, Jonathan Barou wrote: >> >>> Hi everybody, >>> >>> I actually working on mod_dingaling (gtalk). I can make call from FS to >>> Gtalk, and from Gtalk to FS. >>> But I have a problem, in jingle_profile I have a file like this : >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> here when I put an user account like >>> john or bob its doesn't work whereas I put something like 1000 or 8400 it >>> works. >>> >>> When I tried to put a real phone number It doesn't work too (I have a >>> gateway with my PBX). >>> >>> Somebody know, why it doesn't work with name and work with number ? >>> >>> Thanks. >>> >>> >>> -- >>> Jonathan BAROU >>> SQLI LYON - CRCI >>> 0472405368 >>> jbarou at sqli.com >>> lyon.crci at sqli.com >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Jonathan BAROU > SQLI LYON - CRCI > 0472405368 > jbarou at sqli.com > lyon.crci at sqli.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/abcfb106/attachment-0002.html From nagalenoj at gmail.com Thu Nov 5 05:28:22 2009 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Thu, 5 Nov 2009 18:58:22 +0530 Subject: [Freeswitch-users] Filtering a particular event. Message-ID: Hi, I've tried to filter the events like below to filter a particular event. 1) register for all events 2) filter for one unique-id 3) filter only one/more events(ex: DTMF & CHANNEL_EXECUTE) So, I want to receive only these events for the specific unique-id. But, I am receiving other events too. I'm using perl ESL outbound. Is it possible to do like this?! -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/f2cd0c29/attachment-0002.html From rupa at rupa.com Thu Nov 5 05:31:00 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 5 Nov 2009 07:31:00 -0600 Subject: [Freeswitch-users] problem with failover routes for LCR A-Z scenario In-Reply-To: References: <727374.16142.qm@web111917.mail.gq1.yahoo.com> Message-ID: Actually, using custom sql, you can implement the filter yourself in the where clause. No need for code changes. On Thu, Nov 5, 2009 at 7:17 AM, Rupa Schomaker wrote: > Now that user rates are supported in mod_lcr, how about an option that > says to drop the route if the user_rate is < rate ? > > This 1) requires you to use custom sql and 2) be able to represent > your user rates in that sql (join to user rate table perhaps?) > > On Thu, Nov 5, 2009 at 1:44 AM, Hound Dog wrote: >> I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) >> >> it came out a little long , sorry for that :) >> >> >> I found that it difficult/impossible to make LCR use only carriers that I choose >> >> scenario is as follows , taking the UK as example for a destination ?( prices are not real , just an example ) >> >> I have 2 carriers offering routes to the UK , landline and mobile >> >> my buying prices >> >> Destination ? ? ? carrier1 Price ? ?carrier2 price >> 44 ?(all UK) ? ? ?$0.01 ? ? ? ? ? ? $0.01 >> 447 (UK mobile) ? $0.15 ? ? ? ? ? ? $0.19 >> >> my selling prices >> >> Destination ? ? ? ? price >> 44 ?(all UK) ? ? ? ?$0.015 >> 447 (UKmobile) ? ? ?$0.17 >> >> so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable >> >> for UK mobile I can ** only ** make a profit if I use carrier 1 ? ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) >> >> >> translating it to MOD_LCR information >> >> digits ? ? rate ? ? ? ?carrier_id ? ? ? ( other columns ignored ) >> 44 ? ? ? ? 0.01 ? ? ? ?1 >> 44 ? ? ? ? 0.01 ? ? ? ?2 >> 447 ? ? ? ?0.015 ? ? ? 1 >> >> this looks good : >> ? ? 44 prefix will be shared between carrier 1 and 2 >> ? ? 447 prefix will only go to carreir 1 >> >> so it fits perfectly - BUT >> >> testing this I get - >> >> API CALL [lcr(447965404547)] output: >> ?| Digit Match | Carrier | Rate ? ? | Codec | CID Regexp | Dialstring | >> ?| 447 ? ? ? ? | carr1 ? | 0.15 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.1 | >> ?| 44 ? ? ? ? ?| carr2 ? | 0.01 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.2 | >> >> Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) ?- it all makes sense >> >> >> ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail >> >> >> so far I didnt find a solution for that , so if there is one I love bo pointed there >> >> >> >> >> >> >> I did think it over a little and came up with 2 options that could be used , >> ?and I am also planning to code them and propose patch to maintainers , >> ?I would love to get comments on those ( in case there are no existing solution ) >> >> >> option 1 - setting some routes as last option , add another param to the LCR table called ?last_route , >> ?when hitting a route with last_route=1, ?stop processing additional routes and return your routing decision so far >> ?so in our case the route entry with 44 to carrier1 will have last_route=1 ?, the other 44 routes will have last_route=0 to allow for failover >> >> option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. >> in other words dont failover from a finer route to a wider route. >> it will need to be a global option and I will be quite simple to use, >> ?but will require entering mutiple entries of the same length prefix for each carrier you would like to use >> ?its intutive and relatively simple to manage , but requires more lcr entries to get you where you want >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -Rupa > -- -Rupa From testeador01 at gmail.com Thu Nov 5 05:35:23 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 5 Nov 2009 08:35:23 -0500 Subject: [Freeswitch-users] Copy voicemail greeting In-Reply-To: <01a501ca5db4$cf0024b0$6d006e10$@com> References: <011501ca5d8c$415fc340$c41f49c0$@com> <01a501ca5db4$cf0024b0$6d006e10$@com> Message-ID: Use sqlite3: http://souptonuts.sourceforge.net/readme_sqlite_tutorial.html [?] 2009/11/4 Lars Zeb > What tool/GUI do you use to edit the db contents? > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Wednesday, November 04, 2009 12:25 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Copy voicemail greeting > > > > copy the wav file and insert the record. > > > > /b > > > > On Nov 4, 2009, at 2:20 PM, Lars Zeb wrote: > > > > Is it possible to copy an existing wav greeting from one extension to > another? I think something has to be added to db/voicemail_default.db, but > it?s not a text file. > > > > Is it just easier to re-record the message from the 2nd extension? > > > > Thanks, Lars > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/ccc317b9/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 96 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/ccc317b9/attachment-0002.gif From jbarou at sqli.com Thu Nov 5 05:37:31 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Thu, 5 Nov 2009 14:37:31 +0100 Subject: [Freeswitch-users] Question about jingle_profiles In-Reply-To: References: <8048ff7f0911040856m5eb8eb88o12319fd1b1647914@mail.gmail.com> <507898380911041145u431865f8uc8877fce3c2e3778@mail.gmail.com> <8048ff7f0911050040n791b59efp33dcc4a4236c71ca@mail.gmail.com> Message-ID: <8048ff7f0911050537n2211fbf8n8f09d49168eb21f9@mail.gmail.com> Wonderful, thank you very much Milena. 2009/11/5 Milena > Hello :) > if you look at this line > > 2009-11-05 09:25:58.370659 [INFO] mod_dialplan_xml.c:391 Processing >> support.voip at gmail.com/gmail.B8861D13->john in context public > > > you will see that it is looking for john in the public extensions > (freeswitch/conf/dialplan/public.xml). > > The reason why it finds 1000 is because of this portion in public.xml: > > > > > > > You need to create a new extension on the public context for "john" or > "bob" or whatever other name you want to be able to contact from the public > context. > > I hope this answers your question. > > > 2009/11/5 Jonathan Barou > > Hi, >> >> In the dialplan I have the extension "Local_extension" with "john" and >> it's working when I call john from the account 1000 with softphone. >> >> When I try to make a call from Gtalk to FS I have that in the console : >> >> >> >> 09-11-05 09:25:58.370659 [DEBUG] switch_rtp.c:2780 Activate VAD codec PCMU >> 20ms >> >> 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1184 (DingaLing/new) >> State Change CS_INIT -> CS_ROUTING >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal >> DingaLing/new [BREAK] >> >> 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:330 >> (DingaLing/new) State INIT going to sleep >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 >> (DingaLing/new) Running State Change CS_ROUTING >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 >> (DingaLing/new) State ROUTING >> >> 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1198 DingaLing/new >> CHANNEL ROUTING >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:78 >> DingaLing/new Standard ROUTING >> >> 2009-11-05 09:25:58.370659 [INFO] mod_dialplan_xml.c:391 Processing >> support.voip at gmail.com/gmail.B8861D13->john in context public >> >> Dialplan: DingaLing/new parsing [public->unloop] continue=false >> >> Dialplan: DingaLing/new Regex (PASS) [unloop] ${unroll_loops}(true) =~ >> /^true$/ break=on-false >> >> Dialplan: DingaLing/new Regex (FAIL) [unloop] ${sip_looped_call}() =~ >> /^true$/ break=on-false >> >> Dialplan: DingaLing/new parsing [public->outside_call] continue=true >> >> Dialplan: DingaLing/new Absolute Condition [outside_call] >> >> Dialplan: DingaLing/new Action set(outside_call=true) >> >> Dialplan: DingaLing/new parsing [public->call_debug] continue=true >> >> Dialplan: DingaLing/new Regex (FAIL) [call_debug] ${call_debug}(false) =~ >> /^true$/ break=never >> >> Dialplan: DingaLing/new parsing [public->public_extensions] continue=false >> >> Dialplan: DingaLing/new Regex (FAIL) [public_extensions] >> destination_number(john) =~ /^(10[01][0-9])$/ break=on-false >> >> Dialplan: DingaLing/new parsing [public->public_did] continue=false >> >> Dialplan: DingaLing/new Regex (FAIL) [public_did] destination_number(john) >> =~ /^(5551212)$/ break=on-false >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:114 >> (DingaLing/new) State Change CS_ROUTING -> CS_EXECUTE >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_session.c:969 Send signal >> DingaLing/new [BREAK] >> >> 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:333 >> (DingaLing/new) State ROUTING going to sleep >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:306 >> (DingaLing/new) Running State Change CS_EXECUTE >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:340 >> (DingaLing/new) State EXECUTE >> >> 2009-11-05 09:25:58.370659 [DEBUG] mod_dingaling.c:1215 DingaLing/new >> CHANNEL EXECUTE >> >> 2009-11-05 09:25:58.370659 [DEBUG] switch_core_state_machine.c:151 >> DingaLing/new Standard EXECUTE >> >> EXECUTE DingaLing/new set(outside_call=true) >> >> 2009-11-05 09:25:58.381289 [DEBUG] mod_dptools.c:752 DingaLing/new SET >> [outside_call]=[true] >> >> 2009-11-05 09:25:58.381289 [NOTICE] switch_core_state_machine.c:179 Hangup >> DingaLing/new [CS_EXECUTE] [NORMAL_CLEARING] >> >> 2009-11-05 09:25:58.381289 [DEBUG] switch_channel.c:1837 Send signal >> DingaLing/new [KILL] >> >> 2009-11-05 09:25:58.381289 [DEBUG] libdingaling.c:298 Destroyed Session >> c1722311748 >> >> >> 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.381289 [DEBUG] switch_core_session.c:969 Send signal >> DingaLing/new [BREAK] >> >> 2009-11-05 09:25:58.381289 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 >> (DingaLing/new) State HANGUP >> >> 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1293 DingaLing/new >> CHANNEL HANGUP >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:46 >> DingaLing/new Standard HANGUP, cause: NORMAL_CLEARING >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:476 >> (DingaLing/new) State HANGUP going to sleep >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:340 >> (DingaLing/new) State EXECUTE going to sleep >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 >> (DingaLing/new) Running State Change CS_HANGUP >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:454 handler >> already called, skipping state handler. >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:325 >> (DingaLing/new) State Change CS_HANGUP -> CS_REPORTING >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal >> DingaLing/new [BREAK] >> >> 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:306 >> (DingaLing/new) Running State Change CS_REPORTING >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 >> (DingaLing/new) State REPORTING >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:53 >> DingaLing/new Standard REPORTING, cause: NORMAL_CLEARING >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:567 >> (DingaLing/new) State REPORTING going to sleep >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:319 >> (DingaLing/new) State Change CS_REPORTING -> CS_DESTROY >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:969 Send signal >> DingaLing/new [BREAK] >> >> 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1333 DingaLing/new >> CHANNEL KILL >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_session.c:1106 Session 1 >> (DingaLing/new) Locked, Waiting on external entities >> >> 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1124 Session 1 >> (DingaLing/new) Ended >> >> 2009-11-05 09:25:58.390206 [NOTICE] switch_core_session.c:1126 Close >> Channel DingaLing/new [CS_DESTROY] >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:413 >> (DingaLing/new) Running State Change CS_DESTROY >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 >> (DingaLing/new) State DESTROY >> >> 2009-11-05 09:25:58.390206 [DEBUG] mod_dingaling.c:1231 NUKE RTP >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:60 >> DingaLing/new Standard DESTROY >> >> 2009-11-05 09:25:58.390206 [DEBUG] switch_core_state_machine.c:424 >> (DingaLing/new) State DESTROY going to sleep >> >> 2009-11-05 09:25:58.459764 [DEBUG] libdingaling.c:1389 Processing 3 >> packets in retry queue >> >> Thanks >> >> >> >> 2009/11/4 Chris Chen >> >> you have to define the extension "john" or "bob" or whatever number you >>> want in the dialplan for the context "public". >>> >>> Just follow your jingle profile you define. Simple, no other tricks. >>> >>> Thanks, >>> Chris >>> >>> On Wed, Nov 4, 2009 at 11:56 AM, Jonathan Barou wrote: >>> >>>> Hi everybody, >>>> >>>> I actually working on mod_dingaling (gtalk). I can make call from FS to >>>> Gtalk, and from Gtalk to FS. >>>> But I have a problem, in jingle_profile I have a file like this : >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> here when I put an user account like >>>> john or bob its doesn't work whereas I put something like 1000 or 8400 it >>>> works. >>>> >>>> When I tried to put a real phone number It doesn't work too (I have a >>>> gateway with my PBX). >>>> >>>> Somebody know, why it doesn't work with name and work with number ? >>>> >>>> Thanks. >>>> >>>> >>>> -- >>>> Jonathan BAROU >>>> SQLI LYON - CRCI >>>> 0472405368 >>>> jbarou at sqli.com >>>> lyon.crci at sqli.com >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Jonathan BAROU >> SQLI LYON - CRCI >> 0472405368 >> jbarou at sqli.com >> lyon.crci at sqli.com >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jonathan BAROU SQLI LYON - CRCI 0472405368 jbarou at sqli.com lyon.crci at sqli.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/7587c615/attachment-0002.html From anthony.minessale at gmail.com Thu Nov 5 06:13:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Nov 2009 08:13:46 -0600 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: References: Message-ID: <191c3a030911050613o4a0cd40gcef5163574189561@mail.gmail.com> I did not ask you to send me a ladder diagram. I asked you to send me a console trace from FreeSWITCH using latest trunk (1.0.4 does not help me) 1) start FreeSWITCH 2) run the cli command: console loglevel debug 3) run the cli command: sofia profile internal siptrace on 4) reproduce your issue and put the trace on freeswitch pastebin http://pastebin.freeswitch.org (login and pass are stated in the auth dialog) Also please answer brian's question. What phones and/or sip devices are involved in this call. On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana wrote: > > Thanks for your time, > > -The scenario is still the same: > > Always bypass media. > Environment 100% NAT free :-) > Call established from A to B through FS. Then... > Blind transfer from B to C (Refer-to: C) > RTP should go directly between A and C. > > > -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the > REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the > lack of reINVITE to A, after C answers). > > Please check SIP diagram here: > > http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html > > > -What it's wrong with r15332 is there is not such call to C. For sure I > know SIP is a protocol, may be my description was not clear but this SIP > diagram speaks by itself ;-) > > http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html > > > -You could check the sofia debug for r15332 here: > http://pastebin.com/m6f2b3836 > > > Best regards, > > Humberto > > > > > I don't know what you are talking about anymore. > > > > The scenario I had tested is when a call is bridged in bypass_media=true > > bridge > > and you blind transfer that call back to the dialplan > > > > as soon as it hits the routing state it will resume media. > > > > > > it has been confirmed to not work and confirmed to have been fixed > several > > time and if you are still having a problem you must have something > blocking > > some of your packets or something . > > > > You have to understand that sip is a protocol and your description is > > completely non-standard. > > Perhaps you should get a console trace and attach it to a jira. The trace > > probably makes more sense to me. > > > > sofia profile internal siptrace on > > console loglevel debug > > > > reproduce and attach the whole capture. > > > > > > > > On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > > > >> > >> Hi, > >> > >> I tried r15332 and set in the sofia profile: > >> > >> a) bypass_media_after_bridge=true only > >> b) bypass_media_after_bridge=true, param name="media-option" > >> value="resume-media-on-hold"/> > >> > >> > >> In both cases FS is hanging up the initial call (A to FS) after > accepting > >> the REFER to C: > >> > >> A <- reINVITE with FS' SDP <- FS > >> A -> 200 -> FS > >> A <- ACK <- FS > >> A <- BYE <- FS > >> > >> The call to C is not even tried. > >> > >> I found this line is the logs that could give some idea: > >> > >> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup > >> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] > [RECOVERY_ON_TIMER_EXPIRE] > >> after sending the ACK for the reINVITE > >> > >> > >> Regards, > >> > >> > >> Humberto > >> > >>>please try r15326 > >>>I think i have it working. > >>> > >>>I recommend for optimal results you set bypass_media_after_bridge=true > >>>either as a global or in your DP in place of bypass_media=true > >>> > >>> > >>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana > >> hotmail.com>wrote: > >>> > >>>> Hi Mike, > >>>> > >>>> I re-tried with trunk rev 15319 but I got almost the same behavior: > >> There > >>>> is now a reINVITE (with FS' SDP) going to A when the REFER is > accepted. > >> But > >>>> still there is no reINVITE for A (with C's SDP) after the call from FS > >> to C > >>>> is established. > >>>> > >>>> Anyway, we decided for now to do a different implementation but if you > >> want > >>>> to explore more in this issue count me in ;-) > >>>> > >>>> > >>>> Thank you very much! > >>>> > >>>> Humberto > >> > >> > >> _________________________________________________________________ > >> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they > >> e-mail you. > >> http://go.microsoft.com/?linkid=9691817 > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > _________________________________________________________________ > > Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows > 7 now > > http://go.microsoft.com/?linkid=9691818 > > _________________________________________________________________ > Windows Live: Make it easier for your friends to see what you?re up to on > Facebook. > http://go.microsoft.com/?linkid=9691816 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/cfff9439/attachment-0002.html From rob4manhere at gmail.com Thu Nov 5 06:52:05 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 5 Nov 2009 08:52:05 -0600 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> Message-ID: <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> Hi UK, From what I've done and read, the caller-controls (in conference.conf.xml) can be modified to almost anything you can think of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a caller control just for those with the moderator flag. So unless you want everyone able to mute/kick everyone then you can't do it. The wiki seems to indicate this as well: "Be aware that the caller-controls are applied across the entire conference. You cannot enter one member of the conference using caller- controls ABC and then enter a second member using caller-controls XYZ." http://wiki.freeswitch.org/wiki/Mod_conference I think this might be a limitation of mod_conference. Perhaps one of the pros can chime in if I'm off-base or there's some nifty way to accomplish this. Cheers, Rob On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > Any ideas on the below...has anyone implemented the below: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Ujjval Karihaloo > Sent: Monday, November 02, 2009 12:52 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Rob: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 9:34 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hm, strange. I haven't seen that before. Can you pastebin your logs > at debug level? > > On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > >> It's strange... a tcpdump tells me that there is no DTMF from my >> provider when using IVR, but when I call into a TN that goes >> directly into the Conference App, I see DTMF from the provider. >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 7:23 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I've never had any problem with that. Is your logging at debug level >> so you can see the RECV DTFM in the log/fs_cli? Are you calling from >> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >> isn't passing them through. >> >> Make sure your logging is turned up then try something simpler, like >> calling the echo application, and see if DTFM comes through. >> >> Rob >> >> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >> >>> Rob: >>> >>> For some reason, I don't see the DTMF appear on the fs_CLI when >>> using the below configuration....so it basically timesout. >>> >>> UK >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, October 26, 2009 9:21 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Thx a lot Rob, reading the wiki your way or using IVR seems >>> correct.. >>> =============== >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> ====================== >>> >>> I will try it out. >>> >>> Again thx a lot for your help. Will keep everyone posted. >>> >>> ________________________________________ >>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>> Sent: Friday, October 23, 2009 12:22 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I just re-tested with the pin in my dial plan: >>> >>> >>> >>> And it doesn't challenge me for the pin. I just drop right in. I >>> figured this is how it was intended, since the wiki says the pin is >>> set initially and only challenged in later attempts [by future >>> callers]: >>> >>> "The first time a conference name (confname) is used, it will be >>> created on demand, and the pin will be set to what ever is specified >>> at that time: the pin in the data string if specified, or if not, >>> the >>> "pin" setting in the conference profile, and if that is also >>> unspecified, then there is no pin protection. Any later attempt to >>> join the conference must specify the same pin number, if one existed >>> when it was created. " >>> >>> >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> >>> I'm sure there's a scenario where its used and useful, the wiki just >>> doesn't explain it. >>> >>> Rob >>> >>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>> >>>> Well first off you're not defining a pine here... >>>> >>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>> pin >>>> number] >>>> >>>> That might be why its not asking for a pin. >>>> >>>> /b >>>> >>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rob4manhere at gmail.com Thu Nov 5 06:57:08 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 5 Nov 2009 08:57:08 -0600 Subject: [Freeswitch-users] Wideband / HD phones Message-ID: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> Hey all, Looking at buying some high def phones. Any recommendations (preferably based on experience) for hardware based on product quality, standards compliance, features integration with Freeswitch, etc? Thank you! Rob Forman From brian at freeswitch.org Thu Nov 5 07:07:09 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 09:07:09 -0600 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> Message-ID: <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> Polycom ip6000's or bust! /b On Nov 5, 2009, at 8:57 AM, Rob Forman wrote: > Hey all, > > Looking at buying some high def phones. Any recommendations > (preferably based on experience) for hardware based on product > quality, standards compliance, features integration with Freeswitch, > etc? > > Thank you! > Rob Forman From qinglan_zeng at hotmail.com Thu Nov 5 07:19:37 2009 From: qinglan_zeng at hotmail.com (=?gb2312?B?tPPE4MjL?=) Date: Thu, 5 Nov 2009 15:19:37 +0000 Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: Hi All, I once meet the Skypiax load error issue and some guys infomed me that there is no configuration file for Skypiax. When I follow these intructions -> http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances I still have some difficulties unstanding this: ." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml to Debug\conf\autoload_configs." I did not find such directories in my freeswitch folder. Did not understand what "src" means, I checked the freeswitch folder and did not find such a folder named"src". There is a folder named"mod" under freeswitch while look into "mod" folder there are only some DLL files and can not find endpoints and etc. 2.You'll probably build the "Debug" version I just build this from the precompiled binaries and then launched FS. I'm not sure what I launched is in debug mode or not. If anyone can offer some help that really be appriciated. Thanks Daniel Zeng From: freeswitch-users-request at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 41, Issue 40 To: freeswitch-users at lists.freeswitch.org Date: Thu, 5 Nov 2009 06:57:44 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --??????-- From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 5 Nov 2009 08:13:46 -0600 Subject: Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media I did not ask you to send me a ladder diagram. I asked you to send me a console trace from FreeSWITCH using latest trunk (1.0.4 does not help me) 1) start FreeSWITCH 2) run the cli command: console loglevel debug 3) run the cli command: sofia profile internal siptrace on 4) reproduce your issue and put the trace on freeswitch pastebin http://pastebin.freeswitch.org (login and pass are stated in the auth dialog) Also please answer brian's question. What phones and/or sip devices are involved in this call. On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana wrote: Thanks for your time, -The scenario is still the same: Always bypass media. Environment 100% NAT free :-) Call established from A to B through FS. Then... Blind transfer from B to C (Refer-to: C) RTP should go directly between A and C. -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the lack of reINVITE to A, after C answers). Please check SIP diagram here: http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html -What it's wrong with r15332 is there is not such call to C. For sure I know SIP is a protocol, may be my description was not clear but this SIP diagram speaks by itself ;-) http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html -You could check the sofia debug for r15332 here: http://pastebin.com/m6f2b3836 Best regards, Humberto > > I don't know what you are talking about anymore. > > The scenario I had tested is when a call is bridged in bypass_media=true > bridge > and you blind transfer that call back to the dialplan > > as soon as it hits the routing state it will resume media. > > > it has been confirmed to not work and confirmed to have been fixed several > time and if you are still having a problem you must have something blocking > some of your packets or something . > > You have to understand that sip is a protocol and your description is > completely non-standard. > Perhaps you should get a console trace and attach it to a jira. The trace > probably makes more sense to me. > > sofia profile internal siptrace on > console loglevel debug > > reproduce and attach the whole capture. > > > > On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > >> >> Hi, >> >> I tried r15332 and set in the sofia profile: >> >> a) bypass_media_after_bridge=true only >> b) bypass_media_after_bridge=true, param name="media-option" >> value="resume-media-on-hold"/> >> >> >> In both cases FS is hanging up the initial call (A to FS) after accepting >> the REFER to C: >> >> A <- reINVITE with FS' SDP <- FS >> A -> 200 -> FS >> A <- ACK <- FS >> A <- BYE <- FS >> >> The call to C is not even tried. >> >> I found this line is the logs that could give some idea: >> >> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] >> after sending the ACK for the reINVITE >> >> >> Regards, >> >> >> Humberto >> >>>please try r15326 >>>I think i have it working. >>> >>>I recommend for optimal results you set bypass_media_after_bridge=true >>>either as a global or in your DP in place of bypass_media=true >>> >>> >>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >> hotmail.com>wrote: >>> >>>> Hi Mike, >>>> >>>> I re-tried with trunk rev 15319 but I got almost the same behavior: >> There >>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. >> But >>>> still there is no reINVITE for A (with C's SDP) after the call from FS >> to C >>>> is established. >>>> >>>> Anyway, we decided for now to do a different implementation but if you >> want >>>> to explore more in this issue count me in ;-) >>>> >>>> >>>> Thank you very much! >>>> >>>> Humberto >> >> >> _________________________________________________________________ >> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they >> e-mail you. >> http://go.microsoft.com/?linkid=9691817 >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > _________________________________________________________________ > Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now > http://go.microsoft.com/?linkid=9691818 _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook. http://go.microsoft.com/?linkid=9691816 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 --??????-- From: rob4manhere at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 5 Nov 2009 08:52:05 -0600 Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hi UK, From what I've done and read, the caller-controls (in conference.conf.xml) can be modified to almost anything you can think of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a caller control just for those with the moderator flag. So unless you want everyone able to mute/kick everyone then you can't do it. The wiki seems to indicate this as well: "Be aware that the caller-controls are applied across the entire conference. You cannot enter one member of the conference using caller- controls ABC and then enter a second member using caller-controls XYZ." http://wiki.freeswitch.org/wiki/Mod_conference I think this might be a limitation of mod_conference. Perhaps one of the pros can chime in if I'm off-base or there's some nifty way to accomplish this. Cheers, Rob On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > Any ideas on the below...has anyone implemented the below: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Ujjval Karihaloo > Sent: Monday, November 02, 2009 12:52 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Rob: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 9:34 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hm, strange. I haven't seen that before. Can you pastebin your logs > at debug level? > > On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > >> It's strange... a tcpdump tells me that there is no DTMF from my >> provider when using IVR, but when I call into a TN that goes >> directly into the Conference App, I see DTMF from the provider. >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 7:23 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I've never had any problem with that. Is your logging at debug level >> so you can see the RECV DTFM in the log/fs_cli? Are you calling from >> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >> isn't passing them through. >> >> Make sure your logging is turned up then try something simpler, like >> calling the echo application, and see if DTFM comes through. >> >> Rob >> >> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >> >>> Rob: >>> >>> For some reason, I don't see the DTMF appear on the fs_CLI when >>> using the below configuration....so it basically timesout. >>> >>> UK >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, October 26, 2009 9:21 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Thx a lot Rob, reading the wiki your way or using IVR seems >>> correct.. >>> =============== >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> ====================== >>> >>> I will try it out. >>> >>> Again thx a lot for your help. Will keep everyone posted. >>> >>> ________________________________________ >>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>> Sent: Friday, October 23, 2009 12:22 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I just re-tested with the pin in my dial plan: >>> >>> >>> >>> And it doesn't challenge me for the pin. I just drop right in. I >>> figured this is how it was intended, since the wiki says the pin is >>> set initially and only challenged in later attempts [by future >>> callers]: >>> >>> "The first time a conference name (confname) is used, it will be >>> created on demand, and the pin will be set to what ever is specified >>> at that time: the pin in the data string if specified, or if not, >>> the >>> "pin" setting in the conference profile, and if that is also >>> unspecified, then there is no pin protection. Any later attempt to >>> join the conference must specify the same pin number, if one existed >>> when it was created. " >>> >>> >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> >>> I'm sure there's a scenario where its used and useful, the wiki just >>> doesn't explain it. >>> >>> Rob >>> >>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>> >>>> Well first off you're not defining a pine here... >>>> >>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>> pin >>>> number] >>>> >>>> That might be why its not asking for a pin. >>>> >>>> /b >>>> >>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org --??????-- From: rob4manhere at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 5 Nov 2009 08:57:08 -0600 Subject: [Freeswitch-users] Wideband / HD phones Hey all, Looking at buying some high def phones. Any recommendations (preferably based on experience) for hardware based on product quality, standards compliance, features integration with Freeswitch, etc? Thank you! Rob Forman _________________________________________________________________ ?????????????????msn????? http://ditu.live.com/?form=TL&swm=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/0f85d097/attachment-0002.html From maciej.aniserowicz at gmail.com Thu Nov 5 07:35:51 2009 From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz) Date: Thu, 5 Nov 2009 07:35:51 -0800 (PST) Subject: [Freeswitch-users] Users hanged up for unknown reason In-Reply-To: <87f2f3b90911041016u620ca88bk4f0d6a4ceb339b4b@mail.gmail.com> References: <1257244093831-3937601.post@n2.nabble.com> <87f2f3b90911041016u620ca88bk4f0d6a4ceb339b4b@mail.gmail.com> Message-ID: <1257435351018-3952900.post@n2.nabble.com> OK, I put all of the logs in pastebin. Some more background: I use 2 instances of FS. One of them (we call it "prod") is used internally to connect to "the outside world". The other (called "gateway") is "the outside world" for dev and testing purposes. Here is the previous link again, with my commands and "prod" FS events: http://pastebin.freeswitch.org/10955 Here is a log from "prod" FS: part 1: http://pastebin.freeswitch.org/10993, part 2: http://pastebin.freeswitch.org/10994 Here is a log from "gateway" FS: part 1: http://pastebin.freeswitch.org/10995, part 2: http://pastebin.freeswitch.org/10996 Let me know if this is still not enough information, thanks. Maciej Aniserowicz mercutioviz wrote: > > On Tue, Nov 3, 2009 at 2:28 AM, Maciej Aniserowicz < > maciej.aniserowicz at gmail.com> wrote: > >> >> Hi, >> I have a strange problem. I control FS with commands sent by tcp in >> response >> to events published via tcp. I do something like: >> 1) call 1st user >> 2) call 2nd user >> 3) 1st and 2nd talk >> 4) call another user >> 5) 1st and another talk >> etc... >> >> Sometimes (quite regularly) users are hanged up (with cause >> NORMAL_CLEARING) >> even if they do not hangup manually. >> >> I pasted one such scenario in pastebin >> (http://pastebin.freeswitch.org/10955), it includes logs from commands >> sent >> by me and events received from FS. Could someone take a look and see what >> am >> I doing wrong? >> > Seeing only the events it is difficult to see what triggered them. Can you > repeat these tests and capture the debug output from the CLI? > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Users-hanged-up-for-unknown-reason-tp3937601p3952900.html Sent from the freeswitch-users mailing list archive at Nabble.com. From d_hound at ymail.com Thu Nov 5 07:46:28 2009 From: d_hound at ymail.com (Hound Dog) Date: Thu, 5 Nov 2009 07:46:28 -0800 (PST) Subject: [Freeswitch-users] problem with failover routes for LCR A-Z scenario In-Reply-To: References: <727374.16142.qm@web111917.mail.gq1.yahoo.com> Message-ID: <282966.24351.qm@web111917.mail.gq1.yahoo.com> thanks Rupa , its very creative I saw the new feature of user rates on the trunk , but havent worked with it as im using 1.0.4 will have a look and see if it can be used BTW - I already implemented anti-loop on the LCR with custom-sql , adding the incoming carrier-id to a variable on the incoming dial plan and filtering the LCR carriers not to include same carrier , solved some issues for me when carriers send me traffic that they actually supply will document and send it to you soon thank you v much Ori ----- Original Message ---- From: Rupa Schomaker To: freeswitch-users at lists.freeswitch.org Sent: Thu, November 5, 2009 1:31:00 PM Subject: Re: [Freeswitch-users] problem with failover routes for LCR A-Z scenario Actually, using custom sql, you can implement the filter yourself in the where clause. No need for code changes. On Thu, Nov 5, 2009 at 7:17 AM, Rupa Schomaker wrote: > Now that user rates are supported in mod_lcr, how about an option that > says to drop the route if the user_rate is < rate ? > > This 1) requires you to use custom sql and 2) be able to represent > your user rates in that sql (join to user rate table perhaps?) > > On Thu, Nov 5, 2009 at 1:44 AM, Hound Dog wrote: >> I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) >> >> it came out a little long , sorry for that :) >> >> >> I found that it difficult/impossible to make LCR use only carriers that I choose >> >> scenario is as follows , taking the UK as example for a destination ( prices are not real , just an example ) >> >> I have 2 carriers offering routes to the UK , landline and mobile >> >> my buying prices >> >> Destination carrier1 Price carrier2 price >> 44 (all UK) $0.01 $0.01 >> 447 (UK mobile) $0.15 $0.19 >> >> my selling prices >> >> Destination price >> 44 (all UK) $0.015 >> 447 (UKmobile) $0.17 >> >> so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable >> >> for UK mobile I can ** only ** make a profit if I use carrier 1 ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) >> >> >> translating it to MOD_LCR information >> >> digits rate carrier_id ( other columns ignored ) >> 44 0.01 1 >> 44 0.01 2 >> 447 0.015 1 >> >> this looks good : >> 44 prefix will be shared between carrier 1 and 2 >> 447 prefix will only go to carreir 1 >> >> so it fits perfectly - BUT >> >> testing this I get - >> >> API CALL [lcr(447965404547)] output: >> | Digit Match | Carrier | Rate | Codec | CID Regexp | Dialstring | >> | 447 | carr1 | 0.15 | G711 | | [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.1 | >> | 44 | carr2 | 0.01 | G711 | | [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.2 | >> >> Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) - it all makes sense >> >> >> ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail >> >> >> so far I didnt find a solution for that , so if there is one I love bo pointed there >> >> >> >> >> >> >> I did think it over a little and came up with 2 options that could be used , >> and I am also planning to code them and propose patch to maintainers , >> I would love to get comments on those ( in case there are no existing solution ) >> >> >> option 1 - setting some routes as last option , add another param to the LCR table called last_route , >> when hitting a route with last_route=1, stop processing additional routes and return your routing decision so far >> so in our case the route entry with 44 to carrier1 will have last_route=1 , the other 44 routes will have last_route=0 to allow for failover >> >> option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. >> in other words dont failover from a finer route to a wider route. >> it will need to be a global option and I will be quite simple to use, >> but will require entering mutiple entries of the same length prefix for each carrier you would like to use >> its intutive and relatively simple to manage , but requires more lcr entries to get you where you want >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -Rupa > -- -Rupa _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From rupa at rupa.com Thu Nov 5 07:55:41 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 5 Nov 2009 09:55:41 -0600 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> Message-ID: This is true, BUT it is more flexible than it looks. http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E The caller controls can have a key execute a dialplan extension: execute_application You can set a channel var on the moderator prior to joining to the conf. When the extenion is called, you can check the channel var for moderator and act accordingly. Or you can send an event and monitor with an app over ESL and do whatever you want there (probably using the same channel var trick for knowing who is a mod or not). On Thu, Nov 5, 2009 at 8:52 AM, Rob Forman wrote: > Hi UK, > > ?From what I've done and read, the caller-controls (in > conference.conf.xml) can be modified to almost anything you can think > of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a > caller control just for those with the moderator flag. ?So unless you > want everyone able to mute/kick everyone then you can't do it. > > The wiki seems to indicate this as well: > > "Be aware that the caller-controls are applied across the entire > conference. You cannot enter one member of the conference using caller- > controls ABC and then enter a second member using caller-controls XYZ." > > http://wiki.freeswitch.org/wiki/Mod_conference > > > I think this might be a limitation of mod_conference. ?Perhaps one of > the pros can chime in if I'm off-base or there's some nifty way to > accomplish this. > > Cheers, > Rob > > On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > >> Any ideas on the below...has anyone implemented the below: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, November 02, 2009 12:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Rob: >> >> ? Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 9:34 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hm, strange. ?I haven't seen that before. ?Can you pastebin your logs >> at debug level? >> >> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >> >>> It's strange... a tcpdump tells me that there is no DTMF from my >>> provider when using IVR, but when I call into a TN that goes >>> directly into the Conference App, I see DTMF from the provider. >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 7:23 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I've never had any problem with that. ?Is your logging at debug level >>> so you can see the RECV DTFM in the log/fs_cli? ?Are you calling from >>> a SIP phone on the pbx, or via a PSTN provider? ?Maybe your provider >>> isn't passing them through. >>> >>> Make sure your logging is turned up then try something simpler, like >>> calling the echo application, and see if DTFM comes through. >>> >>> Rob >>> >>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>> >>>> Rob: >>>> >>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>> using the below configuration....so it basically timesout. >>>> >>>> UK >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Ujjval Karihaloo >>>> Sent: Monday, October 26, 2009 9:21 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>> correct.. >>>> =============== >>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. ?I guess that's why I didn't understand >>>> the point of the +pin. >>>> ====================== >>>> >>>> I will try it out. >>>> >>>> Again thx a lot for your help. Will keep everyone posted. >>>> >>>> ________________________________________ >>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>> Sent: Friday, October 23, 2009 12:22 PM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> I just re-tested with the pin in my dial plan: >>>> >>>> >>>> >>>> And it doesn't challenge me for the pin. ?I just drop right in. ?I >>>> figured this is how it was intended, since the wiki says the pin is >>>> set initially and only challenged in later attempts [by future >>>> callers]: >>>> >>>> "The first time a conference name (confname) is used, it will be >>>> created on demand, and the pin will be set to what ever is specified >>>> at that time: the pin in the data string if specified, or if not, >>>> the >>>> "pin" setting in the conference profile, and if that is also >>>> unspecified, then there is no pin protection. Any later attempt to >>>> join the conference must specify the same pin number, if one existed >>>> when it was created. " >>>> >>>> >>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. ?I guess that's why I didn't understand >>>> the point of the +pin. >>>> >>>> I'm sure there's a scenario where its used and useful, the wiki just >>>> doesn't explain it. >>>> >>>> Rob >>>> >>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>> >>>>> Well first off you're not defining a pine here... >>>>> >>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>> pin >>>>> number] >>>>> >>>>> That might be why its not asking for a pin. >>>>> >>>>> /b >>>>> >>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>> >>>>>> ? >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From rupa at rupa.com Thu Nov 5 07:59:49 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 5 Nov 2009 09:59:49 -0600 Subject: [Freeswitch-users] problem with failover routes for LCR A-Z scenario In-Reply-To: <282966.24351.qm@web111917.mail.gq1.yahoo.com> References: <727374.16142.qm@web111917.mail.gq1.yahoo.com> <282966.24351.qm@web111917.mail.gq1.yahoo.com> Message-ID: On Thu, Nov 5, 2009 at 9:46 AM, Hound Dog wrote: > thanks Rupa , > > its very creative > > I saw the new feature of user rates on the trunk , but havent worked with it as im using 1.0.4 > will have a look and see if it can be used Cool, if you have issues let me know. We can work through 'em. > BTW - I already implemented anti-loop on the LCR with custom-sql , adding the incoming carrier-id to a variable on the incoming dial plan and filtering the LCR carriers not to include same carrier , solved some issues for me when carriers send me traffic that they actually supply > > will document and send it to you soon custom sql really is a powerful feature. It does require a bit of thinking, but since it can use any channel variable it is very flexible. > > > thank you v much > Ori > > > > > ----- Original Message ---- > From: Rupa Schomaker > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, November 5, 2009 1:31:00 PM > Subject: Re: [Freeswitch-users] problem with failover routes for LCR A-Z scenario > > Actually, using custom sql, you can implement the filter yourself in > the where clause. ?No need for code changes. > > On Thu, Nov 5, 2009 at 7:17 AM, Rupa Schomaker wrote: >> Now that user rates are supported in mod_lcr, how about an option that >> says to drop the route if the user_rate is < rate ? >> >> This 1) requires you to use custom sql and 2) be able to represent >> your user rates in that sql (join to user rate table perhaps?) >> >> On Thu, Nov 5, 2009 at 1:44 AM, Hound Dog wrote: >>> I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) >>> >>> it came out a little long , sorry for that :) >>> >>> >>> I found that it difficult/impossible to make LCR use only carriers that I choose >>> >>> scenario is as follows , taking the UK as example for a destination ?( prices are not real , just an example ) >>> >>> I have 2 carriers offering routes to the UK , landline and mobile >>> >>> my buying prices >>> >>> Destination ? ? ? carrier1 Price ? ?carrier2 price >>> 44 ?(all UK) ? ? ?$0.01 ? ? ? ? ? ? $0.01 >>> 447 (UK mobile) ? $0.15 ? ? ? ? ? ? $0.19 >>> >>> my selling prices >>> >>> Destination ? ? ? ? price >>> 44 ?(all UK) ? ? ? ?$0.015 >>> 447 (UKmobile) ? ? ?$0.17 >>> >>> so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable >>> >>> for UK mobile I can ** only ** make a profit if I use carrier 1 ? ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) >>> >>> >>> translating it to MOD_LCR information >>> >>> digits ? ? rate ? ? ? ?carrier_id ? ? ? ( other columns ignored ) >>> 44 ? ? ? ? 0.01 ? ? ? ?1 >>> 44 ? ? ? ? 0.01 ? ? ? ?2 >>> 447 ? ? ? ?0.015 ? ? ? 1 >>> >>> this looks good : >>> ? ? 44 prefix will be shared between carrier 1 and 2 >>> ? ? 447 prefix will only go to carreir 1 >>> >>> so it fits perfectly - BUT >>> >>> testing this I get - >>> >>> API CALL [lcr(447965404547)] output: >>> ?| Digit Match | Carrier | Rate ? ? | Codec | CID Regexp | Dialstring | >>> ?| 447 ? ? ? ? | carr1 ? | 0.15 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.1 | >>> ?| 44 ? ? ? ? ?| carr2 ? | 0.01 ? ? | G711 ?| ? ? ? ? ? ?| [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547 at 10.10.10.2 | >>> >>> Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) ?- it all makes sense >>> >>> >>> ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail >>> >>> >>> so far I didnt find a solution for that , so if there is one I love bo pointed there >>> >>> >>> >>> >>> >>> >>> I did think it over a little and came up with 2 options that could be used , >>> ?and I am also planning to code them and propose patch to maintainers , >>> ?I would love to get comments on those ( in case there are no existing solution ) >>> >>> >>> option 1 - setting some routes as last option , add another param to the LCR table called ?last_route , >>> ?when hitting a route with last_route=1, ?stop processing additional routes and return your routing decision so far >>> ?so in our case the route entry with 44 to carrier1 will have last_route=1 ?, the other 44 routes will have last_route=0 to allow for failover >>> >>> option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. >>> in other words dont failover from a finer route to a wider route. >>> it will need to be a global option and I will be quite simple to use, >>> ?but will require entering mutiple entries of the same length prefix for each carrier you would like to use >>> ?its intutive and relatively simple to manage , but requires more lcr entries to get you where you want >>> >>> >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> -Rupa >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From pjintheusa at gmail.com Thu Nov 5 08:11:31 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 5 Nov 2009 11:11:31 -0500 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> Message-ID: <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> At $450 on ebay - "Polycom ip6000's AND bust" seems more apt! :) On Thu, Nov 5, 2009 at 10:07 AM, Brian West wrote: > Polycom ip6000's or bust! > > /b > > On Nov 5, 2009, at 8:57 AM, Rob Forman wrote: > > > Hey all, > > > > Looking at buying some high def phones. Any recommendations > > (preferably based on experience) for hardware based on product > > quality, standards compliance, features integration with Freeswitch, > > etc? > > > > Thank you! > > Rob Forman > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/cf930ccb/attachment-0002.html From brian at freeswitch.org Thu Nov 5 08:17:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 10:17:22 -0600 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> Message-ID: Depends... :P If you want quality audio you pay for it... if you want shitty audio you pay for it too just a little less... so your choices are awesome audio or ok audio. /b On Nov 5, 2009, at 10:11 AM, Phillip Jones wrote: > At $450 on ebay - "Polycom ip6000's AND bust" seems more apt! :) From steveu at coppice.org Thu Nov 5 08:43:12 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 06 Nov 2009 00:43:12 +0800 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> Message-ID: <4AF300A0.8000206@coppice.org> Hi Phillip, I'm not sure why Brian suggested the IP6000. Its a big chunky conference room speakerphone (the large triangular type). Very nice, but probably not what you were looking for. If your idea of high def is G.722 there are more conventional phones for half that price. If your idea of high def is something genuinely high definition, like a G.722.1C phone, the choice is rather limited, unless you pay an arm and a leg. Steve On 11/06/2009 12:11 AM, Phillip Jones wrote: > At $450 on ebay - "Polycom ip6000's AND bust" seems more apt! :) > > On Thu, Nov 5, 2009 at 10:07 AM, Brian West > wrote: > > Polycom ip6000's or bust! > > /b > > On Nov 5, 2009, at 8:57 AM, Rob Forman wrote: > > > Hey all, > > > > Looking at buying some high def phones. Any recommendations > > (preferably based on experience) for hardware based on product > > quality, standards compliance, features integration with Freeswitch, > > etc? > > > > Thank you! > > Rob Forman > From mkitchin.public at gmail.com Thu Nov 5 08:46:34 2009 From: mkitchin.public at gmail.com (mkitchin.public at gmail.com) Date: Thu, 05 Nov 2009 10:46:34 -0600 Subject: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones In-Reply-To: <63de75710911041057x472d44aj9bc52bb460a8c8cd@mail.gmail.com> References: <4AF0AC58.3010506@gmail.com> <87f2f3b90911031627o318e771vfb5fdcd2bf936234@mail.gmail.com> <4AF0FEA6.7070308@gmail.com> <025101ca5d10$f81228c0$e8367a40$@com> <20091104064201.GA15804@jdc.jasonjgw.net> <027f01ca5d6c$9c1603a0$d4420ae0$@com> <63de75710911041057x472d44aj9bc52bb460a8c8cd@mail.gmail.com> Message-ID: <4AF3016A.2020100@gmail.com> I hate to say it, but I had to give in and try sipx. the ease of provisioning phones and the ability for helpdesk staff to reset passwords and such through a gui looks like it is too good for me to pass on. mm_202 wrote: > I had the exact same problem with the Cisco phones not being able to > receive calls. > > I fixed it by messing around with the NAT settings in the internal > sofia profile. From what I remember, > I just removed the line > and everything worked fine. > > -- mm_202. > > On Wed, Nov 4, 2009 at 11:33 AM, Peter J. Zandvoort > wrote: > >> Absolutely agreed. To use Matthew's original car metaphor: When you just got >> your learner's permit, the old Chevy may be a better choice than the >> Ferrari. >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason >> White >> Sent: Wednesday, November 04, 2009 1:42 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones >> >> Peter J. Zandvoort wrote: >> >>> After looking at various asterisk distributions, SipX, 3CX and >>> what-have-you, I've come to the conclusion that FreeSWITCH is by far the >>> most advanced platform out there. Its architecture and performance is >>> literally light years ahead of the rest and I have yet to come up with >>> something that it can't do. But all that comes at a price: The learning >>> curve is like scaling a brick wall. >>> >> The most flexible and sophisticated tools tend to have this characteristic, >> the best solution to which is a supportive community and good documentation. >> FreeSWITCH has the community; the documentation is improving thanks to >> ongoing >> efforts to extend, clarify and enhance the wiki. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From larclap at yahoo.com Thu Nov 5 09:27:56 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 5 Nov 2009 09:27:56 -0800 Subject: [Freeswitch-users] FS hangup Message-ID: <00b401ca5e3d$50b7a8b0$f226fa10$@com> I just updated to v15372 from v15311. When calling into FreeSWITCH, it hangs up the call rather than going to voicemail (line 262 in pastebin). I don't know what might be causing this. Can anyone help? Thanks, Lars http://pastebin.freeswitch.org/11006 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/60a65779/attachment-0002.html From dujinfang at gmail.com Thu Nov 5 09:32:25 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Nov 2009 01:32:25 +0800 Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: <23f91030911050932m2695b23eg6a253954b109f16d@mail.gmail.com> 2009/11/5 ??? > Hi All, > > I once meet the Skypiax load error issue and some guys infomed me that > there is no configuration file for Skypiax. > > When I follow these intructions -> > > http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances > > I still have some difficulties unstanding this: > ." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml > to Debug\conf\autoload_configs." > *I did not find such directories in my freeswitch folder. Did not > understand what "src" means, I checked the freeswitch folder and did not > find such a folder named"src". There is a folder named"mod" under freeswitch > while look into "mod" folder there are only some DLL files and can not find > endpoints and etc.* > > src means source code. you can check out from svn trunk or download the source code from files.freeswitch.org. or follow fisheye: http://fisheye.freeswitch.org/browse/FreeSWITCH > 2.You'll probably build the "Debug" version > *I just build this from the precompiled binaries and then launched FS. > I'm not sure what I launched is in debug mode or not. > * > > If anyone can offer some help that really be appriciated. > I never try to do that on windows, but we are running skype trunks on Linux. Also there's a Chinese google group about FreeSWITCH: http://groups.google.com/group/freeswitch-cn And also please don't include long unrelated quoted text in your mail. It wastes time to read and hence some read emails on cellphone. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/bdba7242/attachment-0002.html From gmaruzz at celliax.org Thu Nov 5 09:41:41 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 5 Nov 2009 18:41:41 +0100 Subject: [Freeswitch-users] Skypiax load error In-Reply-To: References: Message-ID: <7b197bef0911050941r1299922amba1f3f1a4b204674@mail.gmail.com> 2009/11/5 ??? : > I once meet the Skypiax load error issue and some guys infomed me that there > is no configuration file for Skypiax. Daniel, maybe this will sound not nice to your hears, but really, you better find a friend that stay there with you and help you and teach you. You will never be able to do things with FS with your present level of skill. FS (and more so mod_skypiax) is not a consumer grade package. It requires a level of skill that you do not have, at the moment. Maybe is sad, but in my opinion is true. I tell you this just to avoid you a lot of frustrations. -gm > > When I follow these intructions -> > http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances > > I still have some difficulties unstanding this: > ." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml to > Debug\conf\autoload_configs." > I did not find such directories in my freeswitch folder. Did not understand > what "src" means, I checked the freeswitch folder and did not find such a > folder named"src". There is a folder named"mod" under freeswitch while look > into "mod" folder there are only some DLL files and can not find endpoints > and etc. > > 2.You'll probably build the "Debug" version > I just build this from the precompiled binaries and then launched FS. I'm > not sure what I launched is in debug mode or not. > > If anyone can offer some help that really be appriciated. > > Thanks > Daniel Zeng > From: freeswitch-users-request at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 41, Issue 40 > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 5 Nov 2009 06:57:44 -0800 > > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > --??????-- > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date : Thu, 5 Nov 2009 08:13:46 -0600 > Subject: Re: [Freeswitch-users] no REINVITE on Blind Transfer with > bypass_media > > I did not ask you to send me a ladder diagram. > I asked you to send me a console trace from FreeSWITCH using latest trunk > (1.0.4 does not help me) > > 1) start FreeSWITCH > 2) run the cli command: console loglevel debug > > 3) run the cli command: sofia profile internal siptrace on > 4) reproduce your issue and put the trace on freeswitch pastebin > http://pastebin.freeswitch.org (login and pass are stated in the auth > dialog) > > > > Also please answer brian's question. What phones and/or sip devices are > involved in this call. > > > > On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana > wrote: > > > > Thanks for your time, > > > > -The scenario is still the same: > > > > Always bypass media. > > Environment 100% NAT free :-) > > Call established from A to B through FS. Then... > > Blind transfer from B to C (Refer-to: C) > > RTP should go directly between A and C. > > > > > > -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the > REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the > lack of reINVITE to A, after C answers). > > > > Please check SIP diagram here: > > > > http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html > > > > > > -What it's wrong with r15332 is there is not such call to C. For sure I know > SIP is a protocol, may be my description was not clear but this SIP diagram > speaks by itself ;-) > > > > http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html > > > > > > -You could check the sofia debug for r15332 here: > > http://pastebin.com/m6f2b3836 > > > > > > Best regards, > > > > Humberto > > > >> > >> I don't know what you are talking about anymore. > >> > >> The scenario I had tested is when a call is bridged in bypass_media=true > >> bridge > >> and you blind transfer that call back to the dialplan > >> > >> as soon as it hits the routing state it will resume media. > >> > >> > >> it has been confirmed to not work and confirmed to have been fixed several > >> time and if you are still having a problem you must have something >> blocking > >> some of your packets or something . > >> > >> You have to understand that sip is a protocol and your description is > >> completely non-standard. > >> Perhaps you should get a console trace and attach it to a jira. The trace > >> probably makes more sense to me. > >> > >> sofia profile internal siptrace on > >> console loglevel debug > >> > >> reprodu ce and attach the whole capture. > >> > >> > >> > >> On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > >> > >>> > >>> Hi, > >>> > >>> I tried r15332 and set in the sofia profile: > >>> > >>> a) bypass_media_after_bridge=true only > >>> b) bypass_media_after_bridge=true, param name="media-option" > >>> value="resume-media-on-hold"/> > >>> > >>> > >>> In both cases FS is hanging up the initial call (A to FS) after accepting > >>> the REFER to C: > >>> > >>> A <- reINVITE with FS' SDP <- FS > >>> A -> 200 -> FS > >>> A <- ACK <- FS > >>> A <- BYE <- FS > >>> > >>> The call to C is not even tried. > >>> > >>> I found this line is the logs that could give some idea: > >>> > >>> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup > >>> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] >>> [RECOVERY_ON_TIMER_EXPIRE] > >>> after sending the ACK for the reINVITE > >>> > >>> > >>> Regards, > >>> > >>> > >>> Humberto > >>> > >>>>please try r15326 > >>>>I think i have it working. > >>>> > >>>>I recommend for optimal results you set bypass_media_after_bridge=true > >>>>either as a global or in your DP in place of bypass_media=true > >>>> > >>>> > >>>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana > >>> hotmail.com>wrote: > >>>> > >>>>> Hi Mike, > >>>>> > >>>>> I re-tried with trunk rev 15319 but I got almost the same behavior: > >>> There > >>>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. > >>> But > >>>>> still there is no reINVITE for A (with C's SDP) after the call from FS > >>> to C > >>>>> is established. > >>>>> > >>>>> Anyway, we decided for now to do a different implementation but if you > >>> want > >>>>> to explore more in this issue count me in ;-) > >>>>> > >>>>> > >>>>> Thank you very much! > >>>>> > >>>>> Humberto > >>> > >>> > >>> _____________________________________ ____________________________ > >>> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they > >>> e-mail you. > >>> http://go.microsoft.com/?linkid=9691817 > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> _________________________________________________________________ > >> Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows >> 7 now > >> http://go.microsoft.com/?linkid=9691818 > > > > _________________________________________________________________ > > Windows Live: Make it easier for your friends to see what you're up to on > Facebook. > > http://go.microsoft.com/?linkid=9691816 > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > --??????-- > From: rob4manhere at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 5 Nov 2009 08:52:05 -0600 > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hi UK, > > From what I've done and read, the caller-controls (in > conference.conf.xml) can be modified to almost anything you can think > of, BUT, > they are mapped 1-to-1 to a conference- ie you can't map a > caller control just for those with the moderator flag. So unless you > want everyone able to mute/kick everyone then you can't do it. > > The wiki seems to indicate this as well: > > "Be aware that the caller-controls are applied across the entire > conference. You cannot enter one member of the conference using caller- > controls ABC and then enter a second member using caller-controls XYZ." > > http://wiki.freeswitch.org/wiki/Mod_conference > > > I think this might be a limitation of mod_conference. Perhaps one of > the pros can chime in if I'm off-base or there's some nifty way to > accomplish this. > > Cheers, > Rob > > On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > >> Any ideas on the below...has anyone implemented the below: >> >> Once I have the Moderator and Participants logg > ed on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, November 02, 2009 12:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Rob: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman> Sent: Friday, October 30, 2009 9:34 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hm, strange. I haven't seen that before. Can you pastebin your logs >> at debug level? >> >> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >> >>> It's strange... a tcpdump tells me that there is no DTMF from my >>> provider when using IVR, but when I call into a TN that goes >>> directly into the Conference App, I see DTMF from the provider. >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 7:23 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference w > ith Moderator >>> >>> I've never had any problem with that. Is your logging at debug level >>> so you can see the RECV DTFM in the log/fs_cli? Are you calling from >>> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >>> isn't passing them through. >>> >>> Make sure your logging is turned up then try something simpler, like >>> calling the echo application, and see if DTFM comes through. >>> >>> Rob >>> >>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>> >>>> Rob: >>>> >>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>> using the below configuration....so it basically timesout. >>>> >>>> UK >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Ujjval Karihaloo >>>> Sent: Monday, October 26, 2009 9:21 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>> correct.. >>>> =============== >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> ====================== >>>> >>>> I will try it out. >>>> >>>> Again thx a lot for your help. Will keep everyone posted. >>>> >>>> _______________________ > _________________ >>>> From: freeswitch-users-bounces at lists.freeswitch.org >>>> [freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>> Sent: Friday, October 23, 2009 12:22 PM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> I just re-tested with the pin in my dial plan: >>>> >>>> >>>> >>>> And it doesn't challenge me for the pin. I just drop right in. I >>>> figured this is how it was intended, since the wiki says the pin is >>>> set initially and only challenged in later attempts [by future >>>> callers]: >>>> >>>> "The first time a conference name (confname) is used, i > t will be >>>> created on demand, and the pin will be set to what ever is specified >>>> at that time: the pin in the data string if specified, or if not, >>>> the >>>> "pin" setting in the conference profile, and if that is also >>>> unspecified, then there is no pin protection. Any later attempt to >>>> join the conference must specify the same pin number, if one existed >>>> when it was created. " >>>> >>>> >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> >>>> I'm sure there's a scenario where its used and useful, the wiki just >>>> doesn't explain it. >>>> >>> > > Rob >>>> >>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>> >>>>> Well first off you're not defining a pine here... >>>>> >>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>> pin >>>>> number] >>>>> >>>>> That might be why its not asking for a pin. >>>>> >>>>> /b >>>>> >>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> __________ > _____________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.free > switch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/ > freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > > --??????-- > From: rob4manhere at gmail.com > To: freeswitch-users at lists.freeswitch.org > Date: T hu, 5 Nov 2009 08:57:08 -0600 > Subject: [Freeswitch-users] Wideband / HD phones > > Hey all, > > Looking at buying some high def phones. Any recommendations > (preferably based on experience) for hardware based on product > quality, standards compliance, features integration with Freeswitch, > etc? > > Thank you! > Rob Forman > > > > ________________________________ > ??????????MSN??? ????? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From shiyanov at gmail.com Thu Nov 5 09:44:28 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 5 Nov 2009 20:44:28 +0300 Subject: [Freeswitch-users] Dialpan: try.. finally analogs Message-ID: Hello! I have to deal with classic problem: "Leaking stream handle" in FS console. I also know the reason - firstly channel is sent to the extension with "playback" and later it is transfered to another extensions with "execute_extension" or, another trouble-case - channel is bridged to some addres. I do not ask (but I wish to) why FS doesn't close stream automatically when channel is gone. I ask whether it is possible to use some "try.. finally" construction in diaplan? If "yes" then I can simply stop playback in the "finally" block.. Any thoughs? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/2e5e5682/attachment-0002.html From frank at carmickle.com Thu Nov 5 09:45:30 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 5 Nov 2009 12:45:30 -0500 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <4AF300A0.8000206@coppice.org> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> <4AF300A0.8000206@coppice.org> Message-ID: <20091105174529.GO10757@base.carmickle.com> On Fri, Nov 06, Steve Underwood wrote: > If your idea of high def is G.722 there are more conventional phones for > half that price. And there is portaudio in freeswitch itself. With a usb headset and celt at 48k how can you go wrong? --FC From anthony.minessale at gmail.com Thu Nov 5 09:47:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Nov 2009 11:47:47 -0600 Subject: [Freeswitch-users] FS hangup In-Reply-To: <00b401ca5e3d$50b7a8b0$f226fa10$@com> References: <00b401ca5e3d$50b7a8b0$f226fa10$@com> Message-ID: <191c3a030911050947o6bbc3e97j24898aa50951f54@mail.gmail.com> do you have continue_on_fail set? if you do you have to include no_answer,busy etc once you set it, you have to set *everything* you want. On Thu, Nov 5, 2009 at 11:27 AM, Lars Zeb wrote: > I just updated to v15372 from v15311. When calling into FreeSWITCH, it > hangs up the call rather than going to voicemail (line 262 in pastebin). I > don?t know what might be causing this. > > > > Can anyone help? > > > > Thanks, Lars > > > > http://pastebin.freeswitch.org/11006 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/6069e3b1/attachment-0002.html From shiyanov at gmail.com Thu Nov 5 09:49:49 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 5 Nov 2009 20:49:49 +0300 Subject: [Freeswitch-users] Filtering a particular event. In-Reply-To: References: Message-ID: it's possible. Mod_socket commands: event myevents event plain On Thu, Nov 5, 2009 at 4:28 PM, Nagalenoj H. wrote: > Hi, > I've tried to filter the events like below to filter a particular event. > > 1) register for all events > 2) filter for one unique-id > 3) filter only one/more events(ex: DTMF & CHANNEL_EXECUTE) > > So, I want to receive only these events for the specific unique-id. But, I > am receiving other events too. I'm using perl ESL outbound. > Is it possible to do like this?! > > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/09492375/attachment-0002.html From shiyanov at gmail.com Thu Nov 5 09:52:15 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 5 Nov 2009 20:52:15 +0300 Subject: [Freeswitch-users] Java example In-Reply-To: <85845D7B-9D9D-4BFA-ACCA-0F28DA4EBA9E@freeswitch.org> References: <44498.1257162831@entvoice.com> <85845D7B-9D9D-4BFA-ACCA-0F28DA4EBA9E@freeswitch.org> Message-ID: let's say: starpound is using FS for business process automation + telephony On Mon, Nov 2, 2009 at 10:07 PM, Brian West wrote: > Is starpound involved in the FS Community? > > /b > > > On Nov 2, 2009, at 12:51 PM, Artem Shiyanov wrote: > > Here is rather big and, let's say, complete example of mod_java usage: > https://starpound.svn.sourceforge.net/svnroot/starpound/trunk/src/fs2agi > The goal of this project is to be a proxy between FreeSwitch and server > application which knows Asterisk AGI. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/44ada5a5/attachment-0002.html From brian at freeswitch.org Thu Nov 5 09:55:31 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 11:55:31 -0600 Subject: [Freeswitch-users] Java example In-Reply-To: References: <44498.1257162831@entvoice.com> <85845D7B-9D9D-4BFA-ACCA-0F28DA4EBA9E@freeswitch.org> Message-ID: But they should be more involved in the Community if possible. /b On Nov 5, 2009, at 11:52 AM, Artem Shiyanov wrote: > let's say: starpound is using FS for business process automation + > telephony > > > > On Mon, Nov 2, 2009 at 10:07 PM, Brian West > wrote: > Is starpound involved in the FS Community? > > /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/28d14896/attachment-0002.html From dujinfang at gmail.com Thu Nov 5 09:57:59 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Nov 2009 01:57:59 +0800 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> Message-ID: <23f91030911050957m796fe88fj5da881875c010e6b@mail.gmail.com> Ciao Giovanni, Do you still plan to merge this? 2009/9/5 Giovanni Maruzzelli > Seven, > > thanks a lot for your efforts. > > I will merge it in the next days, and I will take care that it will > not breaks Windows or Linux. > > If I find problems I will wait for you having more time in the future. > > I send you my super best wishes for your personal things to go well > and solves in the best of the possible ways. > > ciao for now, > > -giovanni > > > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > > > On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: > > gm, > > > > Thanks a lot you can merge into the mainline. I check into my branch > > because it's currently not as useful as on Linux and Windows and the > > solution is not good. But it works and it is a good start that > > mod_skypiax runs on OSX. Sure it would be easier for people want to > > test and improve it if it been merged into trunk. I think you can make > > a diff by > > > > svn diff -r 14472:14772 > http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax > > > > FYI for personal reason I won't have much time put on this in the > > coming month. Actually the code was done a few weeks ago but i only > > got a chance to commit it yesterday. Sure that is not to say I cannot > > do but fixes. But can you please make sure it won't break Linux/ > > windows build when you merge the code? I haven't have a chance to test > > all of them yet. > > > > -7- > > > > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: > >> Seeeeeeeven! > >> > >> I saw the modification you made on the wiki page... > >> > >> You made it, mod_skypiax runs on OSX!!!! > >> > >> Let's merge your mods on the mainline, pleaaaase ;-))) > >> > >> -giovanni > >> > >> > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/5c3ba1c2/attachment-0002.html From gmaruzz at celliax.org Thu Nov 5 10:03:43 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 5 Nov 2009 19:03:43 +0100 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <23f91030911050957m796fe88fj5da881875c010e6b@mail.gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> <23f91030911050957m796fe88fj5da881875c010e6b@mail.gmail.com> Message-ID: <7b197bef0911051003t13e363edqff7b76ecfc099ed5@mail.gmail.com> On Thu, Nov 5, 2009 at 6:57 PM, Seven Du wrote: > Ciao Giovanni, > > Do you still plan to merge this? Sorry Seven, I've lost track of this, and now I'm so sick I'm completely un-useful ;). But yes, I would like to do it, if you think it is in a useful state. Can you please create a Jira and attach an svn diff, so in the next days I can merge it? -giovanni > > 2009/9/5 Giovanni Maruzzelli >> >> Seven, >> >> thanks a lot for your efforts. >> >> I will merge it in the next days, and I will take care that it will >> not breaks Windows or Linux. >> >> If I find problems I will wait for you having more time in the future. >> >> I send you my super best wishes for your personal things to go well >> and solves in the best of the possible ways. >> >> ciao for now, >> >> -giovanni >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> >> >> >> On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: >> > gm, >> > >> > Thanks a lot you can merge into the mainline. I check into my branch >> > because it's currently not as useful as on Linux and Windows and the >> > solution is not good. But it works and it is a good start that >> > mod_skypiax runs on OSX. Sure it would be easier for people want to >> > test and improve it if it been merged into trunk. I think you can make >> > a diff by >> > >> > svn diff -r 14472:14772 >> > http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax >> > >> > FYI for personal reason I won't have much time put on this in the >> > coming month. Actually the code was done a few weeks ago but i only >> > got a chance to commit it yesterday. Sure that is not to say I cannot >> > do but fixes. But can you please make sure it won't break Linux/ >> > windows build when you merge the code? I haven't have a chance to test >> > all of them yet. >> > >> > -7- >> > >> > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: >> >> Seeeeeeeven! >> >> >> >> I saw the modification you made on the wiki page... >> >> >> >> You made it, mod_skypiax runs on OSX!!!! >> >> >> >> Let's merge your mods on the mainline, pleaaaase ;-))) >> >> >> >> -giovanni >> >> >> >> >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> Cell : +39-347-2665618 >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From larclap at yahoo.com Thu Nov 5 10:17:03 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 5 Nov 2009 10:17:03 -0800 Subject: [Freeswitch-users] FS hangup In-Reply-To: <191c3a030911050947o6bbc3e97j24898aa50951f54@mail.gmail.com> References: <00b401ca5e3d$50b7a8b0$f226fa10$@com> <191c3a030911050947o6bbc3e97j24898aa50951f54@mail.gmail.com> Message-ID: <00d201ca5e44$2d59b000$880d1000$@com> Thanks for the help. Yes, I am using a lua script to handle inbound calls with continue_on_fail set to true: session:execute("set", "continue_on_fail=true"); I changed it to: session:execute("set", "continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DEST INATION"); and it works OK now. Did something change between v15311 to v15372 to make this behave differently? I ask because it worked with "true" in the earlier version. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, November 05, 2009 9:48 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS hangup do you have continue_on_fail set? if you do you have to include no_answer,busy etc once you set it, you have to set *everything* you want. On Thu, Nov 5, 2009 at 11:27 AM, Lars Zeb wrote: I just updated to v15372 from v15311. When calling into FreeSWITCH, it hangs up the call rather than going to voicemail (line 262 in pastebin). I don't know what might be causing this. Can anyone help? Thanks, Lars http://pastebin.freeswitch.org/11006 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/ba3ea9f5/attachment-0002.html From anthony.minessale at gmail.com Thu Nov 5 10:35:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Nov 2009 12:35:47 -0600 Subject: [Freeswitch-users] FS hangup In-Reply-To: <00d201ca5e44$2d59b000$880d1000$@com> References: <00b401ca5e3d$50b7a8b0$f226fa10$@com> <191c3a030911050947o6bbc3e97j24898aa50951f54@mail.gmail.com> <00d201ca5e44$2d59b000$880d1000$@com> Message-ID: <191c3a030911051035y739919c7w2663025e93d64976@mail.gmail.com> yes sounds like a bug. I think i redid it and forgot to check for "true" still =0 On Thu, Nov 5, 2009 at 12:17 PM, Lars Zeb wrote: > Thanks for the help. Yes, I am using a lua script to handle inbound calls > with continue_on_fail set to true: > > > > session:execute("set", "continue_on_fail=true"); > > > > I changed it to: > > > > session:execute("set", > "continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"); > > > > and it works OK now. > > > > Did something change between v15311 to v15372 to make this behave > differently? I ask because it worked with ?true? in the earlier version. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Thursday, November 05, 2009 9:48 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FS hangup > > > > do you have continue_on_fail set? > if you do you have to include no_answer,busy etc once you set it, you have > to set *everything* you want. > > > On Thu, Nov 5, 2009 at 11:27 AM, Lars Zeb wrote: > > I just updated to v15372 from v15311. When calling into FreeSWITCH, it > hangs up the call rather than going to voicemail (line 262 in pastebin). I > don?t know what might be causing this. > > > > Can anyone help? > > > > Thanks, Lars > > > > http://pastebin.freeswitch.org/11006 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/6e60e188/attachment-0002.html From anthony.minessale at gmail.com Thu Nov 5 10:40:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Nov 2009 12:40:03 -0600 Subject: [Freeswitch-users] FS hangup In-Reply-To: <191c3a030911051035y739919c7w2663025e93d64976@mail.gmail.com> References: <00b401ca5e3d$50b7a8b0$f226fa10$@com> <191c3a030911050947o6bbc3e97j24898aa50951f54@mail.gmail.com> <00d201ca5e44$2d59b000$880d1000$@com> <191c3a030911051035y739919c7w2663025e93d64976@mail.gmail.com> Message-ID: <191c3a030911051040p31e04e4cj4f27c2821edce64@mail.gmail.com> fixed in 15376 On Thu, Nov 5, 2009 at 12:35 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > yes sounds like a bug. > I think i redid it and forgot to check for "true" still =0 > > > > On Thu, Nov 5, 2009 at 12:17 PM, Lars Zeb wrote: > >> Thanks for the help. Yes, I am using a lua script to handle inbound >> calls with continue_on_fail set to true: >> >> >> >> session:execute("set", "continue_on_fail=true"); >> >> >> >> I changed it to: >> >> >> >> session:execute("set", >> "continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"); >> >> >> >> and it works OK now. >> >> >> >> Did something change between v15311 to v15372 to make this behave >> differently? I ask because it worked with ?true? in the earlier version. >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony >> Minessale >> *Sent:* Thursday, November 05, 2009 9:48 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] FS hangup >> >> >> >> do you have continue_on_fail set? >> if you do you have to include no_answer,busy etc once you set it, you have >> to set *everything* you want. >> >> >> On Thu, Nov 5, 2009 at 11:27 AM, Lars Zeb wrote: >> >> I just updated to v15372 from v15311. When calling into FreeSWITCH, it >> hangs up the call rather than going to voicemail (line 262 in pastebin). I >> don?t know what might be causing this. >> >> >> >> Can anyone help? >> >> >> >> Thanks, Lars >> >> >> >> http://pastebin.freeswitch.org/11006 >> >> >> >> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 >> i386 GNU/Linux >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/2960fde8/attachment-0002.html From hjqlopez at hotmail.com Thu Nov 5 10:39:59 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Thu, 5 Nov 2009 13:39:59 -0500 Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media In-Reply-To: <403475.98637.qm@web51107.mail.re2.yahoo.com> References: <403475.98637.qm@web51107.mail.re2.yahoo.com> Message-ID: >> -You could check the sofia debug for r15332 here: http://pastebin.freeswitch.org/11008 Phone/Devices: The caller is the DID provider's Switch. The callee (which also sends the REFER) is Asterisk 1.4.26. Testing with other devices(linksys SPA962,Grandstream GXV3000) gathers the same results. > I did not ask you to send me a ladder diagram. > I asked you to send me a console trace from FreeSWITCH using latest trunk > (1.0.4 does not help me) > > 1) start FreeSWITCH > 2) run the cli command: console loglevel debug > 3) run the cli command: sofia profile internal siptrace on > 4) reproduce your issue and put the trace on freeswitch pastebin > http://pastebin.freeswitch.org (login and pass are stated in the auth > dialog) > > Also please answer brian's question. What phones and/or sip devices are > involved in this call. > > > > On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana wrote: > >> >> Thanks for your time, >> >> -The scenario is still the same: >> >> Always bypass media. >> Environment 100% NAT free :-) >> Call established from A to B through FS. Then... >> Blind transfer from B to C (Refer-to: C) >> RTP should go directly between A and C. >> >> >> -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the >> REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the >> lack of reINVITE to A, after C answers). >> >> Please check SIP diagram here: >> >> http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html >> >> >> -What it's wrong with r15332 is there is not such call to C. For sure I >> know SIP is a protocol, may be my description was not clear but this SIP >> diagram speaks by itself ;-) >> >> http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html >> >> >> -You could check the sofia debug for r15332 here: >> http://pastebin.com/m6f2b3836 >> >> >> Best regards, >> >> Humberto >> >>> >>> I don't know what you are talking about anymore. >>> >>> The scenario I had tested is when a call is bridged in bypass_media=true >>> bridge >>> and you blind transfer that call back to the dialplan >>> >>> as soon as it hits the routing state it will resume media. >>> >>> >>> it has been confirmed to not work and confirmed to have been fixed >> several >>> time and if you are still having a problem you must have something >> blocking >>> some of your packets or something . >>> >>> You have to understand that sip is a protocol and your description is >>> completely non-standard. >>> Perhaps you should get a console trace and attach it to a jira. The trace >>> probably makes more sense to me. >>> >>> sofia profile internal siptrace on >>> console loglevel debug >>> >>> reproduce and attach the whole capture. >>> >>> >>> >>> On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: >>> >>>> >>>> Hi, >>>> >>>> I tried r15332 and set in the sofia profile: >>>> >>>> a) bypass_media_after_bridge=true only >>>> b) bypass_media_after_bridge=true, param name="media-option" >>>> value="resume-media-on-hold"/> >>>> >>>> >>>> In both cases FS is hanging up the initial call (A to FS) after >> accepting >>>> the REFER to C: >>>> >>>> A <- reINVITE with FS' SDP <- FS >>>> A -> 200 -> FS >>>> A <- ACK <- FS >>>> A <- BYE <- FS >>>> >>>> The call to C is not even tried. >>>> >>>> I found this line is the logs that could give some idea: >>>> >>>> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >>>> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] >> [RECOVERY_ON_TIMER_EXPIRE] >>>> after sending the ACK for the reINVITE >>>> >>>> >>>> Regards, >>>> >>>> >>>> Humberto >>>> >>>>>please try r15326 >>>>>I think i have it working. >>>>> >>>>>I recommend for optimal results you set bypass_media_after_bridge=true >>>>>either as a global or in your DP in place of bypass_media=true >>>>> >>>>> >>>>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >>>> hotmail.com>wrote: >>>>> >>>>>> Hi Mike, >>>>>> >>>>>> I re-tried with trunk rev 15319 but I got almost the same behavior: >>>> There >>>>>> is now a reINVITE (with FS' SDP) going to A when the REFER is >> accepted. >>>> But >>>>>> still there is no reINVITE for A (with C's SDP) after the call from FS >>>> to C >>>>>> is established. >>>>>> >>>>>> Anyway, we decided for now to do a different implementation but if you >>>> want >>>>>> to explore more in this issue count me in ;-) >>>>>> >>>>>> >>>>>> Thank you very much! >>>>>> >>>>>> Humberto > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com _________________________________________________________________ Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now http://go.microsoft.com/?linkid=9691818 From codecomplete at free.fr Thu Nov 5 10:43:54 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 5 Nov 2009 10:43:54 -0800 (PST) Subject: [Freeswitch-users] Does OpenZap support CTR21? Message-ID: <26217371.post@talk.nabble.com> Hello As an alternative to more expensive alternatives like OpenVox or Sangoma, I'd like to order an X100P clone from www.x100p.com for use in France. According to a PDF on the site, the reason this card gets bad reviews is that "the Silicon labs Si3012/Si3035 DAA chip used in the original Digium X100P card and low cost X100P clone cards only supports FCC mode. However, the Si3014/Si3034 DAA chip used on the X100P SE supports global line standards." As for software, "the Silicon labs Si3014/Si3034 DAA chip used in the X100P SE supports 600 Ohm impedance and complex impedance to meet CTR21 line standards. However, the Zaptel wcfxo driver only supports CTR21 mode with 600 Ohm AC termination, which may or may not be the correct setting depending on the country and the phone system in use." So... does someone know if OpenZap, which is apparently required in addition to Zaptel/Dahdi for FreeSwitch to work PCI TDM cards, supports CTR21? Thank you. -- View this message in context: http://old.nabble.com/Does-OpenZap-support-CTR21--tp26217371p26217371.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu Nov 5 10:47:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Nov 2009 10:47:38 -0800 Subject: [Freeswitch-users] Calling more than 1 variable in condition In-Reply-To: References: Message-ID: <87f2f3b90911051047u34559431idd898ee9e687a011@mail.gmail.com> On Wed, Nov 4, 2009 at 10:22 PM, Ahmed Munir wrote: > Hi, > > In my dial plan I've created a variable named SIP_CALL, PSTN_CALL. If > SIP_CALL = true, it dials out to sip call, when PSTN_CALL=true, it dials out > to landline call, as I provide sample below; > > > > > > > > > > > > The problem I'm facing is how can I apply condition when I've to call more > than 1 variables? Like if there are no records in SIP numbering plan table > and PSTN numbering plan table so it get the digits and dial out the to > carrier (how to apply AND operation in condition?) i.e. > > > > > > > > AND operations are very simple - just stack the conditions: <-- actions here --> note that you must close the first condition's tag! BTW, this is covered in the dialplan section of the wiki. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/052f3f50/attachment-0002.html From michal.bielicki at halo2.pl Thu Nov 5 11:05:10 2009 From: michal.bielicki at halo2.pl (Michal Bielicki) Date: Thu, 5 Nov 2009 20:05:10 +0100 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <20091105174529.GO10757@base.carmickle.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <5ACA7190-A042-4DA4-96DA-805825FA26B2@freeswitch.org> <367751820911050811r6947476clee389c5aae6e6209@mail.gmail.com> <4AF300A0.8000206@coppice.org> <20091105174529.GO10757@base.carmickle.com> Message-ID: <4CBF4621-B12A-46F3-9C1B-90AB1D158C0A@halo2.pl> I'd rather go with the ip7000 since it has better audio gear in it. For a deskphone everything Polycom >= ip450 is absolutely wideband enough for a deskphone. Personally I am currently a total fan of the VVX which is a video deskphone with the same audio as a IP6000 Am 05.11.2009 um 18:45 schrieb Frank Carmickle: > On Fri, Nov 06, Steve Underwood wrote: >> If your idea of high def is G.722 there are more conventional >> phones for >> half that price. > > And there is portaudio in freeswitch itself. With a usb headset and > celt at 48k how can you go wrong? > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org Michal Bielicki HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa t. +48228753290 | f. +48228753291 michal.bielicki at halokwadrat.pl | w. www.halokwadrat.pl Knowledge & Low Prices. Guaranteed! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/cb9586f0/attachment-0002.html From info at daccii.it Thu Nov 5 12:35:12 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Thu, 05 Nov 2009 21:35:12 +0100 Subject: [Freeswitch-users] Transfer call to group Message-ID: <4AF33700.1040209@daccii.it> Hi, actually i'm trying to setup an IVR that, when the choice is done, transfer the call to a group, really simply. Here the dialplan in default context to handle call to group (four extensions, one for group, from 2001 to 2004) http://pastebin.freeswitch.org/11014 Here the output log http://pastebin.freeswitch.org/11015 When i call the group directly from a telephone in the default context or when the ivr transfer me to the group i didn't get nothing, looking to log you can see (line 153) EXECUTE sofia/internal/15 at 192.168.0.77 bridge() Data for bridge application is ${group_call(${dialed_extension}+F@${domain_name})} It's, probably, a stupid error, but the only other way to accomplish this is to bridge individually phones using | as separator but i would to mantain a single extension to handle this stuff. Thanks for your support! Best Regards, Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/ebbe66a4/attachment-0002.vcf From mike at jerris.com Thu Nov 5 13:19:42 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Nov 2009 16:19:42 -0500 Subject: [Freeswitch-users] Dialpan: try.. finally analogs In-Reply-To: References: Message-ID: It cleans up after itself fine, but it is an indication of some issue in the code we need to address. if you can reproduce this in svn trunk, please file a bug on jira.freeswitch.org with details how to reproduce. mike On Nov 5, 2009, at 12:44 PM, Artem Shiyanov wrote: > Hello! > > I have to deal with classic problem: "Leaking stream handle" in FS > console. I also know the reason - firstly channel is sent to the > extension with "playback" and later it is transfered to another > extensions with "execute_extension" or, another trouble-case - > channel is bridged to some addres. > I do not ask (but I wish to) why FS doesn't close stream > automatically when channel is gone. > I ask whether it is possible to use some "try.. finally" > construction in diaplan? If "yes" then I can simply stop playback in > the "finally" block.. > > Any thoughs? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Thu Nov 5 13:20:58 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Nov 2009 16:20:58 -0500 Subject: [Freeswitch-users] Does OpenZap support CTR21? In-Reply-To: <26217371.post@talk.nabble.com> References: <26217371.post@talk.nabble.com> Message-ID: <7FD19B47-C121-48CD-98C2-2830BFDF1068@jerris.com> This would be specific to the zaptel driver for that card, not openzap. mike On Nov 5, 2009, at 1:43 PM, Fred-145 wrote: > > Hello > > As an alternative to more expensive alternatives like OpenVox or > Sangoma, > I'd like to order an X100P clone from www.x100p.com for use in France. > > According to a PDF on the site, the reason this card gets bad > reviews is > that "the Silicon labs Si3012/Si3035 DAA chip used in the original > Digium > X100P card and low cost X100P clone cards only supports FCC mode. > However, > the Si3014/Si3034 DAA chip used on the X100P SE supports global line > standards." > > As for software, "the Silicon labs Si3014/Si3034 DAA chip used in > the X100P > SE supports 600 Ohm impedance and complex impedance to meet CTR21 line > standards. However, the Zaptel wcfxo driver only supports CTR21 mode > with > 600 Ohm AC termination, which may or may not be the correct setting > depending on the country and the phone system in use." > > So... does someone know if OpenZap, which is apparently required in > addition > to Zaptel/Dahdi for FreeSwitch to work PCI TDM cards, supports CTR21? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/99ef82d2/attachment-0002.html From frank at carmickle.com Thu Nov 5 14:20:18 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 5 Nov 2009 17:20:18 -0500 Subject: [Freeswitch-users] [ERR] mod_portaudio.c:974 Cannot find an input device Message-ID: <20091105222017.GP10757@base.carmickle.com> Hello I updated to 15376 added some build depends and still no joy. Any more pointers. Thanks. --FC From Russell.Mosemann at cune.org Thu Nov 5 14:20:31 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Thu, 5 Nov 2009 22:20:31 -0000 Subject: [Freeswitch-users] DAHDI issue Message-ID: <20091105222031.562F2424FBD@mail.cune.org> Debian 5.0.3 FreeSWITCH Version 1.0.trunk (15376M) openzap and libpri-1.4.10.2 dahdi-linux-complete-2.2.0.2+2.2.0 Digium Wildcard TE110P T1/E1 Card (running as a T1) This was working with zaptel. I thought that I would upgrade from zaptel to DAHDI, but it's generating "no such device or address" errors. FS is running as root but can't seem to see the channels. I have unloaded and loaded the drivers. Permissions look fine. The dahdi tools can see the card. Any insights? http://pastebin.freeswitch.org/11016 -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From Russell.Mosemann at cune.org Thu Nov 5 14:42:45 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Thu, 5 Nov 2009 22:42:45 -0000 Subject: [Freeswitch-users] DAHDI issue Message-ID: <20091105224245.4CC6F421B07@mail.cune.org> > I thought that I would upgrade from zaptel to DAHDI, After I send the message, the answer comes to me. I guess that's the way things work. :-) I had forgotten to define the channels in /etc/dahdi/system.conf. Here are the settings, and things are working. Thanks for listening. :-) span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us echocanceller=mg2,1-23 -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From JCasale at activenetwerx.com Thu Nov 5 14:54:27 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 5 Nov 2009 22:54:27 +0000 Subject: [Freeswitch-users] sip profile question Message-ID: The internal.xml also has an "ext-rtp-ip" variable and in trying to understand what this is for (my version of fs is <1) I noticed in trunks conf file it is explained. So the available options that I have given my setup is multihomed with a lan/wan setup where the wan interface is dynamic would be a fqdn for fs to lookup, or auto/auto-nat. How exactly does auto and auto-nat work so I may know of its going to work properly/reliably in my scenario. Thanks! jlc From brian at freeswitch.org Thu Nov 5 15:00:10 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:00:10 -0600 Subject: [Freeswitch-users] sip profile question In-Reply-To: References: Message-ID: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> auto-nat tries to use upnp/nat-pmp to figure it out... auto will just put your IP in there. The other values can be stun:host or an IP. The docs in trunk show this now... its really simple to understand but you should NEVER have to set that unless you have a nat scenario that requires you to lie about your IP and such to traverse the nat. /b On Nov 5, 2009, at 4:54 PM, Joseph L. Casale wrote: > The internal.xml also has an "ext-rtp-ip" variable and in trying to > understand what this is for (my version of fs is <1) I noticed in > trunks > conf file it is explained. So the available options that I have given > my setup is multihomed with a lan/wan setup where the wan interface is > dynamic would be a fqdn for fs to lookup, or auto/auto-nat. > > How exactly does auto and auto-nat work so I may know of its going to > work properly/reliably in my scenario. > > Thanks! > jlc From jerry.richards at teotech.com Thu Nov 5 15:00:50 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 5 Nov 2009 15:00:50 -0800 Subject: [Freeswitch-users] Bug in Freeswitch/scripts/gentls_cert.in build file? In-Reply-To: <19C997BDE7BB44419A2B3A1BBFEA1643@greyhawk.tonecommander.com> References: <19C997BDE7BB44419A2B3A1BBFEA1643@greyhawk.tonecommander.com> Message-ID: <776FA39C275F417B90D423E4F0866F09@greyhawk.tonecommander.com> Here is what is believed to be a bug found by Robert Hadley found in Freeswitch1.0.4/scripts/gentls_cert.in build file: Fix for "gentls_cert remove" to work: [scripts]# diff gentls_cert.in gentls_cert.in~ 129c129 < if [ -d "${CONFDIR}/CA" ]; then --- > if [ ! -d "${CONFDIR}/CA" ]; then Best Regards, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/41115d6a/attachment-0002.html From brian at freeswitch.org Thu Nov 5 15:06:01 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:06:01 -0600 Subject: [Freeswitch-users] Bug in Freeswitch/scripts/gentls_cert.in build file? In-Reply-To: <776FA39C275F417B90D423E4F0866F09@greyhawk.tonecommander.com> References: <19C997BDE7BB44419A2B3A1BBFEA1643@greyhawk.tonecommander.com> <776FA39C275F417B90D423E4F0866F09@greyhawk.tonecommander.com> Message-ID: <3F79E299-417E-461A-9DB8-10852D90B6BC@freeswitch.org> In the future please post issues to jira.freeswitch.org along with a diff -u from the root freeswitch source directory. This already seems to be fixed in svn trunk can you verify. Thanks, Brian On Nov 5, 2009, at 5:00 PM, Jerry Richards wrote: > > Here is what is believed to be a bug found by Robert Hadley found > in Freeswitch1.0.4/scripts/gentls_cert.in build file: > > Fix for "gentls_cert remove" to work: > [scripts]# diff gentls_cert.in gentls_cert.in~ > 129c129 > < if [ -d "${CONFDIR}/CA" ]; then > --- > > if [ ! -d "${CONFDIR}/CA" ]; then > > > Best Regards, > Jerry > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/69e8bd52/attachment-0002.html From JCasale at activenetwerx.com Thu Nov 5 15:20:41 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 5 Nov 2009 23:20:41 +0000 Subject: [Freeswitch-users] sip profile question In-Reply-To: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> References: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> Message-ID: >auto-nat tries to use upnp/nat-pmp to figure it out... auto will just >put your IP in there. > >The other values can be stun:host or an IP. > >The docs in trunk show this now... its really simple to understand but >you should NEVER have to set that unless you have a nat scenario that >requires you to lie about your IP and such to traverse the nat. Thanks for the fast reply Brian, so bear with me here... I am just about to go live w/ my first fs box as I move away from a year or two with Asterisk. So I don't have upnp/nat-pmp, I guess "auto" would be my next choice, but if the box is multihomed, how does it decide which of the two (well more as I am going to use vlans) ip's to stick in there? I guess I could use a public stun server, but if there is a self contained way for me to handle it, I would rather do that so that I don't have to worry about someone else's stun server being up so my fs box functions. Thanks! jlc From brian at freeswitch.org Thu Nov 5 15:27:41 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:27:41 -0600 Subject: [Freeswitch-users] sip profile question In-Reply-To: References: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> Message-ID: <5897F88C-7691-4B38-B51B-4403745C2573@freeswitch.org> If you're on a public IP you have no need for ext-rtp-ip or ext-sip-ip REMOVE them. If your multi homed then you'll need to set them.. we don't listen on 0.0.0.0 you'll have to start a profile for each IP you wish to listen on. /b On Nov 5, 2009, at 5:20 PM, Joseph L. Casale wrote: > > Thanks for the fast reply Brian, so bear with me here... I am just > about > to go live w/ my first fs box as I move away from a year or two with > Asterisk. > > So I don't have upnp/nat-pmp, I guess "auto" would be my next > choice, but if > the box is multihomed, how does it decide which of the two (well > more as I > am going to use vlans) ip's to stick in there? > > I guess I could use a public stun server, but if there is a self > contained > way for me to handle it, I would rather do that so that I don't have > to worry > about someone else's stun server being up so my fs box functions. > > Thanks! > jlc From carlos.talbot at gmail.com Thu Nov 5 15:28:14 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Thu, 5 Nov 2009 17:28:14 -0600 Subject: [Freeswitch-users] FusionPBX Message-ID: <5800526b0911051528p4ed099bdy112776128681477f@mail.gmail.com> FYI, the latest Windows SVN build now includes the option to configure FusionPBX, a port of the pfsense/FreeSWITCH gui: http://fusionpbx.com/index.php If you plan to install it someplace other than the default location of C:/FreeSWITCH just make sure to update the paths in "Admin, System Settings" from the FusionPBX web interface. The default username for the GUI is *admin*, password *fusionpbx* Here's the link: http://files.freeswitch.org/windows_installer/freeswitch.exe At this time FusionPBX utilizes sqlite for its data store. The author, mcrane, plans to release a new version soon with support for a MySQL, or PostgreSQL backend. regards, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/da306ac7/attachment-0002.html From JCasale at activenetwerx.com Thu Nov 5 15:40:57 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 5 Nov 2009 23:40:57 +0000 Subject: [Freeswitch-users] evaluate variable through cli Message-ID: How does one show the assigned value that a variable such as $${local_ip_v4} or $${domain} might have through the cli? Thanks, jlc From mrene_lists at avgs.ca Thu Nov 5 15:43:11 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 5 Nov 2009 15:43:11 -0800 Subject: [Freeswitch-users] evaluate variable through cli In-Reply-To: References: Message-ID: global_getvar local_ip_v4 Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 5-Nov-09, at 3:40 PM, Joseph L. Casale wrote: > How does one show the assigned value that a variable such as > $${local_ip_v4} or $${domain} might have through the cli? > > Thanks, > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jerry.richards at teotech.com Thu Nov 5 15:49:30 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 5 Nov 2009 15:49:30 -0800 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP Message-ID: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry From JCasale at activenetwerx.com Thu Nov 5 15:45:40 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 5 Nov 2009 23:45:40 +0000 Subject: [Freeswitch-users] sip profile question In-Reply-To: <5897F88C-7691-4B38-B51B-4403745C2573@freeswitch.org> References: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> <5897F88C-7691-4B38-B51B-4403745C2573@freeswitch.org> Message-ID: >If you're on a public IP you have no need for ext-rtp-ip or ext-sip-ip >REMOVE them. If your multi homed then you'll need to set them.. we >don't listen on 0.0.0.0 you'll have to start a profile for each IP you >wish to listen on. I am multihomed, and the wan nic is dynamic. Is there any way for me to control how it guesses the IP of a `specific` interface without the use of a third party (stun etc). Thanks, jlc From brian at freeswitch.org Thu Nov 5 15:57:18 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:57:18 -0600 Subject: [Freeswitch-users] sip profile question In-Reply-To: References: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> <5897F88C-7691-4B38-B51B-4403745C2573@freeswitch.org> Message-ID: <68EF3129-6E7B-488A-BD7F-2CFBB6EE7EC3@freeswitch.org> Just use ${local_ip_v4} then. and enable auto-restart on the sofia.conf.xml /b On Nov 5, 2009, at 5:45 PM, Joseph L. Casale wrote: > I am multihomed, and the wan nic is dynamic. Is there any way for me > to > control how it guesses the IP of a `specific` interface without the > use > of a third party (stun etc). > > Thanks, > jlc From brian at freeswitch.org Thu Nov 5 15:58:30 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:58:30 -0600 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP In-Reply-To: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> References: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> Message-ID: <23D4A048-8D9F-49A5-B86E-C1CA2B8FAFDB@freeswitch.org> This all depends on what you're doing in your dialplan if you do stuff like record it requires media and will trigger it. A sip trace or some such debug would be more helpful then a terse description of a problem. /b On Nov 5, 2009, at 5:49 PM, Jerry Richards wrote: > > I am trying to make a call through a Gateway that sends the INVITE > with no > SDP and ONLY wants the 200 OK w/SDP when the callee answers. > > For some reason, Freeswitch answers the call with 200 OK w/SDP even > before > the callee answers the phone. Is this to provide ringback? Can I > disable > that action? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Thu Nov 5 15:58:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Nov 2009 17:58:51 -0600 Subject: [Freeswitch-users] evaluate variable through cli In-Reply-To: References: Message-ID: <39C7F95D-477F-40B5-AAA1-34B80A0C2A94@freeswitch.org> vars.xml but watch out the core will trump local_ip_v4 if it happens to change. /b On Nov 5, 2009, at 5:40 PM, Joseph L. Casale wrote: > How does one show the assigned value that a variable such as > $${local_ip_v4} or $${domain} might have through the cli? > > Thanks, > jlc > > ______ From dujinfang at gmail.com Thu Nov 5 18:04:45 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Nov 2009 10:04:45 +0800 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <7b197bef0911051003t13e363edqff7b76ecfc099ed5@mail.gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> <23f91030911050957m796fe88fj5da881875c010e6b@mail.gmail.com> <7b197bef0911051003t13e363edqff7b76ecfc099ed5@mail.gmail.com> Message-ID: <23f91030911051804p31d73789o7e3a14c43c857eb1@mail.gmail.com> 2009/11/6 Giovanni Maruzzelli > On Thu, Nov 5, 2009 at 6:57 PM, Seven Du wrote: > > Ciao Giovanni, > > > > Do you still plan to merge this? > > Sorry Seven, > > I've lost track of this, and now I'm so sick I'm completely un-useful ;). > > That's OK, we all have a lot of things to do each day. > But yes, I would like to do it, if you think it is in a useful state. > > Can you please create a Jira and attach an svn diff, so in the next > days I can merge it? > > I'd like to create a jira and I think it would be easier if you can directly merge from branch. However the branch is a bit old and it would need some days if you need svn diff based on the current trunk. Thanks. > -giovanni > > > > > 2009/9/5 Giovanni Maruzzelli > >> > >> Seven, > >> > >> thanks a lot for your efforts. > >> > >> I will merge it in the next days, and I will take care that it will > >> not breaks Windows or Linux. > >> > >> If I find problems I will wait for you having more time in the future. > >> > >> I send you my super best wishes for your personal things to go well > >> and solves in the best of the possible ways. > >> > >> ciao for now, > >> > >> -giovanni > >> > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> > >> > >> > >> On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: > >> > gm, > >> > > >> > Thanks a lot you can merge into the mainline. I check into my branch > >> > because it's currently not as useful as on Linux and Windows and the > >> > solution is not good. But it works and it is a good start that > >> > mod_skypiax runs on OSX. Sure it would be easier for people want to > >> > test and improve it if it been merged into trunk. I think you can make > >> > a diff by > >> > > >> > svn diff -r 14472:14772 > >> > > http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax > >> > > >> > FYI for personal reason I won't have much time put on this in the > >> > coming month. Actually the code was done a few weeks ago but i only > >> > got a chance to commit it yesterday. Sure that is not to say I cannot > >> > do but fixes. But can you please make sure it won't break Linux/ > >> > windows build when you merge the code? I haven't have a chance to test > >> > all of them yet. > >> > > >> > -7- > >> > > >> > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: > >> >> Seeeeeeeven! > >> >> > >> >> I saw the modification you made on the wiki page... > >> >> > >> >> You made it, mod_skypiax runs on OSX!!!! > >> >> > >> >> Let's merge your mods on the mainline, pleaaaase ;-))) > >> >> > >> >> -giovanni > >> >> > >> >> > >> >> > >> >> > >> >> Sincerely, > >> >> > >> >> Giovanni Maruzzelli > >> >> Cell : +39-347-2665618 > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/ff1b4043/attachment-0002.html From msc at freeswitch.org Thu Nov 5 19:36:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Nov 2009 19:36:07 -0800 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> Message-ID: <87f2f3b90911051936k40fee09ds7d0bd237094d1df1@mail.gmail.com> If you need a really cheap entry-level phone that does Polycom's HD Siren codecs then check out the IP 335 that just came out. It's very basic but I'm hearing good things from people who've used them. -MC On Thu, Nov 5, 2009 at 6:57 AM, Rob Forman wrote: > Hey all, > > Looking at buying some high def phones. Any recommendations > (preferably based on experience) for hardware based on product > quality, standards compliance, features integration with Freeswitch, > etc? > > Thank you! > Rob Forman > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/1aeee1ef/attachment-0002.html From JCasale at activenetwerx.com Thu Nov 5 20:26:36 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Fri, 6 Nov 2009 04:26:36 +0000 Subject: [Freeswitch-users] sip profile question In-Reply-To: <68EF3129-6E7B-488A-BD7F-2CFBB6EE7EC3@freeswitch.org> References: <00E12B10-F814-4C43-BA5C-E9352B2ADB58@freeswitch.org> <5897F88C-7691-4B38-B51B-4403745C2573@freeswitch.org> <68EF3129-6E7B-488A-BD7F-2CFBB6EE7EC3@freeswitch.org> Message-ID: >Just use ${local_ip_v4} then. > >and enable auto-restart on the sofia.conf.xml Cool, it seems to always use the public ip, quite reliably. That is what I am after (why), is there something in the code that forces it to favor for example, non RFC 1918 addresses? It works, I just want to understand exactly how and why rather than be oblivious:) Thanks for all the advice! jlc From mctch at yahoo.com Thu Nov 5 21:02:21 2009 From: mctch at yahoo.com (Mark Crane) Date: Thu, 5 Nov 2009 21:02:21 -0800 (PST) Subject: [Freeswitch-users] FusionPBX In-Reply-To: <5800526b0911051528p4ed099bdy112776128681477f@mail.gmail.com> Message-ID: <616176.22865.qm@web56404.mail.re3.yahoo.com> Screenshots for the FusionPBX graphical interface http://fusionpbx.com/files/fusionpbx_com/screenshots/index.php --- On Thu, 11/5/09, Carlos Talbot wrote: From: Carlos Talbot Subject: [Freeswitch-users] FusionPBX To: freeswitch-users at lists.freeswitch.org Date: Thursday, November 5, 2009, 4:28 PM FYI, the latest Windows SVN build now includes the option to configure FusionPBX, a port of the pfsense/FreeSWITCH gui:?http://fusionpbx.com/index.php If you plan to install it someplace other than the default location of C:/FreeSWITCH just make sure to update the paths in "Admin, System Settings" from the FusionPBX web interface. The default username for the GUI is admin, password fusionpbx Here's the link:?http://files.freeswitch.org/windows_installer/freeswitch.exe At this time FusionPBX utilizes sqlite for its data store. The author, mcrane, plans to release a new version soon with support for a MySQL, or PostgreSQL backend. regards, Carlos -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091105/34c2a75e/attachment-0002.html From lakindia89 at gmail.com Thu Nov 5 22:29:02 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Fri, 6 Nov 2009 11:59:02 +0530 Subject: [Freeswitch-users] Events in mod_perl Message-ID: <7d79b3930911052229k2828ff7dic6d5b887a4897c8c@mail.gmail.com> Hi all, Is there any way to receive events while running a perl program with the help of mod_perl?? I've seen some functions related to sending and receiving events in the mod_perl wiki. But I don't know how to use that. Any help!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/5300ed7c/attachment-0002.html From msc at freeswitch.org Fri Nov 6 01:44:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Nov 2009 01:44:26 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conf Call - Nov 6 Message-ID: <87f2f3b90911060144h1a5e4e96id53fab67d26d5ec1@mail.gmail.com> FYI, The agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_11_06 It's a light agenda this week. Also, I'm out of town and will only be on the call for about an hour before I have to drop off. I'll be checking in off and on. Everyone is welcome to bring items for discussion. Talk to you tomorrow. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/bd6efbc7/attachment-0002.html From claudiu at globtel.ro Fri Nov 6 02:27:00 2009 From: claudiu at globtel.ro (Claudiu Filip) Date: Fri, 6 Nov 2009 12:27:00 +0200 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP In-Reply-To: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> References: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> Message-ID: <1766871856.20091106122700@globtel.ro> Hi Jerry, Have a look at 3pcc-enable option in your sip profile. You may want to set it "proxy", even if it's not that RFC compliant and has some issues with codec negotiation (FS advertise global_codecs to both parties and it may result in having different codecs on each leg => transcoding or call drop if transcoding not possible). Best regards, Claudiu Filip claudiu at departamentul.it Friday, November 6, 2009, 1:49:30 AM, you wrote: Jerry> I am trying to make a call through a Gateway that sends the INVITE with no Jerry> SDP and ONLY wants the 200 OK w/SDP when the callee answers. Jerry> For some reason, Freeswitch answers the call with 200 OK w/SDP even before Jerry> the callee answers the phone. Is this to provide ringback? Can I disable Jerry> that action? Jerry> Best Regards, Jerry> Jerry Jerry> _______________________________________________ Jerry> FreeSWITCH-users mailing list Jerry> FreeSWITCH-users at lists.freeswitch.org Jerry> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users Jerry> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users Jerry> http://www.freeswitch.org From mattdfong at gmail.com Fri Nov 6 04:32:59 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 6 Nov 2009 19:32:59 +0700 Subject: [Freeswitch-users] uuid_broadcast (with mux) or simultanous uuid_displace? Message-ID: <4256bf830911060432r7ca79601x220b050598b28929@mail.gmail.com> I have an application with two channels bridged, and I want to play an audio (.wav) file to both of the channels but have the ability to combine audio sources so the two participants can talk over the audio being played. when I tried to do this with uuid_broadcast specifying --both legs, it did not mix the audio. I believe this can be done with uuid_displace with the mux argument, but uuid_displace only works on one channel, (there is no --both argument). So, is it possible to execute uuid_displace twice at the same time, one for each uuid? If so, is there a single command I can give to FreeSWITCH that will combine 2 commands to be executed simultaneously? (like a & in linux) Or is there another way of performing what I'm trying to do that I'm over looking? Thanks. --matt Hello Hunter Predictive Dialer - http://www.hellohunter.com - Voice Broadcasting -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/2baffa41/attachment-0002.html From stevendt at primrosebank.net Fri Nov 6 06:20:19 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Fri, 6 Nov 2009 14:20:19 -0000 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo Wiki References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com><2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com><1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com><665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com><87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com> <97FBB4B6002848BCA4F2D89F13626754@bp1.ad.bp.com> Message-ID: Hi, I am having some (limited) success with setting up my system ! I have got an SPA-3102 connected and working after a fashion, but I don't understand how to move on. I am setting up an internal only VOIP system - it will (hopefully) use the SPA3102 to take calls from the PSTN and put them through FreeSwitch to transfer them to VOIP. The gateway will work in reverse, taking internal VOIP calls and passing them to the PSTN. So far, based on the "SPA3102 FreeSwitch HowTo Wiki" and some related information, I can get FreeSwitch to see the incoming call and pass it to the extention defined in the SPA3102 Dialplan, i.e., it rings extension 1001. I can make outgoing calls by dialing the PSTN Extension (1000) and then manually entering the PSTN number. I have an "out of the box" FreeSwitch installation, including extensions, dialplans etc. As well as being new to FreeSwitch, all of this phone stuff is new to me - Dialplans so far look like a black art ! My questions are :- Making outgoing calls. Do I need to enter 1000 everytime I make a call - I'm thinking that I should be able to setup a dialplan which knows that if I'm not entering an internal (VOIP) number (format say, xxxx), it should automatically reroute to the SPA3102 through extension 1000? I probably want to set it up so that the user maybe has to dial a number, say 9, for the outside line, but not the 1000 number. When I try appending a "real" number to 1000, I get a "CALL_REJECTED" error in the console and a "bad number" tone, I suspect that the tone is coming from FreeSwitch and not the SPA3102 - would that be right ? Receiving Incoming Calls As shown in the Wiki, the SPA3102 rings phone extension 1001. I want all internal VOIP phones to ring and be available for answer from all phones - again, I think this should be possible, but I have no idea how to achieve it. I'm guessing that I'd create a "dummy" extension that the SPA3102 would call, which a dialplan would then distribute to a group of VOIP extensions ? Any help/pointers would be really appreciated. I will probably try IRC later, but with the time difference, it's a bit awkward for me (I'm in the UK) - I can't face many more late nights ! Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/ad2cef33/attachment-0002.html From stevendt at primrosebank.net Fri Nov 6 06:20:33 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Fri, 6 Nov 2009 14:20:33 -0000 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo Wiki References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com><2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com><1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com><665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com><87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com> <97FBB4B6002848BCA4F2D89F13626754@bp1.ad.bp.com> Message-ID: <41A5CF92E4E94547BCE301E0F5A5B79B@bp1.ad.bp.com> Hi, I am having some (limited) success with setting up my system ! I have got an SPA-3102 connected and working after a fashion, but I don't understand how to move on. I am setting up an internal only VOIP system - it will (hopefully) use the SPA3102 to take calls from the PSTN and put them through FreeSwitch to transfer them to VOIP. The gateway will work in reverse, taking internal VOIP calls and passing them to the PSTN. So far, based on the "SPA3102 FreeSwitch HowTo Wiki" and some related information, I can get FreeSwitch to see the incoming call and pass it to the extention defined in the SPA3102 Dialplan, i.e., it rings extension 1001. I can make outgoing calls by dialing the PSTN Extension (1000) and then manually entering the PSTN number. I have an "out of the box" FreeSwitch installation, including extensions, dialplans etc. As well as being new to FreeSwitch, all of this phone stuff is new to me - Dialplans so far look like a black art ! My questions are :- Making outgoing calls. Do I need to enter 1000 everytime I make a call - I'm thinking that I should be able to setup a dialplan which knows that if I'm not entering an internal (VOIP) number (format say, xxxx), it should automatically reroute to the SPA3102 through extension 1000? I probably want to set it up so that the user maybe has to dial a number, say 9, for the outside line, but not the 1000 number. When I try appending a "real" number to 1000, I get a "CALL_REJECTED" error in the console and a "bad number" tone, I suspect that the tone is coming from FreeSwitch and not the SPA3102 - would that be right ? Receiving Incoming Calls As shown in the Wiki, the SPA3102 rings phone extension 1001. I want all internal VOIP phones to ring and be available for answer from all phones - again, I think this should be possible, but I have no idea how to achieve it. I'm guessing that I'd create a "dummy" extension that the SPA3102 would call, which a dialplan would then distribute to a group of VOIP extensions ? Any help/pointers would be really appreciated. I will probably try IRC later, but with the time difference, it's a bit awkward for me (I'm in the UK) - I can't face many more late nights ! Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/a999ee69/attachment-0002.html From rob4manhere at gmail.com Fri Nov 6 06:32:07 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Fri, 6 Nov 2009 08:32:07 -0600 Subject: [Freeswitch-users] Wideband / HD phones In-Reply-To: <87f2f3b90911051936k40fee09ds7d0bd237094d1df1@mail.gmail.com> References: <654F823C-36C7-4605-9A02-788834C9685C@gmail.com> <87f2f3b90911051936k40fee09ds7d0bd237094d1df1@mail.gmail.com> Message-ID: <993FA8DA-7A3A-4FF7-BA33-168CFF96B4FC@gmail.com> Great- good to know. Thanks for all the responses! Rob On Nov 5, 2009, at 9:36 PM, Michael Collins wrote: > If you need a really cheap entry-level phone that does Polycom's HD > Siren codecs then check out the IP 335 that just came out. It's very > basic but I'm hearing good things from people who've used them. > -MC > > On Thu, Nov 5, 2009 at 6:57 AM, Rob Forman > wrote: > Hey all, > > Looking at buying some high def phones. Any recommendations > (preferably based on experience) for hardware based on product > quality, standards compliance, features integration with Freeswitch, > etc? > > Thank you! > Rob Forman > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/aa3ed3a1/attachment-0002.html From mariusz_kolo at wp.pl Fri Nov 6 06:38:31 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=) Date: Fri, 06 Nov 2009 15:38:31 +0100 Subject: [Freeswitch-users] Problem with hangin bri In-Reply-To: <4AEEC368.5020204@wp.pl> References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com> <87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com> <4AE8B3B6.9000106@wp.pl> <87f2f3b90910281528t633b765x7c2bb858f41ce154@mail.gmail.com> <4AEEC368.5020204@wp.pl> Message-ID: <4AF434E7.6070904@wp.pl> Hi I'm testing behaviour on our Bri Card and i see that only incoming calls hangin up channels pastebin: http://pastebin.freeswitch.org/11020 At start oz dump for 1 and 2 channels are: API CALL [oz(dump 1 1)] output: span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE API CALL [oz(dump 1 2)] output: span_id: 1 chan_id: 2 physical_span_id: 1 physical_chan_id: 2 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE after first incoming call, system chose 2 channel (in log: 2009-11-06 14:46:34.341595 .... Processing 609381316->717949433 ....) we have: API CALL [oz(dump 1 2)] output: span_id: 1 chan_id: 2 physical_span_id: 1 physical_chan_id: 2 type: B state: HANGUP last_state: PROGRESS cid_date: cid_name: 609381316 cid_num: 609381316 ani: 609381316 aniII: dnis: 717949433 rdnis: cause: NORMAL_CLEARING after second incoming call, system chose 1 channel propably because 2 channel is still in use (in log: 2009-11-06 14:47:07.361596 .... Processing 609381316->717949433 ...) we have: API CALL [oz(dump 1 1)] output: span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: B state: HANGUP last_state: PROGRESS cid_date: cid_name: 609381316 cid_num: 609381316 ani: 609381316 aniII: dnis: 717949433 rdnis: cause: NORMAL_CLEARING All next incoming calls has warning 2009-11-06 14:47:40.093836 [WARNING] ozmod_isdn.c:829 Channel 1:2 ~ 1:2 is already in use waiting for it to become available. All next outbound calls has: Warning 2009-11-06 14:48:16.925599 [ERR] mod_openzap.c:1154 No channels available Only unload mod_openzap and load mod_openzap release channels I think channel's state after call should be DOWN not HANGUP If i make only outgoing calls channel's state returns to DOWN Please check it out. I hope my logs can help Thanks Mariusz Ko?odziejczyk pisze: > Hi > > pastebin: > > http://pastebin.freeswitch.org/10926 > and > http://pastebin.freeswitch.org/10927 > > .We invoke calls from one voip phone to cell phone, and vice versa, but > when i make inbound and outbound connection in nearly same time > something goes wrong with chanells > > Thanks > > > > Michael Collins pisze: > >> Thanks. Can you collect debug logs of this happening? See >> http://wiki.freeswitch.org/wiki/Reporting_Bugs for helpful tips on >> collecting debug information. Use pastebin to dump all the log info >> and reply here with the link. We don't have too many BRI users but I >> believe there are a few so hopefully we can help you get up and running. >> -MC >> >> 2009/10/28 Mariusz Ko?odziejczyk > > >> >> Hi >> >> I'm also working on this project, so i can answer your questions >> >> Which version of FreeSWITCH are you running? >> >> FreeSWITCH Version 1.0.trunk (15246) >> >> Which PRI library are you using? >> openzap Native stack >> >> openzap.conf >> >> [span zt BRI1] >> trunk_type => bri >> b-channel => 1-2 >> d-channel=> 3 >> >> openzap.conf >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Which BRI card are you using? >> >> Producer: http://www.phoniceq.com/ >> card model: http://quadbri.phoniceq.com/ >> >> Card instalation process (instruction from producer) >> >> 1) download bristuff staff from >> >> http://junghanns.net/downloads/bristuff-0.4.0-RC3h.tar.gz >> or >> http://junghanns.net/downloads/bristuff-0.3.0-PRE-1y-z.tar.gz >> >> unpack it and go to bristuff-* >> >> 2) download patcher from >> http://quadbri.phoniceq.com/driver/bristuff/qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch >> >> patch it using >> >> patch -p0 < qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch >> >> 3) you can check card using zttest (result should be 99.x) >> >> Producer has said, that we are first client, it wants to use this >> card in freeswitch >> >> we are using 1 port (S/T interface). Our NT is "NT1 plus 2b1q" >> >> >> Thanks >> >> Michael Collins pisze: >> > Okay, obligatory questions: >> > Which version of FreeSWITCH are you running? >> > Which PRI library are you using? >> > Which BRI card are you using? >> > >> > -MC >> > >> > On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawli?ski >> >> > >> wrote: >> > >> > Hi, >> > I have some problems with bri status. I have 3 chanel isdn >> modem, >> > and zaptel compatible quad bri card. I can invoke calls from my >> > voip phone to cell phone, and vice versa, but when i make >> inbound >> > and outbound connection in nearly same time something goes wrong >> > with chanells and after few calls all of them has hangup status. >> > >> > There is log about that in attachement, see "is already in use >> > waiting for it to become available." phrase. Time of this >> event is >> > about 14:43:35. Unload and Load open_zap module helped, but its >> > not an solution because of lost connections. >> > >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> -- >> Mariusz Ko?odziejczyk >> >> Advanced Developing Architecture S.C. >> >> tel. : +48 609 381 316 >> e-mail : mariusz_kolo at wp.pl >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- Mariusz Ko?odziejczyk Advanced Developing Architecture S.C. tel. : +48 609 381 316 e-mail : mariusz_kolo at wp.pl From gshfreesw at gmail.com Fri Nov 6 07:11:34 2009 From: gshfreesw at gmail.com (Shameem Shiek) Date: Fri, 6 Nov 2009 10:11:34 -0500 Subject: [Freeswitch-users] Register multiple DID/extensions with the same provider. Message-ID: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> Hello, I have created a provider configuration file XML file *in conf/directory/default/myprovider.xml . *This configuration has the line: How do I add another DID from the same provider? Do I add another line like this ? or create a new provider XML config file? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/f877531d/attachment-0002.html From brian at freeswitch.org Fri Nov 6 07:23:19 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Nov 2009 09:23:19 -0600 Subject: [Freeswitch-users] Register multiple DID/extensions with the same provider. In-Reply-To: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> References: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> Message-ID: <72FFFD32-4A62-474E-A9D5-497660FD380B@freeswitch.org> Does your provider require you to register once for every did? /b On Nov 6, 2009, at 9:11 AM, Shameem Shiek wrote: > Hello, > > I have created a provider configuration file XML file in conf/ > directory/default/myprovider.xml . This configuration has the line: > > > > > How do I add another DID from the same provider? Do I add another > line like this ? or create a new provider XML config file? > > Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/e013af09/attachment-0002.html From rob4manhere at gmail.com Fri Nov 6 07:30:38 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Fri, 6 Nov 2009 09:30:38 -0600 Subject: [Freeswitch-users] Register multiple DID/extensions with the same provider. In-Reply-To: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> References: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> Message-ID: Hi Shameem, If you're just getting another DID under the same provider registration, that should go under your public dialplan (conf/dialplan/ public) and then route to the extension or application of your choice. Rob On Nov 6, 2009, at 9:11 AM, Shameem Shiek wrote: > Hello, > > I have created a provider configuration file XML file in conf/ > directory/default/myprovider.xml . This configuration has the line: > > > > > How do I add another DID from the same provider? Do I add another > line like this ? or create a new provider XML config file? > > Thanks! > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/9660f825/attachment-0002.html From ujjval at simplesignal.com Fri Nov 6 07:59:42 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Fri, 6 Nov 2009 07:59:42 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175B572244@EXMBXCLUS01.citservers.local> Any examples I can refer to for this? Like for Channel vars and execute_application calls? Does this all need to be doen in dialplan.public.xml or also in other config files? Sorry: I am still learning the Freeswitch world. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Thursday, November 05, 2009 8:56 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator This is true, BUT it is more flexible than it looks. http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E The caller controls can have a key execute a dialplan extension: execute_application You can set a channel var on the moderator prior to joining to the conf. When the extenion is called, you can check the channel var for moderator and act accordingly. Or you can send an event and monitor with an app over ESL and do whatever you want there (probably using the same channel var trick for knowing who is a mod or not). On Thu, Nov 5, 2009 at 8:52 AM, Rob Forman wrote: > Hi UK, > > ?From what I've done and read, the caller-controls (in > conference.conf.xml) can be modified to almost anything you can think > of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a > caller control just for those with the moderator flag. ?So unless you > want everyone able to mute/kick everyone then you can't do it. > > The wiki seems to indicate this as well: > > "Be aware that the caller-controls are applied across the entire > conference. You cannot enter one member of the conference using caller- > controls ABC and then enter a second member using caller-controls XYZ." > > http://wiki.freeswitch.org/wiki/Mod_conference > > > I think this might be a limitation of mod_conference. ?Perhaps one of > the pros can chime in if I'm off-base or there's some nifty way to > accomplish this. > > Cheers, > Rob > > On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > >> Any ideas on the below...has anyone implemented the below: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, November 02, 2009 12:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Rob: >> >> ? Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 9:34 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hm, strange. ?I haven't seen that before. ?Can you pastebin your logs >> at debug level? >> >> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >> >>> It's strange... a tcpdump tells me that there is no DTMF from my >>> provider when using IVR, but when I call into a TN that goes >>> directly into the Conference App, I see DTMF from the provider. >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 7:23 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I've never had any problem with that. ?Is your logging at debug level >>> so you can see the RECV DTFM in the log/fs_cli? ?Are you calling from >>> a SIP phone on the pbx, or via a PSTN provider? ?Maybe your provider >>> isn't passing them through. >>> >>> Make sure your logging is turned up then try something simpler, like >>> calling the echo application, and see if DTFM comes through. >>> >>> Rob >>> >>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>> >>>> Rob: >>>> >>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>> using the below configuration....so it basically timesout. >>>> >>>> UK >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Ujjval Karihaloo >>>> Sent: Monday, October 26, 2009 9:21 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>> correct.. >>>> =============== >>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. ?I guess that's why I didn't understand >>>> the point of the +pin. >>>> ====================== >>>> >>>> I will try it out. >>>> >>>> Again thx a lot for your help. Will keep everyone posted. >>>> >>>> ________________________________________ >>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>> Sent: Friday, October 23, 2009 12:22 PM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> I just re-tested with the pin in my dial plan: >>>> >>>> >>>> >>>> And it doesn't challenge me for the pin. ?I just drop right in. ?I >>>> figured this is how it was intended, since the wiki says the pin is >>>> set initially and only challenged in later attempts [by future >>>> callers]: >>>> >>>> "The first time a conference name (confname) is used, it will be >>>> created on demand, and the pin will be set to what ever is specified >>>> at that time: the pin in the data string if specified, or if not, >>>> the >>>> "pin" setting in the conference profile, and if that is also >>>> unspecified, then there is no pin protection. Any later attempt to >>>> join the conference must specify the same pin number, if one existed >>>> when it was created. " >>>> >>>> >>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. ?I guess that's why I didn't understand >>>> the point of the +pin. >>>> >>>> I'm sure there's a scenario where its used and useful, the wiki just >>>> doesn't explain it. >>>> >>>> Rob >>>> >>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>> >>>>> Well first off you're not defining a pine here... >>>>> >>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>> pin >>>>> number] >>>>> >>>>> That might be why its not asking for a pin. >>>>> >>>>> /b >>>>> >>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>> >>>>>> ? >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jmesquita at freeswitch.org Fri Nov 6 08:31:09 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 6 Nov 2009 14:31:09 -0200 Subject: [Freeswitch-users] Events in mod_perl In-Reply-To: <7d79b3930911052229k2828ff7dic6d5b887a4897c8c@mail.gmail.com> References: <7d79b3930911052229k2828ff7dic6d5b887a4897c8c@mail.gmail.com> Message-ID: I don't know what you are trying to do exactly but I think that you might need to you ESL instead. Why don't you take a look at all the examples inside ${SVNROOT}/libs/esl and see if that fits you? I have a hunch that it would. JM On Fri, Nov 6, 2009 at 4:29 AM, lakshmanan ganapathy wrote: > Hi all, > Is there any way to receive events while running a perl program with > the help of mod_perl?? > > I've seen some functions related to sending and receiving events in the > mod_perl wiki. But I don't know how to use that. > Any help!!! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/fa2c4b02/attachment-0002.html From andrewkt at aktzero.com Fri Nov 6 09:59:47 2009 From: andrewkt at aktzero.com (Andrew Thompson) Date: Fri, 06 Nov 2009 12:59:47 -0500 Subject: [Freeswitch-users] Register multiple DID/extensions with the same provider. In-Reply-To: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> References: <5070fcbd0911060711v4243bf65n8ace1ef831e0bd37@mail.gmail.com> Message-ID: <4AF46413.7070905@aktzero.com> On 11/6/2009 10:11 AM, Shameem Shiek wrote: > How do I add another DID from the same provider? Do I add another line > like this ? or create a new provider XML config file? I have two providers that send me DIDs in different ways. One of them does not provide the number that was dialed, so I have to sub-account and register uniquely for each DID. The other does provide the number dialed and I create seperate extensions in conf/dialplan/public/NPANXXXXXX.xml. TLDR: Test inbound calls and see if you get unique destination_number for each DID. -- Andrew Thompson From mrene_lists at avgs.ca Fri Nov 6 10:33:17 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 6 Nov 2009 10:33:17 -0800 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP In-Reply-To: <1766871856.20091106122700@globtel.ro> References: <83409D0E0CBF4D269D101253723D842B@greyhawk.tonecommander.com> <1766871856.20091106122700@globtel.ro> Message-ID: Are you recording? I recall a recent change to force answer whenever record_session is called. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 6-Nov-09, at 2:27 AM, Claudiu Filip wrote: > > > Hi Jerry, > > > Have a look at 3pcc-enable option in your sip profile. You may want > to set it "proxy", even if it's not that RFC compliant and has some > issues with codec negotiation (FS advertise global_codecs to both > parties and it may result in having different codecs on each leg => > transcoding or call drop if transcoding not possible). > > > Best regards, > > Claudiu Filip > claudiu at departamentul.it > > > Friday, November 6, 2009, 1:49:30 AM, you wrote: > Jerry> I am trying to make a call through a Gateway that sends the > INVITE with no > Jerry> SDP and ONLY wants the 200 OK w/SDP when the callee answers. > > Jerry> For some reason, Freeswitch answers the call with 200 OK w/ > SDP even before > Jerry> the callee answers the phone. Is this to provide ringback? > Can I disable > Jerry> that action? > > Jerry> Best Regards, > Jerry> Jerry > > > Jerry> _______________________________________________ > Jerry> FreeSWITCH-users mailing list > Jerry> FreeSWITCH-users at lists.freeswitch.org > Jerry> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > Jerry> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > Jerry> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From codecomplete at free.fr Fri Nov 6 11:16:40 2009 From: codecomplete at free.fr (Fred-145) Date: Fri, 6 Nov 2009 11:16:40 -0800 (PST) Subject: [Freeswitch-users] Does OpenZap support CTR21? In-Reply-To: <7FD19B47-C121-48CD-98C2-2830BFDF1068@jerris.com> References: <26217371.post@talk.nabble.com> <7FD19B47-C121-48CD-98C2-2830BFDF1068@jerris.com> Message-ID: <26230864.post@talk.nabble.com> Thanks for the feedback. I've never installed this type of card with Freeswitch. Am I correct in understanding that I just need to download and compile the latest Zaptel/Dahdi source from the Asterisk web site, and then install Freeswitch and OpenZap, and it'll work (provided the card works with my hardware)? -- View this message in context: http://old.nabble.com/Does-OpenZap-support-CTR21--tp26217371p26230864.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Russell.Mosemann at cune.org Fri Nov 6 12:04:58 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Fri, 6 Nov 2009 20:04:58 -0000 Subject: [Freeswitch-users] Does OpenZap support CTR21? In-Reply-To: <26230864.post@talk.nabble.com> Message-ID: <20091106200458.AACDC3E5BEF@mail.cune.org> Fred-145 said: > I've never installed this type of card with Freeswitch. Am I correct in > understanding that I just need to download and compile the latest > Zaptel/Dahdi source from the Asterisk web site, and then install Freeswitch > and OpenZap, and it'll work (provided the card works with my hardware)? Yes, it should just work. I'd recommend Dahdi (complete), because Zaptel is not being developed anymore. Check the wiki for little things you have to set in various files, such as /etc/dahdi/modules /etc/dahdi/system.conf wherever/freeswitch/conf/openzap.xml wherever/freeswitch/conf/zt.conf (shouldn't have to change it) wherever/freeswitch/autoload_configs/openzap.conf.xml -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From orien at tx.rr.com Fri Nov 6 15:21:23 2009 From: orien at tx.rr.com (Orien Love) Date: Fri, 06 Nov 2009 17:21:23 -0600 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: References: Message-ID: <4AF4AF73.8070804@tx.rr.com> First of all, Thanks to the help I received on my pfSense installation, especially to Michael. I have a basic test system up and running. I am still waiting on some hardware but the base system is working!!!! I am looking on advice on how to set up a simple office PBX, 20 phones and 4 outside lines.with 2 or 3 "operator" phones and the rest will be extensions. Here is my plan, please let me know if it does not make sense, or if I am going about it System Hardware 4 spa3000's to handle the outside lines. 2-3 polycom 601 phones with expansion modules (Operator phones) 18 polycom 330 or other phones for desks. 2-24 port cisco POE switches 1 pfSense server. System Design. Extension Numbers 2xx Outside line access 1xxxxxxxxxx groups 3xx auto-attendent ??? here are my questions #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of memory handle this proposed system? (Here is the MB I am thing of using MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) #2 how do I pool my spa 3000 FXO lines so that the outgoing calls use the first available line? also how do insure that metro (non long distance) calls go to a specific line if available? I have learned a lot on how to set up Polycom 601 phones, I am planning on writing a how to document, is there any specific format? Thanks Orien From stevendt at primrosebank.net Fri Nov 6 15:59:09 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Fri, 6 Nov 2009 23:59:09 -0000 Subject: [Freeswitch-users] Valid Dial Strings References: <4AF4AF73.8070804@tx.rr.com> Message-ID: <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> Hi, can someone pointme to where the valid dialing strings are specified ? I'm assuming that something, somewhere, tells FS that numbers are invalid before they get dialed ? regards Dave From anthony.minessale at gmail.com Fri Nov 6 16:02:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Nov 2009 18:02:06 -0600 Subject: [Freeswitch-users] uuid_broadcast (with mux) or simultanous uuid_displace? In-Reply-To: <4256bf830911060432r7ca79601x220b050598b28929@mail.gmail.com> References: <4256bf830911060432r7ca79601x220b050598b28929@mail.gmail.com> Message-ID: <191c3a030911061602s4d1ca7fh9222460f8c54a951@mail.gmail.com> no, sorry. We have to draw the line somewhere. You are stepping outside the boundary of what you can do in a 2 channel bridge. Transfer them into a conference and do it there. On Fri, Nov 6, 2009 at 6:32 AM, Matthew Fong wrote: > I have an application with two channels bridged, and I want to play an > audio (.wav) file to both of the channels but have the ability to combine > audio sources so the two participants can talk over the audio being played. > when I tried to do this with uuid_broadcast specifying --both legs, it did > not mix the audio. > > I believe this can be done with uuid_displace with the mux argument, but > uuid_displace only works on one channel, (there is no --both argument). So, > is it possible to execute uuid_displace twice at the same time, one for each > uuid? If so, is there a single command I can give to FreeSWITCH that will > combine 2 commands to be executed simultaneously? (like a & in linux) Or is > there another way of performing what I'm trying to do that I'm over looking? > > Thanks. > > --matt > Hello Hunter > Predictive Dialer - http://www.hellohunter.com - Voice Broadcasting > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/e13d7647/attachment-0002.html From anthony.minessale at gmail.com Fri Nov 6 16:08:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Nov 2009 18:08:48 -0600 Subject: [Freeswitch-users] Dialpan: try.. finally analogs In-Reply-To: References: Message-ID: <191c3a030911061608w5be8af61y7bc10fe2d23dfc4a@mail.gmail.com> If you know the reason, why are you so puzzled by it? I think you should not assume you understand what is happening unless you really do. I think you need to provide an exact description of what you are doing so I can explain to you where you are making the mistake. Make sure you are on latest SVN and reproduce this in a console log for us and add an exact description of what you are doing in detail. On Thu, Nov 5, 2009 at 11:44 AM, Artem Shiyanov wrote: > Hello! > > I have to deal with classic problem: "Leaking stream handle" in FS console. > I also know the reason - firstly channel is sent to the extension with > "playback" and later it is transfered to another extensions with > "execute_extension" or, another trouble-case - channel is bridged to some > addres. > I do not ask (but I wish to) why FS doesn't close stream automatically when > channel is gone. > I ask whether it is possible to use some "try.. finally" construction in > diaplan? If "yes" then I can simply stop playback in the "finally" block.. > > Any thoughs? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091106/f48d592f/attachment-0002.html From lakindia89 at gmail.com Fri Nov 6 21:38:44 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Sat, 7 Nov 2009 11:08:44 +0530 Subject: [Freeswitch-users] Events in mod_perl In-Reply-To: References: <7d79b3930911052229k2828ff7dic6d5b887a4897c8c@mail.gmail.com> Message-ID: <7d79b3930911062138l262db9bema29c620a7b8cef94@mail.gmail.com> Ya. I have done that event processing with ESL. But I wanted to know, whether in mod_perl, we can get the events and process it or not. I've seen function's like events_get etc.. But I don't know how to use those things. In mod_perl if I'm able to get the events, then it will be easier for me. Is it possible!!! 2009/11/6 Jo?o Mesquita > I don't know what you are trying to do exactly but I think that you might > need to you ESL instead. > > Why don't you take a look at all the examples inside ${SVNROOT}/libs/esl > and see if that fits you? I have a hunch that it would. > > JM > > On Fri, Nov 6, 2009 at 4:29 AM, lakshmanan ganapathy > wrote: > >> Hi all, >> Is there any way to receive events while running a perl program with >> the help of mod_perl?? >> >> I've seen some functions related to sending and receiving events in the >> mod_perl wiki. But I don't know how to use that. >> Any help!!! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/9b5f49d3/attachment-0002.html From mayamatakeshi at gmail.com Sat Nov 7 04:47:36 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Sat, 7 Nov 2009 21:47:36 +0900 Subject: [Freeswitch-users] ParkServer: how to publish orbit status Message-ID: <15b9404e0911070447r78b95d9cv7178d0ce735d7d80@mail.gmail.com> Hello, it is my understanding that FreeSWITCH doesn't provide a ParkServer per se. So, to provide for this, I will have an entry in the dialplan to play MOH on the channel continuously till someone retrieve the call. And then, I need to publish a NOTIFY to all subscribed users informing the status of the park orbit. >From the wiki (http://wiki.freeswitch.org/wiki/PRESENCE_IN_event_example), it seems I could use "sendevent PRESENCE_IN" for this, but I was unable to figure out how the message must be sent to fill all required attributes in the xml. >From the examples, I managed to send a NOTIFY with a content like this confirmed but I actually need to send it like this: confirmed So, is it possible to set the lacking attributes? (I've tried to set content-length and pass the xml to sendevent, but nothing changed). br, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/add11ed6/attachment-0002.html From ruda at ruda.com.br Sat Nov 7 05:24:38 2009 From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=) Date: Sat, 7 Nov 2009 10:24:38 -0300 Subject: [Freeswitch-users] Fwd: Microsoft Exchange 2007 - Moved Temporarily In-Reply-To: <7600794b0911070523n41a0351co8a31bfc42128e683@mail.gmail.com> References: <7600794b0911070523n41a0351co8a31bfc42128e683@mail.gmail.com> Message-ID: <7600794b0911070524m191bf5dfmb8e42f8c775b2c29@mail.gmail.com> I'm having a problem when using Microsoft Exchange 2007 with the FreeSWITCH 1.0.trunk Windows on the same server (127.0.0.1) I set everything right and I am analyzing packages Loopback, only when Exchange sends the FreeSWITCH Moved Temporarily not move. He is sending the INVITE at the same time send port to the port that was moved. How can I fix this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/a845e807/attachment-0002.html From rupa at rupa.com Sat Nov 7 05:43:03 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Sat, 7 Nov 2009 05:43:03 -0800 Subject: [Freeswitch-users] Fwd: Microsoft Exchange 2007 - Moved Temporarily In-Reply-To: <7600794b0911070524m191bf5dfmb8e42f8c775b2c29@mail.gmail.com> References: <7600794b0911070523n41a0351co8a31bfc42128e683@mail.gmail.com> <7600794b0911070524m191bf5dfmb8e42f8c775b2c29@mail.gmail.com> Message-ID: I don't believe this is a supported configuration. If you really must run both on the same server, put FS and Exchange on different IP addresses and see if that works. SIP client/server on the same IP has issues. On Sat, Nov 7, 2009 at 5:24 AM, Rud? Cunha wrote: > I'm having a problem when using Microsoft Exchange 2007 with the FreeSWITCH > 1.0.trunk Windows on the same server (127.0.0.1) > I set everything right and I am analyzing packages Loopback, only when > Exchange sends the FreeSWITCH Moved Temporarily not move. He is sending the > INVITE at the same time send port to the port that was moved. > > How can I fix this? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From steve at justfone.com Sat Nov 7 06:26:38 2009 From: steve at justfone.com (Steven Brown) Date: Sat, 7 Nov 2009 14:26:38 +0000 Subject: [Freeswitch-users] leg_delay_start Message-ID: <3e6d7b0c0911070626g32af550fsd80e99d8266a7aa8@mail.gmail.com> Hi I've been trying to experiment with leg_delay_start when bridging to two mobiles via a gateway, however regardless of settings both legs are bridged immediately. I noticed a previous post on problems with leg_delay_start which seemed to go unanswered, just wondered if there is a known issue or if its something I'm doing wrong. Using FS 1.0.3 Dialplan extract as follows : Any pointers appreciated Thanks Steven Brown email steve at justfone.com office 08707706968 mobile 07768755409 fax 07884636663 Justfone - Company Reg. No. : 3926817 Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW The contents of this e-mail may be privileged and are confidential. It may not be disclosed to or used by anyone other than the addressee(s), nor copied in any way. If received in error, please advise sender, then delete it from your system. Internet email communications are not secure and therefore Justfone do not accept legal responsibility for the contents of this message. Any views or opinions presented are solely those of the author and do not necessarily represent those of Justfone unless otherwise specifically stated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/a8086ca8/attachment-0002.html From stevendt at primrosebank.net Sat Nov 7 06:42:09 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Sat, 7 Nov 2009 14:42:09 -0000 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo Wiki - HELP Please! References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com><87f2f3b90911041023h1cb5c069g9376d051fb985065@mail.gmail.com><688D289388594B0F97D89667D6F7E8F5@bp1.ad.bp.com><33167DE5-D670-46F0-BECA-4802B917E206@jerris.com><9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com><2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com><1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com><665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com><87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com><97FBB4B6002848BCA4F2D89F13626754@bp1.ad.bp.com> <41A5CF92E4E94547BCE301E0F5A5B79B@bp1.ad.bp.com> Message-ID: <7B9E5C0A81154EDB8B2F2979EAF0BA17@bp1.ad.bp.com> Follow up to previous post..... regarding making outgoing calls. I ***think*** that I have configured a dialplan that allows the user to dial out but the requests seem to be getting rejected by the SPA3102. I can dial 0 and the FreeSwitch attendant will connect to the PSTN line (FreeSwitch reports that the call has been answered). Similarly, I can dial 1000 - the SPA3102 extension number with similar results. However, if a try to dial an external number, the gateway rejects the call. I have captured some of the debug log but the info in there is way over my head, can anyone help me understand what it's telling me please ? regards Dave ----- Original Message ----- From: Dave Stevenson To: freeswitch-users at lists.freeswitch.org Sent: Friday, November 06, 2009 2:20 PM Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo Wiki Hi, I am having some (limited) success with setting up my system ! I have got an SPA-3102 connected and working after a fashion, but I don't understand how to move on. I am setting up an internal only VOIP system - it will (hopefully) use the SPA3102 to take calls from the PSTN and put them through FreeSwitch to transfer them to VOIP. The gateway will work in reverse, taking internal VOIP calls and passing them to the PSTN. So far, based on the "SPA3102 FreeSwitch HowTo Wiki" and some related information, I can get FreeSwitch to see the incoming call and pass it to the extention defined in the SPA3102 Dialplan, i.e., it rings extension 1001. I can make outgoing calls by dialing the PSTN Extension (1000) and then manually entering the PSTN number. I have an "out of the box" FreeSwitch installation, including extensions, dialplans etc. As well as being new to FreeSwitch, all of this phone stuff is new to me - Dialplans so far look like a black art ! My questions are :- Making outgoing calls. Do I need to enter 1000 everytime I make a call - I'm thinking that I should be able to setup a dialplan which knows that if I'm not entering an internal (VOIP) number (format say, xxxx), it should automatically reroute to the SPA3102 through extension 1000? I probably want to set it up so that the user maybe has to dial a number, say 9, for the outside line, but not the 1000 number. When I try appending a "real" number to 1000, I get a "CALL_REJECTED" error in the console and a "bad number" tone, I suspect that the tone is coming from FreeSwitch and not the SPA3102 - would that be right ? Receiving Incoming Calls As shown in the Wiki, the SPA3102 rings phone extension 1001. I want all internal VOIP phones to ring and be available for answer from all phones - again, I think this should be possible, but I have no idea how to achieve it. I'm guessing that I'd create a "dummy" extension that the SPA3102 would call, which a dialplan would then distribute to a group of VOIP extensions ? Any help/pointers would be really appreciated. I will probably try IRC later, but with the time difference, it's a bit awkward for me (I'm in the UK) - I can't face many more late nights ! Regards Dave ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/f2a55514/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Copy of freeswitch.zip Type: application/x-zip-compressed Size: 3248 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/f2a55514/attachment-0002.bin From fsdevlist at seanf.me Sat Nov 7 00:59:57 2009 From: fsdevlist at seanf.me (Sean Ferguson) Date: Sat, 7 Nov 2009 03:59:57 -0500 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote Message-ID: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> FreeSWITCH seems to be unable to read MP3 files, citing that it's an unknown format. Looking through the log, I found this during startup: 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_sync_wrote** There don't seem to be any compile-time errors, yet I can't seem to eliminate this issue. Any help would be appreciated. From john_re at fastmail.us Sat Nov 7 05:52:53 2009 From: john_re at fastmail.us (john_re) Date: Sat, 07 Nov 2009 05:52:53 -0800 Subject: [Freeswitch-users] Hi - Nov 7 TODAY & Nov 22 - Join Global FreeSW GNU(Linux) HW Culture meeting via VOIP - BerkeleyTIP GlobalTIP - For Forwarding Message-ID: <1257601973.13636.1344039469@webmail.messagingengine.com> Hi Anthony, FreeSWITCH list members. :) 1) Anthony, is it you who's in charge of this mail list & group? 2) Starting June 2008 I've been putting together a global Free SW, HW & Culture group, BerkeleyTIP, & GlobalTIP, which meets via VOIP globally. :) It's purpose is educational, productive & social. TIP = Talks, Installfest & Project/Programming Party. I do all this in my small "spare/free" time as a contribution to the communit(ies). {Thanks for all the SW! :) } http://sites.google.com/site/berkeleytip/ The meetings are on the 1st Saturday & 3rd Sunday of each month, from 12N-3PM Pacific usa UTC-8hr time. We are currently looking at moving from Asterisk to freeswitch on our voip server box. http://sites.google.com/site/berkeleytip/remote-attendance I'm writing to invite you all to join with the global BTIP community & attend online. Join #berkeleytip on irc.freenode.net, & we'll help you get on our VOIP conference. [I'm sure you all could probably help us way more than we could help you. ;) ] We also encourage all groups (such as the freeswitch community) to hold a simultaneous meeting, & join together with us all online. I know you already have a friday voip conference, iirc. If all free sw groups had a simultaneous meeting to BerkeleyTIP, it would be like a global meeting/conference - everyone would know that at that time they could meet all their friends & community members online on voip conferences, from all the free sw projects in the world. 3) May I send to this mailing list the monthly BTIPGlobal announcement? This month's announcement is included right below. Best Wishes, John Re =================================================================== CONTENTS: Meeting days/times & Howto - Mark your calendar's dates; Videos; Hot topics; Opportunities; Announcement Flyers; New webpages ===== Come join in with the Global Free SW HW & Culture community at the BerkeleyTIP/GlobalTIP meeting, via VOIP. Two meetings this month: Sat Nov 7, 12Noon - 3PM Pacific Time (=UTC-8) Sun Nov 22, 12Noon - 3PM Pacific Time (=UTC-8) Mark your calendars, 1st Sat, 3rd Sun every month. {Note: 4th Sunday this November, to give 2 week spacing.} Join online #berkeleytip on irc.freenode.net & we'll help you get your voip HW & SW working: http://sites.google.com/site/berkeleytip/remote-attendance Or come to the FreeSpeech Cafe at UC Berkeley in person meeting. Join the global mailing list http://groups.google.com/group/BerkTIPGlobal I hope to see you there. :) ===== Talk Videos for November 2009: Django Development - Richard Kiss, Eddy Mulyono, Glen Jarvis, Simeon Franklin; BayPiggies Python for scientific research, discussion with Guido van Rossum; UCBSciPy Netbooks - Michael Gorven, Dave Mackie, and Jonathan Carter; CLUG Japan Linux Symposium Keynote, Linus Torvalds & Jim Zemlin; Linux Foundation http://sites.google.com/site/berkeleytip/talk-videos Download & watch them before the meetings, discuss at the meetings. Thanks to all the Speakers, Videographers, & Groups! :) [Record your local meeting! Put the video online, & email me for inclusion for next month. :) ] ===== Hot topics: Ubuntu 9.10 - Problems? Fixes? Upgrade? Install? Freeswitch VOIP server - setup for BTIP Flyers & outreach to UCBerkeley. Outreach to other UC campuses next semester. ===== Opportunities - Learn new, or increase your job skills, &/or volunteer & help the community: Set up any of: a BTIP Mailing List, web server/site, Freeswitch VOIP server, or Virtual Private Network & SSL ===== Announcement Flyers: Print & Post them in your community. 4/5 available - Freedom, Karmic Koala, Free Culture, SciPy, OLPC. See bottom of page: http://groups.google.com/group/BerkTIPGlobal ===== New BTIP Webpages @ http://sites.google.com/site/berkeleytip/ UC Campus local groups; Free Hardware; System Administration; Announcement Flyers; Opportunities For Forwarding - You are invited to forward this announcement wherever it would be appreciated. From mike at jerris.com Sat Nov 7 09:47:22 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 7 Nov 2009 12:47:22 -0500 Subject: [Freeswitch-users] Events in mod_perl In-Reply-To: <7d79b3930911062138l262db9bema29c620a7b8cef94@mail.gmail.com> References: <7d79b3930911052229k2828ff7dic6d5b887a4897c8c@mail.gmail.com> <7d79b3930911062138l262db9bema29c620a7b8cef94@mail.gmail.com> Message-ID: You can use EventConsumer class for this, I am afraid its not very documented, but I do recall either a sample or discussion on the mailing list that you should be able to find. Mike On Nov 7, 2009, at 12:38 AM, lakshmanan ganapathy wrote: > Ya. I have done that event processing with ESL. But I wanted to > know, whether in mod_perl, we can get the events and process it or > not. I've seen function's like events_get etc.. But I don't know how > to use those things. > > In mod_perl if I'm able to get the events, then it will be easier > for me. > Is it possible!!! > > 2009/11/6 Jo?o Mesquita > I don't know what you are trying to do exactly but I think that you > might need to you ESL instead. > > Why don't you take a look at all the examples inside ${SVNROOT}/libs/ > esl and see if that fits you? I have a hunch that it would. > > JM > > On Fri, Nov 6, 2009 at 4:29 AM, lakshmanan ganapathy > wrote: > Hi all, > Is there any way to receive events while running a perl program > with the help of mod_perl?? > > I've seen some functions related to sending and receiving events in > the mod_perl wiki. But I don't know how to use that. > Any help!!! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/9fec2f91/attachment-0002.html From mike at jerris.com Sat Nov 7 09:50:23 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 7 Nov 2009 12:50:23 -0500 Subject: [Freeswitch-users] leg_delay_start In-Reply-To: <3e6d7b0c0911070626g32af550fsd80e99d8266a7aa8@mail.gmail.com> References: <3e6d7b0c0911070626g32af550fsd80e99d8266a7aa8@mail.gmail.com> Message-ID: <057FECFA-306B-4710-93BA-6998AB15E5B8@jerris.com> Those vars were not even available in 1.0.3. I can't recall if they were in 1.0.4 or if you will need to use the latest 1.0.5 pre-release. Mike On Nov 7, 2009, at 9:26 AM, Steven Brown wrote: > Hi > > I've been trying to experiment with leg_delay_start when bridging to > two mobiles via a gateway, however regardless of settings both legs > are bridged immediately. I noticed a previous post on problems with > leg_delay_start which seemed to go unanswered, just wondered if > there is a known issue or if its something I'm doing wrong. > > Using FS 1.0.3 > > Dialplan extract as follows : > > > > Any pointers appreciated From mike at jerris.com Sat Nov 7 09:52:00 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 7 Nov 2009 12:52:00 -0500 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> Message-ID: <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> looks like ogg devel packages are installed but ogg lib is not? On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: > FreeSWITCH seems to be unable to read MP3 files, citing that it's an > unknown format. Looking through the log, I found this during startup: > > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error > Loading module /usr/local/freeswitch/mod/mod_shout.so > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: > ogg_sync_wrote** > > There don't seem to be any compile-time errors, yet I can't seem to > eliminate this issue. Any help would be appreciated. From msc at freeswitch.org Sat Nov 7 10:49:27 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 7 Nov 2009 10:49:27 -0800 Subject: [Freeswitch-users] Valid Dial Strings In-Reply-To: <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> Message-ID: <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" wrote: > Hi, > > can someone pointme to where the valid dialing strings are specified ? > For SIP dialstrings check here: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings Also, if you send us examples of what you've tried we can help you figure out what's wrong. > I'm assuming that something, somewhere, tells FS that numbers are > invalid > before they get dialed ? Pastebin some debug logs of what's happening. Check out this page which has lots of useful information on how to collect information: http://wiki.freeswitch.org/wiki/Reporting_Bugs It sounds like it's just a matter of figuring out how to configure your specific setup. Please report back with more information and we'll be happy to help. -MC From msc at freeswitch.org Sat Nov 7 11:02:45 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 7 Nov 2009 11:02:45 -0800 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: <4AF4AF73.8070804@tx.rr.com> References: <4AF4AF73.8070804@tx.rr.com> Message-ID: <21938E73-E566-431B-A0EA-7DE1731E1F8B@freeswitch.org> On Nov 6, 2009, at 3:21 PM, Orien Love wrote: > First of all, Thanks to the help I received on my pfSense > installation, > especially to Michael. I have a basic test system up and running. I > am > still waiting on some hardware but the base system is working!!!! > > I am looking on advice on how to set up a simple office PBX, 20 phones > and 4 outside lines.with 2 or 3 "operator" phones and the rest will be > extensions. > > Here is my plan, please let me know if it does not make sense, or if I > am going about it > > System Hardware > 4 spa3000's to handle the outside lines. > 2-3 polycom 601 phones with expansion modules (Operator phones) > 18 polycom 330 or other phones for desks. > 2-24 port cisco POE switches > 1 pfSense server. > > System Design. > > Extension Numbers 2xx > Outside line access 1xxxxxxxxxx > groups 3xx > auto-attendent ??? > > here are my questions > #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB > of > memory handle this proposed system? (Here is the MB I am thing of > using > MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) > #2 how do I pool my spa 3000 FXO lines so that the outgoing calls > use the first available line? also how do insure that metro (non long > distance) calls go to a specific line if available? > > I have learned a lot on how to set up Polycom 601 phones, I am > planning > on writing a how to document, is there any specific format? > > Thanks Orien > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From msc at freeswitch.org Sat Nov 7 11:07:55 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 7 Nov 2009 11:07:55 -0800 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: <4AF4AF73.8070804@tx.rr.com> References: <4AF4AF73.8070804@tx.rr.com> Message-ID: <92D8199B-5094-4170-9208-CD299BAF9D31@freeswitch.org> On Nov 6, 2009, at 3:21 PM, Orien Love wrote: > First of all, Thanks to the help I received on my pfSense > installation, > especially to Michael. I have a basic test system up and running. I > am > still waiting on some hardware but the base system is working!!!! > > I am looking on advice on how to set up a simple office PBX, 20 phones > and 4 outside lines.with 2 or 3 "operator" phones and the rest will be > extensions. > > Here is my plan, please let me know if it does not make sense, or if I > am going about it > > System Hardware > 4 spa3000's to handle the outside lines. > 2-3 polycom 601 phones with expansion modules (Operator phones) > 18 polycom 330 or other phones for desks. > 2-24 port cisco POE switches > 1 pfSense server. > > System Design. > > Extension Numbers 2xx > Outside line access 1xxxxxxxxxx > groups 3xx > auto-attendent ??? > > here are my questions > #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB > of > memory handle this proposed system? (Here is the MB I am thing of > using > MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) The FS devs don't endorse or recommend any specific hardware. However, many FreeSWITCH user are quite vocal about the hardware they prefer so we will let them speak. Just remember that YMMV depending on your specific setup. That being said, it does not sound like your HW reqs are very intense. Some of our members who've used the atom can probably give you specific feedback. > #2 how do I pool my spa 3000 FXO lines so that the outgoing calls > use the first available line? also how do insure that metro (non long > distance) calls go to a specific line if available? > > I have learned a lot on how to set up Polycom 601 phones, I am > planning > on writing a how to document, is there any specific format? > > Thanks Orien > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From msc at freeswitch.org Sat Nov 7 11:19:25 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 7 Nov 2009 11:19:25 -0800 Subject: [Freeswitch-users] suggestions for hardware. Message-ID: <194F9101-604D-46A9-9530-AD3C47CFC54C@freeswitch.org> On Nov 6, 2009, at 3:21 PM, Orien Love wrote: > First of all, Thanks to the help I received on my pfSense > installation, > especially to Michael. I have a basic test system up and running. I > am > still waiting on some hardware but the base system is working!!!! > > I am looking on advice on how to set up a simple office PBX, 20 phones > and 4 outside lines.with 2 or 3 "operator" phones and the rest will be > extensions. > > Here is my plan, please let me know if it does not make sense, or if I > am going about it > > System Hardware > 4 spa3000's to handle the outside lines. > 2-3 polycom 601 phones with expansion modules (Operator phones) > 18 polycom 330 or other phones for desks. > 2-24 port cisco POE switches > 1 pfSense server. > > System Design. > > Extension Numbers 2xx > Outside line access 1xxxxxxxxxx > groups 3xx > auto-attendent ??? > > here are my questions > #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of > memory handle this proposed system? (Here is the MB I am thing of > using > MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) > #2 how do I pool my spa 3000 FXO lines so that the outgoing calls > use the first available line? also how do insure that metro (non long > distance) calls go to a specific line if available? Sorry for the multiple posts. (Stinking iPhone without a slide-out keyboard.) This can be done with a little dialplan logic and using the pipe sep list of dialstrings. Check out the bridge examples here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall > > I have learned a lot on how to set up Polycom 601 phones, I am > planning > on writing a how to document, is there any specific format? > I looked around and I didn't see any other fully documented examples of hardware setup and config. My recommendation is to create a new wiki page and do your write-up. When you're done let me know and we will figure out how to index it. -MC > Thanks Orien > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From steve at justfone.com Sat Nov 7 11:59:02 2009 From: steve at justfone.com (Steven Brown) Date: Sat, 7 Nov 2009 19:59:02 +0000 Subject: [Freeswitch-users] leg_delay_start Message-ID: <3e6d7b0c0911071159q56faf627w710fcede6d6c031b@mail.gmail.com> Thanks Mike, I should have checked that, I've just done a make current on my other FS box and tested on it and can confirm that leg_delay_start works a treat, exactly as I need. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091107/9d896ea2/attachment-0002.html From stevendt at primrosebank.net Sat Nov 7 13:34:26 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Sat, 7 Nov 2009 21:34:26 -0000 Subject: [Freeswitch-users] Valid Dial Strings References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> Message-ID: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> Hi Michael, thanks for the reply. I think that I have got to the bottom of how to allow numbers to get to the VOIP gateway - at the moment, my dialplan just allows any. The big problem is that the VOIP Gateway (Linksys 3102) rejects any calls to it from VOIP to the PSTN and I don't know why. I have posted a dump to the pastebin, hopefully, the messages in there will allow someone to see what the problem is and give me some pointers on how I might fix it regards Dave ----- Original Message ----- From: "Michael S Collins" To: Sent: Saturday, November 07, 2009 6:49 PM Subject: Re: [Freeswitch-users] Valid Dial Strings > > On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" > wrote: > >> Hi, >> >> can someone pointme to where the valid dialing strings are specified ? >> > For SIP dialstrings check here: > http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings > > Also, if you send us examples of what you've tried we can help you > figure out what's wrong. > >> I'm assuming that something, somewhere, tells FS that numbers are >> invalid >> before they get dialed ? > > Pastebin some debug logs of what's happening. Check out this page > which has lots of useful information on how to collect information: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It sounds like it's just a matter of figuring out how to configure > your specific setup. Please report back with more information and > we'll be happy to help. > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Sat Nov 7 18:20:42 2009 From: dujinfang at gmail.com (Seven Du) Date: Sun, 8 Nov 2009 10:20:42 +0800 Subject: [Freeswitch-users] Announce FreeSWITCH-CN - the Chinese community Message-ID: <23f91030911071820w43f72b68lb81e1785f3ac780b@mail.gmail.com> ALL, FreeSWITCH-CN is a non-official, non-profit Chinese community. There was some arguments of language specified sites vs. a central site, freeswitch.org, on this list. However, facts are that people would like to find information in their native language and here are already some language specified community exists - it, ru es etc.. And I'm sure that given time we will get more and more people involve in around the world. In my opinion, I still perfer to host documents on the office wiki as long as it support multi-language translations which I think would be quick as Janitors already starting to re-organizing the wiki. And make http://www.freeswitch.org.cn just a landing page - search engines hit the page and we will guide users to the proper information. Also there's a google group: http://groups.google.com/group/freeswitch-cn?hl=en available. Thanks Anthony and his team's permission of us using the FreeSWITCH logo and the sexy domain name in a non-profit condition. Thanks all involved in this project to make the community stronger and keep it rolling. Join us and leave your advices to make it better. -7- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091108/f90e08fb/attachment-0002.html From lei.tlfly at gmail.com Sat Nov 7 19:36:09 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Sun, 8 Nov 2009 11:36:09 +0800 Subject: [Freeswitch-users] Announce FreeSWITCH-CN - the Chinese community In-Reply-To: <23f91030911071820w43f72b68lb81e1785f3ac780b@mail.gmail.com> References: <23f91030911071820w43f72b68lb81e1785f3ac780b@mail.gmail.com> Message-ID: <50c41b4e0911071936o3c963a2av9cdfb8fe66bfbd5@mail.gmail.com> Congratulations? 2009/11/8 Seven Du > ALL, > > FreeSWITCH-CN is a non-official, non-profit Chinese community. > > There was some arguments of language specified sites vs. a central site, > freeswitch.org, on this list. However, facts are that people would like to > find information in their native language and here are already some language > specified community exists - it, ru es etc.. And I'm sure that given time > we will get more and more people involve in around the world. > > In my opinion, I still perfer to host documents on the office wiki as long > as it support multi-language translations which I think would be quick as > Janitors already starting to re-organizing the wiki. And make > http://www.freeswitch.org.cn just a landing page - search engines hit the > page and we will guide users to the proper information. > > Also there's a google group: > http://groups.google.com/group/freeswitch-cn?hl=en available. > > Thanks Anthony and his team's permission of us using the FreeSWITCH logo > and the sexy domain name in a non-profit condition. Thanks all involved in > this project to make the community stronger and keep it rolling. > > Join us and leave your advices to make it better. > > -7- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091108/13078c6a/attachment-0002.html From yehavi.bourvine at gmail.com Sun Nov 8 00:46:13 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 8 Nov 2009 10:46:13 +0200 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information Message-ID: Hello, While trying to display the *called party *name on SNOM phones I've found that the field sent to the phone needs to be changed slightly in order to make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and Cisco work ok. Just wanted to let the developers know... And now a question: We have SNOM phones monitoring other extensions (BLF feature). When a call comes in, the monitoring phones get notification, but the name field (identity display) contains the calling extension number and not its display name. Can this be fixed? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091108/85f87fea/attachment-0002.html From list.subscription at alexrambau.com Sun Nov 8 02:42:38 2009 From: list.subscription at alexrambau.com (Alex Rambau) Date: Sun, 8 Nov 2009 03:42:38 -0700 Subject: [Freeswitch-users] Event Socket Timeout - Outbound Message-ID: Currently, is there any way to set the timeout on an outbound event socket? In case, for whatever reason, the socket application at 192.168.1.108:4444 is unresponsive or offline, I would like the call to not wait the extraordinary amount of time it takes to timeout so that I can handle it in other ways. My dial plan entry is as follows: Thanks in advance, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091108/e31a2912/attachment-0002.html From mcampbellsmith at gmail.com Sun Nov 8 03:59:32 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 8 Nov 2009 22:59:32 +1100 Subject: [Freeswitch-users] Extension: No audio Message-ID: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> Hi! I have FS natted and am connecting with an 'external' extension that is registered to FS. ie the extension 2000 is registered on the internet with a public IP through my router to FS (192.168.1.120 IP address). uPnP works and I see that the extension is registered successfully. The problem is that I do not get any audio When looking at the SIP trace, I see the INVITE but do not see a TRYING or RINGING message. The extension is actually ringing. I modified the RTP port range on the remote end to match the RTP ports of freeswitch. I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 If anyone has an idea what needs to be set to get audio, help appreciated. Thanks! From god.nirvana at gmail.com Sun Nov 8 08:34:09 2009 From: god.nirvana at gmail.com (god.nirvana) Date: Mon, 9 Nov 2009 00:34:09 +0800 Subject: [Freeswitch-users] javascript parameter Message-ID: <200911090034059066890@gmail.com> hi all: how can i get the value of the myArg1 myArg2 in test.js. like this originate sofia/example/1000 at somewhere.com '&javascript(test.js myArg1 myArg2)' thanks! 2009-11-09 god.nirvana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/e372f11e/attachment-0002.html From brian at freeswitch.org Sun Nov 8 09:21:18 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 8 Nov 2009 11:21:18 -0600 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information In-Reply-To: References: Message-ID: Can you elaborate on this and provide a patch on jira? /b On Nov 8, 2009, at 2:46 AM, Yehavi Bourvine wrote: > Hello, > > While trying to display the called party name on SNOM phones I've > found that the field sent to the phone needs to be changed slightly > in order to make SNOM work: Insetad of sending P-Assterted-Identity > SNOM expects Remote-Party-ID. I changed it in mod_sofia and now > SNOM, Polycom and Cisco work ok. Just wanted to let the developers > know... > > And now a question: We have SNOM phones monitoring other > extensions (BLF feature). When a call comes in, the monitoring > phones get notification, but the name field (identity display) > contains the calling extension number and not its display name. Can > this be fixed? > > Thanks! __Yehavi: From bruce at nani.ca Sun Nov 8 01:12:26 2009 From: bruce at nani.ca (Bruce Fletcher) Date: Sun, 8 Nov 2009 01:12:26 -0800 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 Message-ID: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> I'm trying to get through the noobie tutorial that c888 recommends in IRC, but PortAudio doesn't seem to build properly on Mac OS X 10.6. It failed due to some code that wasn't 64bit ready, apparently. The error I got was exactly the same as this 4 month old error from MacPorts: http://trac.macports.org/ticket/20338#comment:4 Someone there produced the following patch, which almost got me going again: http://trac.macports.org/changeset/53938 This has to be manually applied because the line numbers don't match up between the MacPorts and FreeSWITCH versions of PortAudio. At least, not in the configure script. This still doesn't quite get portaudio built, but I found a suggestion to configure it with -- disable-shared here: http://music.columbia.edu/pipermail/portaudio/2008-February/008188.html This finally got me to the point of having a compiled version of portaudio, but of course with that much hacking around nothing is going to go smoothly, is it. I can boot freeswitch fine and 'load mod_portaudio' seems to work, but when I try to hear some MOH output, I get this: freeswitch at Media-Centre.local> pa call 9999 2009-11-08 00:50:51.894512 [NOTICE] switch_channel.c:613 New Channel portaudio/9999 [3079018b-0df8-4c61-bffa-2f2eb681d06d] Assertion failed: (sizeof( UInt32 ) == sizeof( long )), function ringBufferIOProc, file src/hostapi/coreaudio/pa_mac_core.c, line 1713. I'd be happy to poke around on the PortAudio site and see if there is any useful information or patches there, but it's really late now so I just wanted to document where I've gotten to in case someone else has the same problem or a better clue on how to fix things. This is all using version 15396 out of subversion, if it matters. Thanks, - Bruce From brian at freeswitch.org Sun Nov 8 09:26:09 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 8 Nov 2009 11:26:09 -0600 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> Message-ID: <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> The problem is the patch isn't backwards compatible and blows away any chance of being so. We have looked at this... and that patch IS NOT RIGHT. /b On Nov 8, 2009, at 3:12 AM, Bruce Fletcher wrote: > I'm trying to get through the noobie tutorial that c888 recommends in > IRC, but PortAudio doesn't seem to build properly on Mac OS X 10.6. > It failed due to some code that wasn't 64bit ready, apparently. The > error I got was exactly the same as this 4 month old error from > MacPorts: > > http://trac.macports.org/ticket/20338#comment:4 > > Someone there produced the following patch, which almost got me going > again: > > http://trac.macports.org/changeset/53938 > > This has to be manually applied because the line numbers don't match > up between the MacPorts and FreeSWITCH versions of PortAudio. At > least, not in the configure script. This still doesn't quite get > portaudio built, but I found a suggestion to configure it with -- > disable-shared here: > > http://music.columbia.edu/pipermail/portaudio/2008-February/008188.html > > This finally got me to the point of having a compiled version of > portaudio, but of course with that much hacking around nothing is > going to go smoothly, is it. I can boot freeswitch fine and 'load > mod_portaudio' seems to work, but when I try to hear some MOH output, > I get this: > > freeswitch at Media-Centre.local> pa call 9999 > 2009-11-08 00:50:51.894512 [NOTICE] switch_channel.c:613 New Channel > portaudio/9999 [3079018b-0df8-4c61-bffa-2f2eb681d06d] > Assertion failed: (sizeof( UInt32 ) == sizeof( long )), function > ringBufferIOProc, file src/hostapi/coreaudio/pa_mac_core.c, line 1713. > > I'd be happy to poke around on the PortAudio site and see if there is > any useful information or patches there, but it's really late now so I > just wanted to document where I've gotten to in case someone else has > the same problem or a better clue on how to fix things. > > This is all using version 15396 out of subversion, if it matters. > > Thanks, > - Bruce > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From bruce at nani.ca Sun Nov 8 10:03:01 2009 From: bruce at nani.ca (Bruce Fletcher) Date: Sun, 8 Nov 2009 10:03:01 -0800 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> Message-ID: <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> OK, I'll ignore that MacPorts patch for now and try to find a better approach. I'll look into this further tonight, but this morning I found a more recent promising patch on the PortAudio site: http://www.portaudio.com/trac/changeset/1418 It seems to push some data types to 32 bit regardless of platform, which might work better than the MacPorts approach of migrating some data structures to 64 bit. At any rate, this patch being on the PortAudio site suggests it might be a more approved fix. I'll keep plugging at this in my free time and report any significant progress back to the list. Thanks, - Bruce On 2009-11-08, at 9:26 AM, Brian West wrote: > The problem is the patch isn't backwards compatible and blows away any > chance of being so. We have looked at this... and that patch IS NOT > RIGHT. > > /b > > On Nov 8, 2009, at 3:12 AM, Bruce Fletcher wrote: > >> I'm trying to get through the noobie tutorial that c888 recommends in >> IRC, but PortAudio doesn't seem to build properly on Mac OS X 10.6. >> It failed due to some code that wasn't 64bit ready, apparently. The >> error I got was exactly the same as this 4 month old error from >> MacPorts: >> >> http://trac.macports.org/ticket/20338#comment:4 >> >> Someone there produced the following patch, which almost got me going >> again: >> >> http://trac.macports.org/changeset/53938 >> >> This has to be manually applied because the line numbers don't match >> up between the MacPorts and FreeSWITCH versions of PortAudio. At >> least, not in the configure script. This still doesn't quite get >> portaudio built, but I found a suggestion to configure it with -- >> disable-shared here: >> >> http://music.columbia.edu/pipermail/portaudio/2008-February/008188.html >> >> This finally got me to the point of having a compiled version of >> portaudio, but of course with that much hacking around nothing is >> going to go smoothly, is it. I can boot freeswitch fine and 'load >> mod_portaudio' seems to work, but when I try to hear some MOH output, >> I get this: >> >> freeswitch at Media-Centre.local> pa call 9999 >> 2009-11-08 00:50:51.894512 [NOTICE] switch_channel.c:613 New Channel >> portaudio/9999 [3079018b-0df8-4c61-bffa-2f2eb681d06d] >> Assertion failed: (sizeof( UInt32 ) == sizeof( long )), function >> ringBufferIOProc, file src/hostapi/coreaudio/pa_mac_core.c, line >> 1713. >> >> I'd be happy to poke around on the PortAudio site and see if there is >> any useful information or patches there, but it's really late now >> so I >> just wanted to document where I've gotten to in case someone else has >> the same problem or a better clue on how to fix things. >> >> This is all using version 15396 out of subversion, if it matters. >> >> Thanks, >> - Bruce >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From frank at carmickle.com Sun Nov 8 11:22:42 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sun, 8 Nov 2009 14:22:42 -0500 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> Message-ID: <20091108192242.GA10757@base.carmickle.com> Hello I am also having trouble with portaudio. I still haven't figured out what it is that is wrong. I have had it working in the past on this same machine same install of debian lenny. Now it just reports [ERR] mod_portaudio.c:964 Cannot find an input device Also seeing this on a fedor 12 64 bit machine but not a sid 64 bit machine or a fedora 11 32 bit machine. I'm not sure if it's the same problem but please do keep me informed about what you find. Thanks --Frank From mike at jerris.com Sun Nov 8 12:25:28 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 8 Nov 2009 15:25:28 -0500 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> Message-ID: <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> If you can figure out a clean way for us to do this with proper ifdefs in tree in a way that will not break others that would be the most preferred. Mike On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote: > OK, I'll ignore that MacPorts patch for now and try to find a better > approach. > > I'll look into this further tonight, but this morning I found a more > recent promising patch on the PortAudio site: > > http://www.portaudio.com/trac/changeset/1418 > > It seems to push some data types to 32 bit regardless of platform, > which might work better than the MacPorts approach of migrating some > data structures to 64 bit. At any rate, this patch being on the > PortAudio site suggests it might be a more approved fix. > > I'll keep plugging at this in my free time and report any significant > progress back to the list. > > Thanks, > - Bruce > From mike at jerris.com Sun Nov 8 12:28:21 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 8 Nov 2009 15:28:21 -0500 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> Message-ID: You don't have ext-rtp-ip set in your config. Mike On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: > Hi! > > I have FS natted and am connecting with an 'external' extension that > is registered to FS. ie the extension 2000 is registered on the > internet with a public IP through my router to FS (192.168.1.120 IP > address). uPnP works and I see that the extension is registered > successfully. > > The problem is that I do not get any audio > > When looking at the SIP trace, I see the INVITE but do not see a > TRYING or RINGING message. The extension is actually ringing. I > modified the RTP port range on the remote end to match the RTP ports > of freeswitch. > > I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 > > If anyone has an idea what needs to be set to get audio, help > appreciated. > > Thanks! From mcampbellsmith at gmail.com Sun Nov 8 13:59:44 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 9 Nov 2009 08:59:44 +1100 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> Message-ID: <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> Hi Mike, I should have put that in also. I do have external_rtp_ip set in my config. I have it set to my domain name: I should also mention that if I use flaphone.com (which registers with an external IP address), then I get audio. In sofia, I see my IP addresses: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.1.120 Ext-RTP-IP 124.xxx.xxx.xxx SIP-IP 192.168.1.120 Ext-SIP-IP 124.xxx.xxx.x URL sip:mod_sofia at 192.168.1.120:5060 BIND-URL sip:mod_sofia at 192.168.1.120:5060 HOLD-MUSIC silence OUTBOUND-PROXY N/A CODECS G726-32,G722,PCMU,PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris wrote: > You don't have ext-rtp-ip set in your config. > > Mike > > On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: > >> Hi! >> >> I have FS natted and am connecting with an 'external' extension that >> is registered to FS. ?ie the extension 2000 is registered on the >> internet with a public IP through my router to FS (192.168.1.120 IP >> address). ?uPnP works and I see that the extension is registered >> successfully. >> >> The problem is that I do not get any audio >> >> When looking at the SIP trace, I see the INVITE but do not see a >> TRYING or RINGING message. ?The extension is actually ringing. ?I >> modified the RTP port range on the remote end to match the RTP ports >> of freeswitch. >> >> I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 >> >> If anyone has an idea what needs to be set to get audio, help >> appreciated. >> >> Thanks! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Sun Nov 8 16:41:42 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 8 Nov 2009 19:41:42 -0500 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> Message-ID: <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> Your packet traces would disagree with the statements below. It is sending your internal address in rtp, so its not set correctly on whatever profile your using to call out, MIke On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: > Hi Mike, > > I should have put that in also. > > I do have external_rtp_ip set in my config. I have it set to my > domain name: > > > I should also mention that if I use flaphone.com (which registers with > an external IP address), then I get audio. In sofia, I see my IP > addresses: > > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > Name internal > Domain Name N/A > DBName sofia_reg_internal > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_from > RTP-IP 192.168.1.120 > Ext-RTP-IP 124.xxx.xxx.xxx > SIP-IP 192.168.1.120 > Ext-SIP-IP 124.xxx.xxx.x > URL sip:mod_sofia at 192.168.1.120:5060 > BIND-URL sip:mod_sofia at 192.168.1.120:5060 > HOLD-MUSIC silence > OUTBOUND-PROXY N/A > CODECS G726-32,G722,PCMU,PCMA > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT true > STUN-ENABLED true > STUN-AUTO-DISABLE false > > On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris > wrote: >> You don't have ext-rtp-ip set in your config. >> >> Mike >> >> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: >> >>> Hi! >>> >>> I have FS natted and am connecting with an 'external' extension that >>> is registered to FS. ie the extension 2000 is registered on the >>> internet with a public IP through my router to FS (192.168.1.120 IP >>> address). uPnP works and I see that the extension is registered >>> successfully. >>> >>> The problem is that I do not get any audio >>> >>> When looking at the SIP trace, I see the INVITE but do not see a >>> TRYING or RINGING message. The extension is actually ringing. I >>> modified the RTP port range on the remote end to match the RTP ports >>> of freeswitch. >>> >>> I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 >>> >>> If anyone has an idea what needs to be set to get audio, help >>> appreciated. >>> >>> Thanks! >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mcampbellsmith at gmail.com Sun Nov 8 18:14:48 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 9 Nov 2009 13:14:48 +1100 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> Message-ID: <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> OK.. thanks Mike. I assume I am using the Internal profile. I have defined user 2000 in the 'directory' using a context called family: switch_ivr.c:1367 Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family] This is an extract from sofia: sofia status profile internal ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.1.120 Ext-RTP-IP 124.xxx.xxx.xxx SIP-IP 192.168.1.120 Ext-SIP-IP 124.xxx.xxx.xxx URL sip:mod_sofia at 192.168.1.120:5060 BIND-URL sip:mod_sofia at 192.168.1.120:5060 HOLD-MUSIC silence OUTBOUND-PROXY N/A CODECS G726-32,G722,PCMU,PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 100 FAILED-CALLS-IN 25 CALLS-OUT 38 FAILED-CALLS-OUT 31 Registrations: ================================================================================================= Call-ID: 68534BBA9B461526 at 58.169.138.53 User: 2000 at 192.168.1.120 Contact: "user" Agent: dunno Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) Host: freeswitch IP: 58.xxx.xxx.xxx Port: 5060 Auth-User: 2000 Auth-Realm: markcs.dyndns.org MWI-Account: 2000 at 192.168.1.120 The internal.xml file has a lot in it, but I guess these are the important things for this profile: I will try to change auto-nat to be $${external_sip_ip} One question though: Any idea why I never see the TRYING or RINGING messages? Are tehse related to the RTP IP address or not? Without these I assume something is incorrect and I do not hear ringback.... Thanks! On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris wrote: > Your packet traces would disagree with the statements below. ?It is > sending your internal address in rtp, so its not set correctly on > whatever profile your using to call out, > > MIke > > On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: > >> Hi Mike, >> >> I should have put that in also. >> >> I do have external_rtp_ip set in my config. ?I have it set to my >> domain name: >> >> >> I should also mention that if I use flaphone.com (which registers with >> an external IP address), then I get audio. ?In sofia, I see my IP >> addresses: >> >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> ====================================================================== >> Name ? ? ? ? ? ? ? ? ? ?internal >> Domain Name ? ? ? ? ? ? N/A >> DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >> Pres Hosts >> Dialplan ? ? ? ? ? ? ? ?XML >> Context ? ? ? ? ? ? ? ? public >> Challenge Realm ? ? ? ? auto_from >> RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >> SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.x >> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >> HOLD-MUSIC ? ? ? ? ? ? ?silence >> OUTBOUND-PROXY ? ? ? ? ?N/A >> CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >> TEL-EVENT ? ? ? ? ? ? ? 101 >> DTMF-MODE ? ? ? ? ? ? ? rfc2833 >> CNG ? ? ? ? ? ? ? ? ? ? 13 >> SESSION-TO ? ? ? ? ? ? ?0 >> MAX-DIALOG ? ? ? ? ? ? ?0 >> NOMEDIA ? ? ? ? ? ? ? ? false >> LATE-NEG ? ? ? ? ? ? ? ?false >> PROXY-MEDIA ? ? ? ? ? ? false >> AGGRESSIVENAT ? ? ? ? ? true >> STUN-ENABLED ? ? ? ? ? ?true >> STUN-AUTO-DISABLE ? ? ? false >> >> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris >> wrote: >>> You don't have ext-rtp-ip set in your config. >>> >>> Mike >>> >>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: >>> >>>> Hi! >>>> >>>> I have FS natted and am connecting with an 'external' extension that >>>> is registered to FS. ?ie the extension 2000 is registered on the >>>> internet with a public IP through my router to FS (192.168.1.120 IP >>>> address). ?uPnP works and I see that the extension is registered >>>> successfully. >>>> >>>> The problem is that I do not get any audio >>>> >>>> When looking at the SIP trace, I see the INVITE but do not see a >>>> TRYING or RINGING message. ?The extension is actually ringing. ?I >>>> modified the RTP port range on the remote end to match the RTP ports >>>> of freeswitch. >>>> >>>> I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 >>>> >>>> If anyone has an idea what needs to be set to get audio, help >>>> appreciated. >>>> >>>> Thanks! >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sprice at gmail.com Sun Nov 8 18:26:20 2009 From: sprice at gmail.com (SP) Date: Sun, 8 Nov 2009 20:26:20 -0600 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information In-Reply-To: References: Message-ID: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> before playing with mod_sofia, did you try the sip_cid_type variable? http://wiki.freeswitch.org/wiki/Variable_sip_cid_type On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine wrote: > Hello, > > ??While?trying to display the called party name ?on SNOM phones I've found > that the field sent to the phone needs to be changed slightly in order to > make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects > Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and Cisco > work ok. Just wanted to let the developers know... > > ? And now a question: We have SNOM phones monitoring other extensions (BLF > feature). When a call comes in, the monitoring phones get notification, but > the name field (identity display) contains the calling extension number and > not its display name. Can this be fixed? > > ??????????????????????????????? Thanks! __Yehavi: > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Shannon From mcampbellsmith at gmail.com Sun Nov 8 18:32:04 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 9 Nov 2009 13:32:04 +1100 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> Message-ID: <33c87fa30911081832qdd7357ncfe016f10ba87160@mail.gmail.com> Hi again, Actually, changing the to means that I now see the IP address in the INVITE message: v=0 o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx s=FreeSWITCH c=IN IP4 124.xxx.xxx.xxx t=0 0 m=audio 21234 RTP/AVP 0 2 9 8 101 13 Why would this be? I thought auto-nat was meant to solve these issues? However, I still do not see the TRYING or RINGING messages.... ideas appreciated. Thanks! On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith wrote: > OK.. thanks Mike. > > I assume I am using the Internal profile. ? I have defined user 2000 > in the 'directory' using a context called family: ? switch_ivr.c:1367 > Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family] > > This is an extract from sofia: > > sofia status profile internal > ================================================================================================= > Name ? ? ? ? ? ? ? ? ? ?internal > Domain Name ? ? ? ? ? ? N/A > DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal > Pres Hosts > Dialplan ? ? ? ? ? ? ? ?XML > Context ? ? ? ? ? ? ? ? public > Challenge Realm ? ? ? ? auto_from > RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 > Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx > SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 > Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx > URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 > BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 > HOLD-MUSIC ? ? ? ? ? ? ?silence > OUTBOUND-PROXY ? ? ? ? ?N/A > CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA > TEL-EVENT ? ? ? ? ? ? ? 101 > DTMF-MODE ? ? ? ? ? ? ? rfc2833 > CNG ? ? ? ? ? ? ? ? ? ? 13 > SESSION-TO ? ? ? ? ? ? ?0 > MAX-DIALOG ? ? ? ? ? ? ?0 > NOMEDIA ? ? ? ? ? ? ? ? false > LATE-NEG ? ? ? ? ? ? ? ?false > PROXY-MEDIA ? ? ? ? ? ? false > AGGRESSIVENAT ? ? ? ? ? true > STUN-ENABLED ? ? ? ? ? ?true > STUN-AUTO-DISABLE ? ? ? false > CALLS-IN ? ? ? ? ? ? ? ?100 > FAILED-CALLS-IN ? ? ? ? 25 > CALLS-OUT ? ? ? ? ? ? ? 38 > FAILED-CALLS-OUT ? ? ? ?31 > > Registrations: > ================================================================================================= > Call-ID: ? ? ? ?68534BBA9B461526 at 58.169.138.53 > User: ? ? ? ? ? 2000 at 192.168.1.120 > Contact: ? ? ? ?"user" > Agent: ? ? ? ? ?dunno > Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) > Host: ? ? ? ? ? freeswitch > IP: ? ? ? ? ? ? 58.xxx.xxx.xxx > Port: ? ? ? ? ? 5060 > Auth-User: ? ? ?2000 > Auth-Realm: ? ? markcs.dyndns.org > MWI-Account: ? ?2000 at 192.168.1.120 > > The internal.xml file has a lot in it, but I guess these are the > important things for this profile: > > ? ? > ? ? > > ? ? > ? ? > > I will try to change auto-nat to be $${external_sip_ip} > > One question though: ?Any idea why I never see the TRYING or RINGING > messages? ? Are tehse related to the RTP IP address or not? ?Without > these I assume something is incorrect and I do not hear ringback.... > > Thanks! > > On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris wrote: >> Your packet traces would disagree with the statements below. ?It is >> sending your internal address in rtp, so its not set correctly on >> whatever profile your using to call out, >> >> MIke >> >> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: >> >>> Hi Mike, >>> >>> I should have put that in also. >>> >>> I do have external_rtp_ip set in my config. ?I have it set to my >>> domain name: >>> >>> >>> I should also mention that if I use flaphone.com (which registers with >>> an external IP address), then I get audio. ?In sofia, I see my IP >>> addresses: >>> >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> = >>> ====================================================================== >>> Name ? ? ? ? ? ? ? ? ? ?internal >>> Domain Name ? ? ? ? ? ? N/A >>> DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >>> Pres Hosts >>> Dialplan ? ? ? ? ? ? ? ?XML >>> Context ? ? ? ? ? ? ? ? public >>> Challenge Realm ? ? ? ? auto_from >>> RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >>> SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.x >>> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >>> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >>> HOLD-MUSIC ? ? ? ? ? ? ?silence >>> OUTBOUND-PROXY ? ? ? ? ?N/A >>> CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >>> TEL-EVENT ? ? ? ? ? ? ? 101 >>> DTMF-MODE ? ? ? ? ? ? ? rfc2833 >>> CNG ? ? ? ? ? ? ? ? ? ? 13 >>> SESSION-TO ? ? ? ? ? ? ?0 >>> MAX-DIALOG ? ? ? ? ? ? ?0 >>> NOMEDIA ? ? ? ? ? ? ? ? false >>> LATE-NEG ? ? ? ? ? ? ? ?false >>> PROXY-MEDIA ? ? ? ? ? ? false >>> AGGRESSIVENAT ? ? ? ? ? true >>> STUN-ENABLED ? ? ? ? ? ?true >>> STUN-AUTO-DISABLE ? ? ? false >>> >>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris >>> wrote: >>>> You don't have ext-rtp-ip set in your config. >>>> >>>> Mike >>>> >>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: >>>> >>>>> Hi! >>>>> >>>>> I have FS natted and am connecting with an 'external' extension that >>>>> is registered to FS. ?ie the extension 2000 is registered on the >>>>> internet with a public IP through my router to FS (192.168.1.120 IP >>>>> address). ?uPnP works and I see that the extension is registered >>>>> successfully. >>>>> >>>>> The problem is that I do not get any audio >>>>> >>>>> When looking at the SIP trace, I see the INVITE but do not see a >>>>> TRYING or RINGING message. ?The extension is actually ringing. ?I >>>>> modified the RTP port range on the remote end to match the RTP ports >>>>> of freeswitch. >>>>> >>>>> I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 >>>>> >>>>> If anyone has an idea what needs to be set to get audio, help >>>>> appreciated. >>>>> >>>>> Thanks! >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>>> users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From jmesquita at freeswitch.org Sun Nov 8 18:39:21 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 9 Nov 2009 00:39:21 -0200 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <33c87fa30911081832qdd7357ncfe016f10ba87160@mail.gmail.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> <33c87fa30911081832qdd7357ncfe016f10ba87160@mail.gmail.com> Message-ID: It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise, no go.... Have you changed the ext-sip-ip too? Regards, JM On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi again, > > Actually, changing the to > means that I > now see the IP address in the INVITE message: > > v=0 > o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx > s=FreeSWITCH > c=IN IP4 124.xxx.xxx.xxx > t=0 0 > m=audio 21234 RTP/AVP 0 2 9 8 101 13 > > Why would this be? I thought auto-nat was meant to solve these issues? > > However, I still do not see the TRYING or RINGING messages.... ideas > appreciated. > > Thanks! > > On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith > wrote: > > OK.. thanks Mike. > > > > I assume I am using the Internal profile. I have defined user 2000 > > in the 'directory' using a context called family: switch_ivr.c:1367 > > Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family] > > > > This is an extract from sofia: > > > > sofia status profile internal > > > ================================================================================================= > > Name internal > > Domain Name N/A > > DBName sofia_reg_internal > > Pres Hosts > > Dialplan XML > > Context public > > Challenge Realm auto_from > > RTP-IP 192.168.1.120 > > Ext-RTP-IP 124.xxx.xxx.xxx > > SIP-IP 192.168.1.120 > > Ext-SIP-IP 124.xxx.xxx.xxx > > URL sip:mod_sofia at 192.168.1.120:5060 > > BIND-URL sip:mod_sofia at 192.168.1.120:5060 > > HOLD-MUSIC silence > > OUTBOUND-PROXY N/A > > CODECS G726-32,G722,PCMU,PCMA > > TEL-EVENT 101 > > DTMF-MODE rfc2833 > > CNG 13 > > SESSION-TO 0 > > MAX-DIALOG 0 > > NOMEDIA false > > LATE-NEG false > > PROXY-MEDIA false > > AGGRESSIVENAT true > > STUN-ENABLED true > > STUN-AUTO-DISABLE false > > CALLS-IN 100 > > FAILED-CALLS-IN 25 > > CALLS-OUT 38 > > FAILED-CALLS-OUT 31 > > > > Registrations: > > > ================================================================================================= > > Call-ID: 68534BBA9B461526 at 58.169.138.53 > > User: 2000 at 192.168.1.120 > > Contact: "user" > > Agent: dunno > > Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) > > Host: freeswitch > > IP: 58.xxx.xxx.xxx > > Port: 5060 > > Auth-User: 2000 > > Auth-Realm: markcs.dyndns.org > > MWI-Account: 2000 at 192.168.1.120 > > > > The internal.xml file has a lot in it, but I guess these are the > > important things for this profile: > > > > > > > > > > > > > > > > I will try to change auto-nat to be $${external_sip_ip} > > > > One question though: Any idea why I never see the TRYING or RINGING > > messages? Are tehse related to the RTP IP address or not? Without > > these I assume something is incorrect and I do not hear ringback.... > > > > Thanks! > > > > On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris wrote: > >> Your packet traces would disagree with the statements below. It is > >> sending your internal address in rtp, so its not set correctly on > >> whatever profile your using to call out, > >> > >> MIke > >> > >> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: > >> > >>> Hi Mike, > >>> > >>> I should have put that in also. > >>> > >>> I do have external_rtp_ip set in my config. I have it set to my > >>> domain name: > >>> > >>> > >>> I should also mention that if I use flaphone.com (which registers with > >>> an external IP address), then I get audio. In sofia, I see my IP > >>> addresses: > >>> > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> ====================================================================== > >>> Name internal > >>> Domain Name N/A > >>> DBName sofia_reg_internal > >>> Pres Hosts > >>> Dialplan XML > >>> Context public > >>> Challenge Realm auto_from > >>> RTP-IP 192.168.1.120 > >>> Ext-RTP-IP 124.xxx.xxx.xxx > >>> SIP-IP 192.168.1.120 > >>> Ext-SIP-IP 124.xxx.xxx.x > >>> URL sip:mod_sofia at 192.168.1.120:5060 > >>> BIND-URL sip:mod_sofia at 192.168.1.120:5060 > >>> HOLD-MUSIC silence > >>> OUTBOUND-PROXY N/A > >>> CODECS G726-32,G722,PCMU,PCMA > >>> TEL-EVENT 101 > >>> DTMF-MODE rfc2833 > >>> CNG 13 > >>> SESSION-TO 0 > >>> MAX-DIALOG 0 > >>> NOMEDIA false > >>> LATE-NEG false > >>> PROXY-MEDIA false > >>> AGGRESSIVENAT true > >>> STUN-ENABLED true > >>> STUN-AUTO-DISABLE false > >>> > >>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris > >>> wrote: > >>>> You don't have ext-rtp-ip set in your config. > >>>> > >>>> Mike > >>>> > >>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: > >>>> > >>>>> Hi! > >>>>> > >>>>> I have FS natted and am connecting with an 'external' extension that > >>>>> is registered to FS. ie the extension 2000 is registered on the > >>>>> internet with a public IP through my router to FS (192.168.1.120 IP > >>>>> address). uPnP works and I see that the extension is registered > >>>>> successfully. > >>>>> > >>>>> The problem is that I do not get any audio > >>>>> > >>>>> When looking at the SIP trace, I see the INVITE but do not see a > >>>>> TRYING or RINGING message. The extension is actually ringing. I > >>>>> modified the RTP port range on the remote end to match the RTP ports > >>>>> of freeswitch. > >>>>> > >>>>> I have put a sip trace in the pastebin at > http://pastebin.freeswitch.org/11035 > >>>>> > >>>>> If anyone has an idea what needs to be set to get audio, help > >>>>> appreciated. > >>>>> > >>>>> Thanks! > >>>> > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>>> users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/fe572a26/attachment-0002.html From yehavi.bourvine at gmail.com Sun Nov 8 18:57:02 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 9 Nov 2009 04:57:02 +0200 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information In-Reply-To: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> References: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> Message-ID: I was not aware of this variable; I will take a look on it tomorrow. However, when looking in the code I did not find something which looks like "Remote-Party-ID'". Thanks! __Yehavi: 2009/11/9 SP > before playing with mod_sofia, did you try the sip_cid_type variable? > > http://wiki.freeswitch.org/wiki/Variable_sip_cid_type > > On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine > wrote: > > Hello, > > > > While trying to display the called party name on SNOM phones I've > found > > that the field sent to the phone needs to be changed slightly in order to > > make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects > > Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and > Cisco > > work ok. Just wanted to let the developers know... > > > > And now a question: We have SNOM phones monitoring other extensions > (BLF > > feature). When a call comes in, the monitoring phones get notification, > but > > the name field (identity display) contains the calling extension number > and > > not its display name. Can this be fixed? > > > > Thanks! __Yehavi: > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Shannon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/530e1fab/attachment-0002.html From mcampbellsmith at gmail.com Sun Nov 8 19:40:17 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 9 Nov 2009 14:40:17 +1100 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> <33c87fa30911081832qdd7357ncfe016f10ba87160@mail.gmail.com> Message-ID: <33c87fa30911081940o4381f35dy6369b66384dbf90f@mail.gmail.com> Is there a way to determine if FS has detected nat? I am behind UPnP and I can see on the router the mappings for Freeswitch. 2009/11/9 Jo?o Mesquita : > It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise, > no go.... > > Have you changed the ext-sip-ip too? > > Regards, > > JM > > > On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith > wrote: >> >> Hi again, >> >> Actually, changing the to >> means that I >> now see the IP address in the INVITE message: >> >> ? v=0 >> ? o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx >> ? s=FreeSWITCH >> ? c=IN IP4 124.xxx.xxx.xxx >> ? t=0 0 >> ? m=audio 21234 RTP/AVP 0 2 9 8 101 13 >> >> Why would this be? ?I thought auto-nat was meant to solve these issues? >> >> However, I still do not see the TRYING or RINGING messages.... ?ideas >> appreciated. >> >> Thanks! >> >> On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith >> wrote: >> > OK.. thanks Mike. >> > >> > I assume I am using the Internal profile. ? I have defined user 2000 >> > in the 'directory' using a context called family: ? switch_ivr.c:1367 >> > Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family] >> > >> > This is an extract from sofia: >> > >> > sofia status profile internal >> > >> > ================================================================================================= >> > Name ? ? ? ? ? ? ? ? ? ?internal >> > Domain Name ? ? ? ? ? ? N/A >> > DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >> > Pres Hosts >> > Dialplan ? ? ? ? ? ? ? ?XML >> > Context ? ? ? ? ? ? ? ? public >> > Challenge Realm ? ? ? ? auto_from >> > RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> > Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >> > SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> > Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >> > URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >> > BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >> > HOLD-MUSIC ? ? ? ? ? ? ?silence >> > OUTBOUND-PROXY ? ? ? ? ?N/A >> > CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >> > TEL-EVENT ? ? ? ? ? ? ? 101 >> > DTMF-MODE ? ? ? ? ? ? ? rfc2833 >> > CNG ? ? ? ? ? ? ? ? ? ? 13 >> > SESSION-TO ? ? ? ? ? ? ?0 >> > MAX-DIALOG ? ? ? ? ? ? ?0 >> > NOMEDIA ? ? ? ? ? ? ? ? false >> > LATE-NEG ? ? ? ? ? ? ? ?false >> > PROXY-MEDIA ? ? ? ? ? ? false >> > AGGRESSIVENAT ? ? ? ? ? true >> > STUN-ENABLED ? ? ? ? ? ?true >> > STUN-AUTO-DISABLE ? ? ? false >> > CALLS-IN ? ? ? ? ? ? ? ?100 >> > FAILED-CALLS-IN ? ? ? ? 25 >> > CALLS-OUT ? ? ? ? ? ? ? 38 >> > FAILED-CALLS-OUT ? ? ? ?31 >> > >> > Registrations: >> > >> > ================================================================================================= >> > Call-ID: ? ? ? ?68534BBA9B461526 at 58.169.138.53 >> > User: ? ? ? ? ? 2000 at 192.168.1.120 >> > Contact: ? ? ? ?"user" >> > Agent: ? ? ? ? ?dunno >> > Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) >> > Host: ? ? ? ? ? freeswitch >> > IP: ? ? ? ? ? ? 58.xxx.xxx.xxx >> > Port: ? ? ? ? ? 5060 >> > Auth-User: ? ? ?2000 >> > Auth-Realm: ? ? markcs.dyndns.org >> > MWI-Account: ? ?2000 at 192.168.1.120 >> > >> > The internal.xml file has a lot in it, but I guess these are the >> > important things for this profile: >> > >> > ? ? >> > ? ? >> > >> > ? ? >> > ? ? >> > >> > I will try to change auto-nat to be $${external_sip_ip} >> > >> > One question though: ?Any idea why I never see the TRYING or RINGING >> > messages? ? Are tehse related to the RTP IP address or not? ?Without >> > these I assume something is incorrect and I do not hear ringback.... >> > >> > Thanks! >> > >> > On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris wrote: >> >> Your packet traces would disagree with the statements below. ?It is >> >> sending your internal address in rtp, so its not set correctly on >> >> whatever profile your using to call out, >> >> >> >> MIke >> >> >> >> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: >> >> >> >>> Hi Mike, >> >>> >> >>> I should have put that in also. >> >>> >> >>> I do have external_rtp_ip set in my config. ?I have it set to my >> >>> domain name: >> >>> >> >>> >> >>> I should also mention that if I use flaphone.com (which registers with >> >>> an external IP address), then I get audio. ?In sofia, I see my IP >> >>> addresses: >> >>> >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> = >> >>> ====================================================================== >> >>> Name ? ? ? ? ? ? ? ? ? ?internal >> >>> Domain Name ? ? ? ? ? ? N/A >> >>> DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >> >>> Pres Hosts >> >>> Dialplan ? ? ? ? ? ? ? ?XML >> >>> Context ? ? ? ? ? ? ? ? public >> >>> Challenge Realm ? ? ? ? auto_from >> >>> RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> >>> Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >> >>> SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >> >>> Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.x >> >>> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >> >>> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >> >>> HOLD-MUSIC ? ? ? ? ? ? ?silence >> >>> OUTBOUND-PROXY ? ? ? ? ?N/A >> >>> CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >> >>> TEL-EVENT ? ? ? ? ? ? ? 101 >> >>> DTMF-MODE ? ? ? ? ? ? ? rfc2833 >> >>> CNG ? ? ? ? ? ? ? ? ? ? 13 >> >>> SESSION-TO ? ? ? ? ? ? ?0 >> >>> MAX-DIALOG ? ? ? ? ? ? ?0 >> >>> NOMEDIA ? ? ? ? ? ? ? ? false >> >>> LATE-NEG ? ? ? ? ? ? ? ?false >> >>> PROXY-MEDIA ? ? ? ? ? ? false >> >>> AGGRESSIVENAT ? ? ? ? ? true >> >>> STUN-ENABLED ? ? ? ? ? ?true >> >>> STUN-AUTO-DISABLE ? ? ? false >> >>> >> >>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris >> >>> wrote: >> >>>> You don't have ext-rtp-ip set in your config. >> >>>> >> >>>> Mike >> >>>> >> >>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: >> >>>> >> >>>>> Hi! >> >>>>> >> >>>>> I have FS natted and am connecting with an 'external' extension that >> >>>>> is registered to FS. ?ie the extension 2000 is registered on the >> >>>>> internet with a public IP through my router to FS (192.168.1.120 IP >> >>>>> address). ?uPnP works and I see that the extension is registered >> >>>>> successfully. >> >>>>> >> >>>>> The problem is that I do not get any audio >> >>>>> >> >>>>> When looking at the SIP trace, I see the INVITE but do not see a >> >>>>> TRYING or RINGING message. ?The extension is actually ringing. ?I >> >>>>> modified the RTP port range on the remote end to match the RTP ports >> >>>>> of freeswitch. >> >>>>> >> >>>>> I have put a sip trace in the pastebin at >> >>>>> http://pastebin.freeswitch.org/11035 >> >>>>> >> >>>>> If anyone has an idea what needs to be set to get audio, help >> >>>>> appreciated. >> >>>>> >> >>>>> Thanks! >> >>>> >> >>>> >> >>>> _______________________________________________ >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>>> users >> >>>> http://www.freeswitch.org >> >>>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>> users >> >>> http://www.freeswitch.org >> >> >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mcampbellsmith at gmail.com Sun Nov 8 20:18:14 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 9 Nov 2009 15:18:14 +1100 Subject: [Freeswitch-users] Extension: No audio In-Reply-To: <33c87fa30911081940o4381f35dy6369b66384dbf90f@mail.gmail.com> References: <33c87fa30911080359n301af420o8c4c7abd291c9aa9@mail.gmail.com> <33c87fa30911081359p1f05072bw9895ed5aa3c5defe@mail.gmail.com> <7BC1FB98-87FB-4618-98E5-0145F8F637C5@jerris.com> <33c87fa30911081814r52d4d738q9a9e97c1ee4b2db9@mail.gmail.com> <33c87fa30911081832qdd7357ncfe016f10ba87160@mail.gmail.com> <33c87fa30911081940o4381f35dy6369b66384dbf90f@mail.gmail.com> Message-ID: <33c87fa30911082018r75bb2d9cvf9359846c4fb281f@mail.gmail.com> I think I've fixed it, but I had to change a few things... I had a host name set in vars.xml for external_rtp_ip and for external_sip_ip. Having the external_rtp_ip set to a hostname, sofia showed the RTP-IP 192.168.1.120 Ext-RTP-IP host:myhostname SIP-IP 192.168.1.120 Ext-SIP-IP 124.190.249.9 I think this caused some problems. Once this was changed back to stun, I now get RINGING messages and I get audio. I still have ext-rtp-ip and ext-sip-ip set to auto-nat in internal.xml. Could this be the cause or is there something else that caused this issue? I am using FreeSWITCH Version 1.0.trunk (15126) On Mon, Nov 9, 2009 at 2:40 PM, Mark Campbell-Smith wrote: > Is there a way to determine if FS has detected nat? ?I am behind UPnP > and I can see on the router the mappings for Freeswitch. > > 2009/11/9 Jo?o Mesquita : >> It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise, >> no go.... >> >> Have you changed the ext-sip-ip too? >> >> Regards, >> >> JM >> >> >> On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith >> wrote: >>> >>> Hi again, >>> >>> Actually, changing the to >>> means that I >>> now see the IP address in the INVITE message: >>> >>> ? v=0 >>> ? o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx >>> ? s=FreeSWITCH >>> ? c=IN IP4 124.xxx.xxx.xxx >>> ? t=0 0 >>> ? m=audio 21234 RTP/AVP 0 2 9 8 101 13 >>> >>> Why would this be? ?I thought auto-nat was meant to solve these issues? >>> >>> However, I still do not see the TRYING or RINGING messages.... ?ideas >>> appreciated. >>> >>> Thanks! >>> >>> On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith >>> wrote: >>> > OK.. thanks Mike. >>> > >>> > I assume I am using the Internal profile. ? I have defined user 2000 >>> > in the 'directory' using a context called family: ? switch_ivr.c:1367 >>> > Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family] >>> > >>> > This is an extract from sofia: >>> > >>> > sofia status profile internal >>> > >>> > ================================================================================================= >>> > Name ? ? ? ? ? ? ? ? ? ?internal >>> > Domain Name ? ? ? ? ? ? N/A >>> > DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >>> > Pres Hosts >>> > Dialplan ? ? ? ? ? ? ? ?XML >>> > Context ? ? ? ? ? ? ? ? public >>> > Challenge Realm ? ? ? ? auto_from >>> > RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> > Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >>> > SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> > Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >>> > URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >>> > BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >>> > HOLD-MUSIC ? ? ? ? ? ? ?silence >>> > OUTBOUND-PROXY ? ? ? ? ?N/A >>> > CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >>> > TEL-EVENT ? ? ? ? ? ? ? 101 >>> > DTMF-MODE ? ? ? ? ? ? ? rfc2833 >>> > CNG ? ? ? ? ? ? ? ? ? ? 13 >>> > SESSION-TO ? ? ? ? ? ? ?0 >>> > MAX-DIALOG ? ? ? ? ? ? ?0 >>> > NOMEDIA ? ? ? ? ? ? ? ? false >>> > LATE-NEG ? ? ? ? ? ? ? ?false >>> > PROXY-MEDIA ? ? ? ? ? ? false >>> > AGGRESSIVENAT ? ? ? ? ? true >>> > STUN-ENABLED ? ? ? ? ? ?true >>> > STUN-AUTO-DISABLE ? ? ? false >>> > CALLS-IN ? ? ? ? ? ? ? ?100 >>> > FAILED-CALLS-IN ? ? ? ? 25 >>> > CALLS-OUT ? ? ? ? ? ? ? 38 >>> > FAILED-CALLS-OUT ? ? ? ?31 >>> > >>> > Registrations: >>> > >>> > ================================================================================================= >>> > Call-ID: ? ? ? ?68534BBA9B461526 at 58.169.138.53 >>> > User: ? ? ? ? ? 2000 at 192.168.1.120 >>> > Contact: ? ? ? ?"user" >>> > Agent: ? ? ? ? ?dunno >>> > Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) >>> > Host: ? ? ? ? ? freeswitch >>> > IP: ? ? ? ? ? ? 58.xxx.xxx.xxx >>> > Port: ? ? ? ? ? 5060 >>> > Auth-User: ? ? ?2000 >>> > Auth-Realm: ? ? markcs.dyndns.org >>> > MWI-Account: ? ?2000 at 192.168.1.120 >>> > >>> > The internal.xml file has a lot in it, but I guess these are the >>> > important things for this profile: >>> > >>> > ? ? >>> > ? ? >>> > >>> > ? ? >>> > ? ? >>> > >>> > I will try to change auto-nat to be $${external_sip_ip} >>> > >>> > One question though: ?Any idea why I never see the TRYING or RINGING >>> > messages? ? Are tehse related to the RTP IP address or not? ?Without >>> > these I assume something is incorrect and I do not hear ringback.... >>> > >>> > Thanks! >>> > >>> > On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris wrote: >>> >> Your packet traces would disagree with the statements below. ?It is >>> >> sending your internal address in rtp, so its not set correctly on >>> >> whatever profile your using to call out, >>> >> >>> >> MIke >>> >> >>> >> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: >>> >> >>> >>> Hi Mike, >>> >>> >>> >>> I should have put that in also. >>> >>> >>> >>> I do have external_rtp_ip set in my config. ?I have it set to my >>> >>> domain name: >>> >>> >>> >>> >>> >>> I should also mention that if I use flaphone.com (which registers with >>> >>> an external IP address), then I get audio. ?In sofia, I see my IP >>> >>> addresses: >>> >>> >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> = >>> >>> ====================================================================== >>> >>> Name ? ? ? ? ? ? ? ? ? ?internal >>> >>> Domain Name ? ? ? ? ? ? N/A >>> >>> DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal >>> >>> Pres Hosts >>> >>> Dialplan ? ? ? ? ? ? ? ?XML >>> >>> Context ? ? ? ? ? ? ? ? public >>> >>> Challenge Realm ? ? ? ? auto_from >>> >>> RTP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> >>> Ext-RTP-IP ? ? ? ? ? ? ?124.xxx.xxx.xxx >>> >>> SIP-IP ? ? ? ? ? ? ? ? ?192.168.1.120 >>> >>> Ext-SIP-IP ? ? ? ? ? ? ?124.xxx.xxx.x >>> >>> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.1.120:5060 >>> >>> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.1.120:5060 >>> >>> HOLD-MUSIC ? ? ? ? ? ? ?silence >>> >>> OUTBOUND-PROXY ? ? ? ? ?N/A >>> >>> CODECS ? ? ? ? ? ? ? ? ?G726-32,G722,PCMU,PCMA >>> >>> TEL-EVENT ? ? ? ? ? ? ? 101 >>> >>> DTMF-MODE ? ? ? ? ? ? ? rfc2833 >>> >>> CNG ? ? ? ? ? ? ? ? ? ? 13 >>> >>> SESSION-TO ? ? ? ? ? ? ?0 >>> >>> MAX-DIALOG ? ? ? ? ? ? ?0 >>> >>> NOMEDIA ? ? ? ? ? ? ? ? false >>> >>> LATE-NEG ? ? ? ? ? ? ? ?false >>> >>> PROXY-MEDIA ? ? ? ? ? ? false >>> >>> AGGRESSIVENAT ? ? ? ? ? true >>> >>> STUN-ENABLED ? ? ? ? ? ?true >>> >>> STUN-AUTO-DISABLE ? ? ? false >>> >>> >>> >>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris >>> >>> wrote: >>> >>>> You don't have ext-rtp-ip set in your config. >>> >>>> >>> >>>> Mike >>> >>>> >>> >>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: >>> >>>> >>> >>>>> Hi! >>> >>>>> >>> >>>>> I have FS natted and am connecting with an 'external' extension that >>> >>>>> is registered to FS. ?ie the extension 2000 is registered on the >>> >>>>> internet with a public IP through my router to FS (192.168.1.120 IP >>> >>>>> address). ?uPnP works and I see that the extension is registered >>> >>>>> successfully. >>> >>>>> >>> >>>>> The problem is that I do not get any audio >>> >>>>> >>> >>>>> When looking at the SIP trace, I see the INVITE but do not see a >>> >>>>> TRYING or RINGING message. ?The extension is actually ringing. ?I >>> >>>>> modified the RTP port range on the remote end to match the RTP ports >>> >>>>> of freeswitch. >>> >>>>> >>> >>>>> I have put a sip trace in the pastebin at >>> >>>>> http://pastebin.freeswitch.org/11035 >>> >>>>> >>> >>>>> If anyone has an idea what needs to be set to get audio, help >>> >>>>> appreciated. >>> >>>>> >>> >>>>> Thanks! >>> >>>> >>> >>>> >>> >>>> _______________________________________________ >>> >>>> FreeSWITCH-users mailing list >>> >>>> FreeSWITCH-users at lists.freeswitch.org >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> >>>> users >>> >>>> http://www.freeswitch.org >>> >>>> >>> >>> >>> >>> _______________________________________________ >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> >>> users >>> >>> http://www.freeswitch.org >>> >> >>> >> >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From shiyanov at gmail.com Sun Nov 8 23:04:10 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Mon, 9 Nov 2009 10:04:10 +0300 Subject: [Freeswitch-users] Dialpan: try.. finally analogs In-Reply-To: <191c3a030911061608w5be8af61y7bc10fe2d23dfc4a@mail.gmail.com> References: <191c3a030911061608w5be8af61y7bc10fe2d23dfc4a@mail.gmail.com> Message-ID: Closed. As (almost) usual the reason was me. Anthony's hint works perfectly: api uuid_transfer bridge:sofia/gateway// inline Sorry for bothering! On Sat, Nov 7, 2009 at 3:08 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > If you know the reason, why are you so puzzled by it? > I think you should not assume you understand what is happening unless you > really do. > > I think you need to provide an exact description of what you are doing so I > can explain to you where you are making the mistake. > > Make sure you are on latest SVN and reproduce this in a console log for us > and add an exact description of what you are doing in detail. > > > On Thu, Nov 5, 2009 at 11:44 AM, Artem Shiyanov wrote: > >> Hello! >> >> I have to deal with classic problem: "Leaking stream handle" in FS >> console. I also know the reason - firstly channel is sent to the extension >> with "playback" and later it is transfered to another extensions with >> "execute_extension" or, another trouble-case - channel is bridged to some >> addres. >> I do not ask (but I wish to) why FS doesn't close stream automatically >> when channel is gone. >> I ask whether it is possible to use some "try.. finally" construction in >> diaplan? If "yes" then I can simply stop playback in the "finally" block.. >> >> Any thoughs? >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/bda5ce2f/attachment-0002.html From bruce at nani.ca Mon Nov 9 00:07:14 2009 From: bruce at nani.ca (Bruce Fletcher) Date: Mon, 9 Nov 2009 00:07:14 -0800 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> Message-ID: <2742E007-4C51-4E1A-96C6-B047F82174F2@nani.ca> The patch from the PortAudio site does get the library to build, but it still fails with the same assertion when I try to play MOH. The patch I'm talking about is this one: http://www.portaudio.com/trac/changeset/1418 If the same build problem applies to other 64 bit systems, it might be a good idea to incorporate this patch. It looks clean and reasonable to me, at least. I've managed to work around the problem entirely by building FreeSWITCH for i386, but I'll go ask the PortAudio folks what the status is of their 64 bit support. They are clearly assuming 32 bit long integers in some places, which is hopefully on a to-fix list somewhere. Thanks, - Bruce On 2009-11-08, at 12:25 PM, Michael Jerris wrote: > If you can figure out a clean way for us to do this with proper ifdefs > in tree in a way that will not break others that would be the most > preferred. > > Mike > > On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote: > >> OK, I'll ignore that MacPorts patch for now and try to find a better >> approach. >> >> I'll look into this further tonight, but this morning I found a more >> recent promising patch on the PortAudio site: >> >> http://www.portaudio.com/trac/changeset/1418 >> >> It seems to push some data types to 32 bit regardless of platform, >> which might work better than the MacPorts approach of migrating some >> data structures to 64 bit. At any rate, this patch being on the >> PortAudio site suggests it might be a more approved fix. >> >> I'll keep plugging at this in my free time and report any significant >> progress back to the list. >> >> Thanks, >> - Bruce >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From shouldbeq931 at googlemail.com Mon Nov 9 03:23:43 2009 From: shouldbeq931 at googlemail.com (shouldbe q931) Date: Mon, 9 Nov 2009 11:23:43 +0000 Subject: [Freeswitch-users] building on Fedora 12 Message-ID: <649eaa470911090323m543cca02td862834979d09949@mail.gmail.com> Hi, While I appreciate that Fedora 12 is still only in beta. Because I want to try FusionPBX and I've had no success getting pfSense to work in a single NIC environment, and FusionPBX needs PHP 5.3, and Fedora 12 appears to be the first distro with PHP5.3... Anyway, these are the steps that I've done yum install autoconf automake binutils bison curl-devel db4 db4-devel expat-devel flex gcc-c++ gettext gdb gdbm gdbm-devel gnutls-devel httpd kernel-devel libogg-devel libtiff libtiff-devel libtool libvorbis-devel make mkxauth mysql-server mysql mysql-devel ncurses ncurses-devel ncurses-libs openssl-devel perl-Apache2-SOAP perl-devel php php-common php-mysql php-pdo php-soap php-xmlrpc postfix python python-devel screen sqlite sqlite-devel sqlite2 sqlite2-devel subversion unixODBC-devel wget wireshark wireshark-gnome zlib zlib-devel mkdir /usr/src/freeswitch mkdir /usr/src/freeswitch/svn svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk /usr/src/freeswitch/svn cd /usr/src/freeswitch/svn ./bootstrap.sh ./configure make SVN reports Checked out external at revision 849. Checked out revision 15396. I then get the following error at the end -------------------------------------------------------------- Making all in soa Making all in tport LTCOMPILE tport_tls.lo cc1: warnings being treated as errors tport_tls.c: In function ?tls_init_context?: tport_tls.c:280: error: assignment discards qualifiers from pointer target type tport_tls.c:282: error: assignment discards qualifiers from pointer target type tport_tls.c: In function ?tls_post_connection_check?: tport_tls.c:527: error: assignment discards qualifiers from pointer target type make[9]: *** [tport_tls.lo] Error 1 make[8]: *** [all] Error 2 Making all in nta Making all in nth Making all in nea Making all in iptsec Making all in nua make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [//usr/src/freeswitch/svn/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 [root at conf-2 svn]# -------------------------------------------------------------- I then tried using the "quick and dirty" method from http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install, but get a similar failure -------------------------------------------------------------- Making all in nua make[9]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' make[10]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la make[10]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua' make[9]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[9]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua' make[8]: *** [all-recursive] Error 1 make[8]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua' Making all in packages make[8]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[8]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[7]: *** [all-recursive] Error 1 make[7]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[6]: *** [all] Error 2 make[6]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[5]: *** [/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[5]: Leaving directory `/usr/src/freeswitch.trunk/src/mod/endpoints/mod_sofia' make[4]: *** [mod_sofia-all] Error 1 make[4]: Leaving directory `/usr/src/freeswitch.trunk/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch.trunk/src' Making all in build make[3]: Entering directory `/usr/src/freeswitch.trunk/build' +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[3]: Leaving directory `/usr/src/freeswitch.trunk/build' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch.trunk' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch.trunk' make: *** [freeswitch] Error 2 [root at conf-2 src]# -------------------------------------------------------------- I'm still very much a noob when it comes to building, but I'd welcome any constructive suggestions on how to resolve this. Cheers Arne From lakindia89 at gmail.com Mon Nov 9 03:53:52 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 9 Nov 2009 17:23:52 +0530 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events Message-ID: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> Dear all, I did the below code, to callback a function when CHANNEL_EXECUTE_COMPLETE event comes. I executed the script for the 1st time and I got nothing. When I executed the script for the 2nd time, it ended with Sedmentation fault with core dumped. I was unable to attach the core dump file with this mail. Please specify how to send files to freeswitch user mailing list if need be. The freeswitch log is here: http://pastebin.freeswitch.org/11038 #!/usr/bin/perl use strict; use Data::Dumper; our $session; $session->answer(); my $events=new freeswitch::EventConsumer("CHANNEL_EXECUTE_COMPLETE"); $events->pop(1); $events->swig_e_callback_set("playvoice"); sub playvoice() { freeswitch::consoleLog("INFO","Call back function called\n"); } return 1; -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/858ab30b/attachment-0002.html From edpimentl at gmail.com Mon Nov 9 04:05:53 2009 From: edpimentl at gmail.com (EdPimentl) Date: Mon, 9 Nov 2009 07:05:53 -0500 Subject: [Freeswitch-users] building on Fedora 12 In-Reply-To: <649eaa470911090323m543cca02td862834979d09949@mail.gmail.com> References: <649eaa470911090323m543cca02td862834979d09949@mail.gmail.com> Message-ID: <9dc4a1670911090405u56502857pcc647204dc1ffc4@mail.gmail.com> Any reason for not using uBuntu? Install Freeswitch + FusionPBX on Ubuntu step 1) add the fallowing lines to /etc/apt/ file. deb http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntuhardy main deb-src http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntuhardy main step 2) apt-get update step3) apt-get install freeswitch deps -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/4dfd7248/attachment-0002.html From markmorreny at gmail.com Mon Nov 9 04:59:54 2009 From: markmorreny at gmail.com (mark morreny) Date: Mon, 9 Nov 2009 20:59:54 +0800 Subject: [Freeswitch-users] playback from hadoop Message-ID: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> Hi, Does anyone know how to playback based on files from hadoop storage. There is a libhdcp, and java api. Is there anyway to put together a sample middle piece to move files from hadoop to freeswitch using memory space, so there is no disk I/O? Any feedback or suggestion will be greatly appreciated. thx, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/bce7dde5/attachment-0002.html From testeador01 at gmail.com Mon Nov 9 05:30:53 2009 From: testeador01 at gmail.com (Milena) Date: Mon, 9 Nov 2009 08:30:53 -0500 Subject: [Freeswitch-users] Valid Dial Strings In-Reply-To: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> Message-ID: Hello, When you post something on pastebin, please post the link to your post so everyone can find it, what is the link to it? Have a nice day :) 2009/11/7 Dave Stevenson > Hi Michael, > > thanks for the reply. I think that I have got to the bottom of how to allow > numbers to get to the VOIP gateway - at the moment, my dialplan just allows > any. > > The big problem is that the VOIP Gateway (Linksys 3102) rejects any calls > to > it from VOIP to the PSTN and I don't know why. > I have posted a dump to the pastebin, hopefully, the messages in there will > allow someone to see what the problem is and give me some pointers on how I > might fix it > > regards > Dave > > > > > ----- Original Message ----- > From: "Michael S Collins" > To: > Sent: Saturday, November 07, 2009 6:49 PM > Subject: Re: [Freeswitch-users] Valid Dial Strings > > > > > > On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" > > wrote: > > > >> Hi, > >> > >> can someone pointme to where the valid dialing strings are specified ? > >> > > For SIP dialstrings check here: > > http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings > > > > Also, if you send us examples of what you've tried we can help you > > figure out what's wrong. > > > >> I'm assuming that something, somewhere, tells FS that numbers are > >> invalid > >> before they get dialed ? > > > > Pastebin some debug logs of what's happening. Check out this page > > which has lots of useful information on how to collect information: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > > > It sounds like it's just a matter of figuring out how to configure > > your specific setup. Please report back with more information and > > we'll be happy to help. > > > > -MC > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/9870b09b/attachment-0002.html From shouldbeq931 at googlemail.com Mon Nov 9 05:48:26 2009 From: shouldbeq931 at googlemail.com (shouldbe q931) Date: Mon, 9 Nov 2009 13:48:26 +0000 Subject: [Freeswitch-users] building on Fedora 12 In-Reply-To: <9dc4a1670911090405u56502857pcc647204dc1ffc4@mail.gmail.com> References: <649eaa470911090323m543cca02td862834979d09949@mail.gmail.com> <9dc4a1670911090405u56502857pcc647204dc1ffc4@mail.gmail.com> Message-ID: <649eaa470911090548m9002e98l36def833e64b7c84@mail.gmail.com> Hi Ed, I installed Jaunty ( I don't have Hardy to hand) rather than /etc/apt, I presume you mean /etc/apt/sources.list after a "sudo apt-get update" I did a "sudo apt-get install freeswitch" I'm not sure what you meant by "deps" by your step 3 I then edited /etc/defaults/freeswitch and set false to true, saved the file and restarted, unfortunately freeswitch does not start, I had seen on the wiki that the debian packager was putting something in the wrong place, but I'm not sure where I would look for logs to show why it doesn't start. I will try and download a copy of hardy later on, and see if it has the same issues. Cheers Arne On Mon, Nov 9, 2009 at 12:05 PM, EdPimentl wrote: > Any reason for? not using uBuntu? > > Install Freeswitch + FusionPBX on Ubuntu > > step 1) add the fallowing?lines to /etc/apt/?file. > > deb > http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntu > hardy main > deb-src > http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntu > hardy main > > step 2) apt-get update > > step3) apt-get install freeswitch deps > > -E > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From stevendt at primrosebank.net Mon Nov 9 05:55:48 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 13:55:48 -0000 Subject: [Freeswitch-users] Valid Dial Strings References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> Message-ID: <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> Milena, thanks a lot for the reply - sorry, I'm new to this, but I'll remember that for next time. Actually, I found my way to the IRC site and the helpful chaps there got to the bottom of my problem. I had made an error copying the dialplan data from the "SPA3102 FreeSwitch HowTo" http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo I had read as i.e., substituted "(...)" for "{...}" Being unfamiliar with the FreeSwitch dialplans, I'd NEVER have found the problem without help, regards Dave ----- Original Message ----- From: Milena To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 1:30 PM Subject: Re: [Freeswitch-users] Valid Dial Strings Hello, When you post something on pastebin, please post the link to your post so everyone can find it, what is the link to it? Have a nice day :) 2009/11/7 Dave Stevenson Hi Michael, thanks for the reply. I think that I have got to the bottom of how to allow numbers to get to the VOIP gateway - at the moment, my dialplan just allows any. The big problem is that the VOIP Gateway (Linksys 3102) rejects any calls to it from VOIP to the PSTN and I don't know why. I have posted a dump to the pastebin, hopefully, the messages in there will allow someone to see what the problem is and give me some pointers on how I might fix it regards Dave ----- Original Message ----- From: "Michael S Collins" To: Sent: Saturday, November 07, 2009 6:49 PM Subject: Re: [Freeswitch-users] Valid Dial Strings > > On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" > wrote: > >> Hi, >> >> can someone pointme to where the valid dialing strings are specified ? >> > For SIP dialstrings check here: > http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings > > Also, if you send us examples of what you've tried we can help you > figure out what's wrong. > >> I'm assuming that something, somewhere, tells FS that numbers are >> invalid >> before they get dialed ? > > Pastebin some debug logs of what's happening. Check out this page > which has lots of useful information on how to collect information: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It sounds like it's just a matter of figuring out how to configure > your specific setup. Please report back with more information and > we'll be happy to help. > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/ac0e3bde/attachment-0002.html From rupa at rupa.com Mon Nov 9 06:08:52 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 9 Nov 2009 06:08:52 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175B572244@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> <3C04B27FC880044F8FCD735D0D952FF7175B572244@EXMBXCLUS01.citservers.local> Message-ID: On Fri, Nov 6, 2009 at 7:59 AM, Ujjval Karihaloo wrote: > ?Any examples I can refer to for this? not that i know of > > Like for Channel vars and execute_application calls? Does this all need to be doen in dialplan.public.xml or also in other config files? most can be done in public.xml if you want to do it all in dialplan. > Sorry: I am still learning the Freeswitch world. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker > Sent: Thursday, November 05, 2009 8:56 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > This is true, BUT it is more flexible than it looks. > > http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E > > The caller controls can have a key execute a dialplan extension: > execute_application > You can set a channel var on the moderator prior to joining to the conf. > When the extenion is called, you can check the channel var for > moderator and act accordingly. > > Or you can send an event and monitor with an app over ESL and do > whatever you want there (probably using the same channel var trick for > knowing who is a mod or not). > > > On Thu, Nov 5, 2009 at 8:52 AM, Rob Forman wrote: >> Hi UK, >> >> ?From what I've done and read, the caller-controls (in >> conference.conf.xml) can be modified to almost anything you can think >> of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a >> caller control just for those with the moderator flag. ?So unless you >> want everyone able to mute/kick everyone then you can't do it. >> >> The wiki seems to indicate this as well: >> >> "Be aware that the caller-controls are applied across the entire >> conference. You cannot enter one member of the conference using caller- >> controls ABC and then enter a second member using caller-controls XYZ." >> >> http://wiki.freeswitch.org/wiki/Mod_conference >> >> >> I think this might be a limitation of mod_conference. ?Perhaps one of >> the pros can chime in if I'm off-base or there's some nifty way to >> accomplish this. >> >> Cheers, >> Rob >> >> On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: >> >>> Any ideas on the below...has anyone implemented the below: >>> >>> Once I have the Moderator and Participants logged on, how do I >>> invoke the moderator previlidges, LIk esay muting everyone/someone >>> or kicking someone out of the Conf and the like? >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, November 02, 2009 12:52 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Rob: >>> >>> ? Once I have the Moderator and Participants logged on, how do I >>> invoke the moderator previlidges, LIk esay muting everyone/someone >>> or kicking someone out of the Conf and the like? >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 9:34 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Hm, strange. ?I haven't seen that before. ?Can you pastebin your logs >>> at debug level? >>> >>> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >>> >>>> It's strange... a tcpdump tells me that there is no DTMF from my >>>> provider when using IVR, but when I call into a TN that goes >>>> directly into the Conference App, I see DTMF from the provider. >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman >>>> Sent: Friday, October 30, 2009 7:23 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>> >>>> I've never had any problem with that. ?Is your logging at debug level >>>> so you can see the RECV DTFM in the log/fs_cli? ?Are you calling from >>>> a SIP phone on the pbx, or via a PSTN provider? ?Maybe your provider >>>> isn't passing them through. >>>> >>>> Make sure your logging is turned up then try something simpler, like >>>> calling the echo application, and see if DTFM comes through. >>>> >>>> Rob >>>> >>>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>>> >>>>> Rob: >>>>> >>>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>>> using the below configuration....so it basically timesout. >>>>> >>>>> UK >>>>> >>>>> >>>>> >>>>> -----Original Message----- >>>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>>> ] On Behalf Of Ujjval Karihaloo >>>>> Sent: Monday, October 26, 2009 9:21 AM >>>>> To: freeswitch-users at lists.freeswitch.org >>>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>>> >>>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>>> correct.. >>>>> =============== >>>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>>> with an IVR where the moderators are authenticated with an extra >>>>> pass- >>>>> code", which is what I did. ?I guess that's why I didn't understand >>>>> the point of the +pin. >>>>> ====================== >>>>> >>>>> I will try it out. >>>>> >>>>> Again thx a lot for your help. Will keep everyone posted. >>>>> >>>>> ________________________________________ >>>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>>> Sent: Friday, October 23, 2009 12:22 PM >>>>> To: freeswitch-users at lists.freeswitch.org >>>>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>>>> >>>>> I just re-tested with the pin in my dial plan: >>>>> >>>>> >>>>> >>>>> And it doesn't challenge me for the pin. ?I just drop right in. ?I >>>>> figured this is how it was intended, since the wiki says the pin is >>>>> set initially and only challenged in later attempts [by future >>>>> callers]: >>>>> >>>>> "The first time a conference name (confname) is used, it will be >>>>> created on demand, and the pin will be set to what ever is specified >>>>> at that time: the pin in the data string if specified, or if not, >>>>> the >>>>> "pin" setting in the conference profile, and if that is also >>>>> unspecified, then there is no pin protection. Any later attempt to >>>>> join the conference must specify the same pin number, if one existed >>>>> when it was created. " >>>>> >>>>> >>>>> The wiki also says that the wait-mod might be ?"used in conjunction >>>>> with an IVR where the moderators are authenticated with an extra >>>>> pass- >>>>> code", which is what I did. ?I guess that's why I didn't understand >>>>> the point of the +pin. >>>>> >>>>> I'm sure there's a scenario where its used and useful, the wiki just >>>>> doesn't explain it. >>>>> >>>>> Rob >>>>> >>>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>>> >>>>>> Well first off you're not defining a pine here... >>>>>> >>>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>>> pin >>>>>> number] >>>>>> >>>>>> That might be why its not asking for a pin. >>>>>> >>>>>> /b >>>>>> >>>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>>> >>>>>>> ? >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From gshfreesw at gmail.com Mon Nov 9 06:44:16 2009 From: gshfreesw at gmail.com (Shameem Shiek) Date: Mon, 9 Nov 2009 09:44:16 -0500 Subject: [Freeswitch-users] SIP Provider with unlimited channels Message-ID: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> Dear Freeswitch Users, I am looking for a SIP Provider who can provide a DID with unlimited channels. Currently I am using junction networks but they have a high 2.9c/minute charge. I am looking for someone who has a flat rate for X minutes. Any advise would be much appreciated. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/8e762338/attachment-0002.html From kristian.kielhofner at gmail.com Mon Nov 9 07:27:55 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 9 Nov 2009 10:27:55 -0500 Subject: [Freeswitch-users] SIP Provider with unlimited channels In-Reply-To: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> References: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> Message-ID: <2d9149cd0911090727j71de4c2bv9e5219ce37212037@mail.gmail.com> Beware of anyone that claims to offer "unlimited" channels. We're still fundamentally a TDM world and there is no such thing as unlimited. Depending on what you are looking for there are probably plenty of providers with a high enough limit to satisfy your actual needs. I just frown upon anyone claiming to offer "unlimited" channels because they either don't have an understanding of their true capacity (some tier X reseller who doesn't realize the upstream providers ALL have limits) or they don't care they are advertising falsely. Sure you don't really need "unlimited" channels. What are you looking for? 100? 1000? On Mon, Nov 9, 2009 at 9:44 AM, Shameem Shiek wrote: > Dear Freeswitch Users, > > ?I am looking for a SIP Provider who can provide? a DID with unlimited > channels. Currently I am using junction networks but they have a high > 2.9c/minute charge. I am looking for someone who has a flat rate for X > minutes. > > Any advise would be much appreciated. > > Thanks. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From kristian.kielhofner at gmail.com Mon Nov 9 07:30:53 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 9 Nov 2009 10:30:53 -0500 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information In-Reply-To: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> References: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> Message-ID: <2d9149cd0911090730j41c4738qe14fa9b14b9b2069@mail.gmail.com> This is for outbound calls, calling party name. The OP is talking about called party name, which is the neat feature of being able to update the display of the calling user with the name of the called user (instead of just displaying their numeric extension for the duration of the call). On Sun, Nov 8, 2009 at 9:26 PM, SP wrote: > before playing with mod_sofia, did you try the sip_cid_type variable? > > http://wiki.freeswitch.org/wiki/Variable_sip_cid_type > > On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine wrote: >> Hello, >> >> ??While?trying to display the called party name ?on SNOM phones I've found >> that the field sent to the phone needs to be changed slightly in order to >> make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects >> Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and Cisco >> work ok. Just wanted to let the developers know... >> >> ? And now a question: We have SNOM phones monitoring other extensions (BLF >> feature). When a call comes in, the monitoring phones get notification, but >> the name field (identity display) contains the calling extension number and >> not its display name. Can this be fixed? >> >> ??????????????????????????????? Thanks! __Yehavi: >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Shannon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Mon Nov 9 07:45:49 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Nov 2009 09:45:49 -0600 Subject: [Freeswitch-users] SIP Provider with unlimited channels In-Reply-To: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> References: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> Message-ID: <614163A1-E814-42CB-B953-218FB55D7F63@freeswitch.org> Have you tried Bandwidth.com or iCall? /b On Nov 9, 2009, at 8:44 AM, Shameem Shiek wrote: > Dear Freeswitch Users, > > I am looking for a SIP Provider who can provide a DID with > unlimited channels. Currently I am using junction networks but they > have a high 2.9c/minute charge. I am looking for someone who has a > flat rate for X minutes. > > Any advise would be much appreciated. > > Thanks. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Nov 9 08:01:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 10:01:00 -0600 Subject: [Freeswitch-users] Remote-Party-ID issue and call pickup information In-Reply-To: <2d9149cd0911090730j41c4738qe14fa9b14b9b2069@mail.gmail.com> References: <7e2ac3270911081826k3f5fe71fp9f14d28b87e0239c@mail.gmail.com> <2d9149cd0911090730j41c4738qe14fa9b14b9b2069@mail.gmail.com> Message-ID: <191c3a030911090801k79b42baeyf31059d67d41fcd@mail.gmail.com> If the patch is not received today it will not make it into 1.0.5 On Mon, Nov 9, 2009 at 9:30 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > This is for outbound calls, calling party name. The OP is talking > about called party name, which is the neat feature of being able to > update the display of the calling user with the name of the called > user (instead of just displaying their numeric extension for the > duration of the call). > > On Sun, Nov 8, 2009 at 9:26 PM, SP wrote: > > before playing with mod_sofia, did you try the sip_cid_type variable? > > > > http://wiki.freeswitch.org/wiki/Variable_sip_cid_type > > > > On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine > wrote: > >> Hello, > >> > >> While trying to display the called party name on SNOM phones I've > found > >> that the field sent to the phone needs to be changed slightly in order > to > >> make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects > >> Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and > Cisco > >> work ok. Just wanted to let the developers know... > >> > >> And now a question: We have SNOM phones monitoring other extensions > (BLF > >> feature). When a call comes in, the monitoring phones get notification, > but > >> the name field (identity display) contains the calling extension number > and > >> not its display name. Can this be fixed? > >> > >> Thanks! __Yehavi: > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > > > -- > > Shannon > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/63790bfb/attachment-0002.html From rob4manhere at gmail.com Mon Nov 9 08:02:07 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Mon, 9 Nov 2009 10:02:07 -0600 Subject: [Freeswitch-users] SIP Provider with unlimited channels In-Reply-To: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> References: <5070fcbd0911090644y6ddf48e5h9e33961f6935314d@mail.gmail.com> Message-ID: <613F14B9-1AF1-4FEE-A206-C3FCECE4DE33@gmail.com> I agree there is no such thing as unlimited. The three ways most SIP providers will structure pricing is 1) per minute (ie $0.02/minute), 2) per channel (ie $15/month) or 3) "unlimited" with a channel limit (ie $7/month for any amount of minutes but after two simultaneous channels its ring busy). If you don't want to be limited by channels, then it sounds like your only choice is per minute. Some providers will say their per-minute is unlimited channels but you should ask them what there hard limit is (because there is one) and if it can be increased should you need more capacity. I've used iCall and am pretty happy with their service and pricing choices. I think their default channel limit on per-minute billing is 100. On Nov 9, 2009, at 8:44 AM, Shameem Shiek wrote: > Dear Freeswitch Users, > > I am looking for a SIP Provider who can provide a DID with > unlimited channels. Currently I am using junction networks but they > have a high 2.9c/minute charge. I am looking for someone who has a > flat rate for X minutes. > > Any advise would be much appreciated. > > Thanks. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shouldbeq931 at googlemail.com Mon Nov 9 08:08:33 2009 From: shouldbeq931 at googlemail.com (shouldbe q931) Date: Mon, 9 Nov 2009 16:08:33 +0000 Subject: [Freeswitch-users] building on Fedora 12 In-Reply-To: <649eaa470911090548m9002e98l36def833e64b7c84@mail.gmail.com> References: <649eaa470911090323m543cca02td862834979d09949@mail.gmail.com> <9dc4a1670911090405u56502857pcc647204dc1ffc4@mail.gmail.com> <649eaa470911090548m9002e98l36def833e64b7c84@mail.gmail.com> Message-ID: <649eaa470911090808g2d2e2b97nccf627533a49d2c3@mail.gmail.com> Hi Ed, I've just finished installing Hardy, and following the same steps again, freeswitch is not running. Any suggestions ? On Mon, Nov 9, 2009 at 1:48 PM, shouldbe q931 wrote: > Hi Ed, > > I installed Jaunty ( I don't have Hardy to hand) > > rather than /etc/apt, I presume you mean /etc/apt/sources.list > > after a "sudo apt-get update" I did a "sudo apt-get install > freeswitch" I'm not sure what you meant by "deps" by your step 3 > > I then edited /etc/defaults/freeswitch and set false to true, saved > the file and restarted, unfortunately freeswitch does not start, I had > seen on the wiki that the debian packager was putting something in the > wrong place, but I'm not sure where I would look for logs to show why > it doesn't start. > > I will try and download a copy of hardy later on, and see if it has > the same issues. > > Cheers > > Arne > > > On Mon, Nov 9, 2009 at 12:05 PM, EdPimentl wrote: >> Any reason for? not using uBuntu? >> >> Install Freeswitch + FusionPBX on Ubuntu >> >> step 1) add the fallowing?lines to /etc/apt/?file. >> >> deb >> http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntu >> hardy main >> deb-src >> http://ppa.launchpad.net/freeswitch-drivers/freeswitch-nightly-drivers/ubuntu >> hardy main >> >> step 2) apt-get update >> >> step3) apt-get install freeswitch deps >> >> -E >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From rob4manhere at gmail.com Mon Nov 9 08:19:27 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Mon, 9 Nov 2009 10:19:27 -0600 Subject: [Freeswitch-users] javascript parameter In-Reply-To: <200911090034059066890@gmail.com> References: <200911090034059066890@gmail.com> Message-ID: <0323900B-D84A-4872-8690-728D62C74BC7@gmail.com> You can check the numbers of arguments passed with argc, and access them via argv[0], argv[1], etc. Its hinted at on the main Javascript wiki page, and also detailed in the FAQ. http://wiki.freeswitch.org/wiki/Javascript_FAQ On Nov 8, 2009, at 10:34 AM, god.nirvana wrote: > hi all: > how can i get the value of the myArg1 myArg2 in test.js. > like this originate sofia/example/1000 at somewhere.com > '&javascript(test.js myArg1 myArg2)' > thanks! > > 2009-11-09 > god.nirvana > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/1117c449/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 9 08:34:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 10:34:58 -0600 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events In-Reply-To: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> References: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> Message-ID: <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> 1) install gdb 2) run support_d/fscore_db in the tree from the working directory of the core. 3) if you are not on svn trunk, "make current" and start over. On Mon, Nov 9, 2009 at 5:53 AM, lakshmanan ganapathy wrote: > Dear all, > I did the below code, to callback a function when CHANNEL_EXECUTE_COMPLETE > event comes. > I executed the script for the 1st time and I got nothing. > When I executed the script for the 2nd time, it ended with Sedmentation > fault with core dumped. > > I was unable to attach the core dump file with this mail. > Please specify how to send files to freeswitch user mailing list if need > be. > > The freeswitch log is here: > http://pastebin.freeswitch.org/11038 > > #!/usr/bin/perl > use strict; > use Data::Dumper; > our $session; > $session->answer(); > my $events=new freeswitch::EventConsumer("CHANNEL_EXECUTE_COMPLETE"); > $events->pop(1); > $events->swig_e_callback_set("playvoice"); > sub playvoice() > { > freeswitch::consoleLog("INFO","Call back function called\n"); > } > return 1; > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/1561093e/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 9 08:50:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 10:50:40 -0600 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <2742E007-4C51-4E1A-96C6-B047F82174F2@nani.ca> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> <2742E007-4C51-4E1A-96C6-B047F82174F2@nani.ca> Message-ID: <191c3a030911090850j4d9e7972m15fa4661a2da7926@mail.gmail.com> maybe we should write a new audio abstraction lib =D On Mon, Nov 9, 2009 at 2:07 AM, Bruce Fletcher wrote: > The patch from the PortAudio site does get the library to build, but > it still fails with the same assertion when I try to play MOH. The > patch I'm talking about is this one: > > http://www.portaudio.com/trac/changeset/1418 > > If the same build problem applies to other 64 bit systems, it might be > a good idea to incorporate this patch. It looks clean and reasonable > to me, at least. > > I've managed to work around the problem entirely by building > FreeSWITCH for i386, but I'll go ask the PortAudio folks what the > status is of their 64 bit support. They are clearly assuming 32 bit > long integers in some places, which is hopefully on a to-fix list > somewhere. > > Thanks, > - Bruce > > > On 2009-11-08, at 12:25 PM, Michael Jerris wrote: > > > If you can figure out a clean way for us to do this with proper ifdefs > > in tree in a way that will not break others that would be the most > > preferred. > > > > Mike > > > > On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote: > > > >> OK, I'll ignore that MacPorts patch for now and try to find a better > >> approach. > >> > >> I'll look into this further tonight, but this morning I found a more > >> recent promising patch on the PortAudio site: > >> > >> http://www.portaudio.com/trac/changeset/1418 > >> > >> It seems to push some data types to 32 bit regardless of platform, > >> which might work better than the MacPorts approach of migrating some > >> data structures to 64 bit. At any rate, this patch being on the > >> PortAudio site suggests it might be a more approved fix. > >> > >> I'll keep plugging at this in my free time and report any significant > >> progress back to the list. > >> > >> Thanks, > >> - Bruce > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/6b131013/attachment-0002.html From codecomplete at free.fr Mon Nov 9 09:01:45 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 9 Nov 2009 09:01:45 -0800 (PST) Subject: [Freeswitch-users] cd-sounds vs. sounds? Message-ID: <26269842.post@talk.nabble.com> Hello I successfully installed FreeSwitch from SVN, and am now prompted to install the sound files. Am I correct in understanding that "sounds" are POTS-grade files (8KHz?) while "cd-sounds" are closer to VoIP-grade (16KHz?), and "hd-sounds" and "uhd-sounds" are for Skype-grade sound files? In that case, if I only need to play sound files to POTS callers, I only need the "sounds" files? Thank you. -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26269842.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Nov 9 09:34:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 09:34:17 -0800 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <26269842.post@talk.nabble.com> References: <26269842.post@talk.nabble.com> Message-ID: <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> On Mon, Nov 9, 2009 at 9:01 AM, Fred-145 wrote: > > Hello > > I successfully installed FreeSwitch from SVN, and am now prompted to > install > the sound files. Am I correct in understanding that "sounds" are POTS-grade > files (8KHz?) while "cd-sounds" are closer to VoIP-grade (16KHz?), and > "hd-sounds" and "uhd-sounds" are for Skype-grade sound files? > > In that case, if I only need to play sound files to POTS callers, I only > need the "sounds" files? > I recommend you just do this: make cd-sounds-install && make cd-moh-install This will install all the sounds (8k, 16k, 32k, 48k) and thus FS can play whatever sampling rate is necessary. I promise you that this is the easiest way to go. Others have tried to save a few megabytes of disk space by not installing the higher quality sounds and music and have ended up wasting hours debugging "issues with playing sound files." For the record, the 8k files are just "sounds" and then you have hd (16k), uhd (32k), and cd (48k). -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/b1318cf7/attachment-0002.html From msc at freeswitch.org Mon Nov 9 09:37:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 09:37:46 -0800 Subject: [Freeswitch-users] SPA3102 FreeSwitch HowTo Wiki - HELP Please! In-Reply-To: <7B9E5C0A81154EDB8B2F2979EAF0BA17@bp1.ad.bp.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <9A3B9B304B1B440FB55BE1F88627437D@bp1.ad.bp.com> <2d9149cd0911041257w3f65b32bpe19c4e6feac77d6a@mail.gmail.com> <1D5C5D5D073043D5AA5705EF9474E0A1@bp1.ad.bp.com> <665C8F93976F422486C2A81A8A4B5483@bp1.ad.bp.com> <87f2f3b90911041627r6869139ej39712eeed1456288@mail.gmail.com> <97FBB4B6002848BCA4F2D89F13626754@bp1.ad.bp.com> <41A5CF92E4E94547BCE301E0F5A5B79B@bp1.ad.bp.com> <7B9E5C0A81154EDB8B2F2979EAF0BA17@bp1.ad.bp.com> Message-ID: <87f2f3b90911090937t39d74fc7h3f887bac28c9e1b5@mail.gmail.com> Two quick questions: which version of FreeSWITCH are you running? If you're not on the latest then I recommend getting SVN trunk or at the very least the latest prerelease. Second, can you capture the debug information and use pastebin? It makes it much easier for everyone to help you. http://pastebin.freeswitch.org. Also, this page might be helpful: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has handy tips on collecting information and asking the community for assistance. -MC On Sat, Nov 7, 2009 at 6:42 AM, Dave Stevenson wrote: > Follow up to previous post..... regarding making outgoing calls. > > I ***think*** that I have configured a dialplan that allows the user to > dial out but the requests seem to be getting rejected by the SPA3102. > > I can dial 0 and the FreeSwitch attendant will connect to the PSTN line > (FreeSwitch reports that the call has been answered). > Similarly, I can dial 1000 - the SPA3102 extension number with similar > results. > > However, if a try to dial an external number, the gateway rejects the call. > > I have captured some of the debug log but the info in there is way over my > head, can anyone help me understand what it's telling me please ? > > regards > Dave > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/78acf0ba/attachment-0002.html From ddechev at nutel.cc Mon Nov 9 02:10:26 2009 From: ddechev at nutel.cc (Dimitar Dechev) Date: Mon, 9 Nov 2009 12:10:26 +0200 Subject: [Freeswitch-users] Monitoring via SNMP Message-ID: <001001ca6124$dc9268e0$95b73aa0$@cc> Dear All, I couldn't find much information about how to monitor Freeswitch via SNMP like how many calls/legs I have, how many CAPs, and etc. One of the thing I do currently is to make simple bash script which in general runs "fs_cli -x 'show calls count'" or some other command and call that script via snmpd.conf. I would appreciate if somebody tell me if they is snmp module in Freeswitch, or provide with link/method from where I can read some information. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/b6c4f2fa/attachment-0002.html From dharding at nucleus.com Mon Nov 9 04:10:56 2009 From: dharding at nucleus.com (Dana Harding) Date: Mon, 9 Nov 2009 05:10:56 -0700 Subject: [Freeswitch-users] suggestions for hardware. References: <4AF4AF73.8070804@tx.rr.com> Message-ID: <11A13463A6CB4AC196401DA7AE7E6F34@danahome> > I am looking on advice on how to set up a simple office PBX, 20 phones > and 4 outside lines.with 2 or 3 "operator" phones and the rest will be > extensions. > > 4 spa3000's to handle the outside lines. > 2-3 polycom 601 phones with expansion modules (Operator phones) > 18 polycom 330 or other phones for desks. > 2-24 port cisco POE switches > 1 pfSense server. The last time I tried using spa3000's for PSTN connections, I had a lot of difficulty tuning the settings to get rid of some echo the users were hearing. I don't know what the local loop length was, and the wiring in that neighbourhood is pretty old - both probably had a strong influence on what I was seeing. My determination at that time (~3 years ago) was that they were good for home connections, but not suitable for business use. I ultimately went with a Sangoma card (A200 +HWEC module), it killed the echo and sounded great. YMMV - Firmware and hardware has probably changed quite a bit in 3 years, and your loop characteristics might be better. My inclination would be to buy one, and see how well it works at your site. > System Design. > > Extension Numbers 2xx > Outside line access 1xxxxxxxxxx > groups 3xx > auto-attendent ??? Depending on how you want your system to work - auto-attendant is a good way to go if you won't have DIDs for all or most individuals. Our receptionist's (and our client's) time was being chewed up by all calls being forced to go through her before we implemented an auto-attendant. From shouldbeq931 at googlemail.com Mon Nov 9 09:48:16 2009 From: shouldbeq931 at googlemail.com (shouldbe q931) Date: Mon, 9 Nov 2009 17:48:16 +0000 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> Message-ID: <649eaa470911090948v34ee9239l8eb98ba964d42ad7@mail.gmail.com> While I'm very happy to hear this, the wiki has in more than one place suggestions to install multiple "sound and moh" 'sets'... On Mon, Nov 9, 2009 at 5:34 PM, Michael Collins wrote: > > > On Mon, Nov 9, 2009 at 9:01 AM, Fred-145 wrote: >> >> Hello >> >> I successfully installed FreeSwitch from SVN, and am now prompted to >> install >> the sound files. Am I correct in understanding that "sounds" are >> POTS-grade >> files (8KHz?) while "cd-sounds" are closer to VoIP-grade (16KHz?), and >> "hd-sounds" and "uhd-sounds" are for Skype-grade sound files? >> >> In that case, if I only need to play sound files to POTS callers, I only >> need the "sounds" files? > > I recommend you just do this: > make cd-sounds-install && make cd-moh-install > > This will install all the sounds (8k, 16k, 32k, 48k) and thus FS can play > whatever sampling rate is necessary. I promise you that this is the easiest > way to go. Others have tried to save a few megabytes of disk space by not > installing the higher quality sounds and music and have ended up wasting > hours debugging "issues with playing sound files." > > For the record, the 8k files are just "sounds" and then you have hd (16k), > uhd (32k), and cd (48k). > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Nov 9 10:13:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 10:13:47 -0800 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <649eaa470911090948v34ee9239l8eb98ba964d42ad7@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> <649eaa470911090948v34ee9239l8eb98ba964d42ad7@mail.gmail.com> Message-ID: <87f2f3b90911091013i1f74fa87s2b9f79112e67bc48@mail.gmail.com> On Mon, Nov 9, 2009 at 9:48 AM, shouldbe q931 wrote: > While I'm very happy to hear this, the wiki has in more than one place > suggestions to install multiple "sound and moh" 'sets'... > > Link(s) please? I'll take care of the wiki. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/85e066e1/attachment-0002.html From shouldbeq931 at googlemail.com Mon Nov 9 10:28:16 2009 From: shouldbeq931 at googlemail.com (shouldbe q931) Date: Mon, 9 Nov 2009 18:28:16 +0000 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <87f2f3b90911091013i1f74fa87s2b9f79112e67bc48@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> <649eaa470911090948v34ee9239l8eb98ba964d42ad7@mail.gmail.com> <87f2f3b90911091013i1f74fa87s2b9f79112e67bc48@mail.gmail.com> Message-ID: <649eaa470911091028s202aad27x351d09b0f6925b0f@mail.gmail.com> I was sure I'd seen more, but http://wiki.freeswitch.org/wiki/Installation_Guide search for "There are also higher bitrate sounds available for download and installation with:" Cheers Arne On Mon, Nov 9, 2009 at 6:13 PM, Michael Collins wrote: > > > On Mon, Nov 9, 2009 at 9:48 AM, shouldbe q931 > wrote: >> >> While I'm very happy to hear this, the wiki has in more than one place >> suggestions to install multiple "sound and moh" 'sets'... >> > Link(s) please? I'll take care of the wiki. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Nov 9 10:32:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 10:32:03 -0800 Subject: [Freeswitch-users] Monitoring via SNMP In-Reply-To: <001001ca6124$dc9268e0$95b73aa0$@cc> References: <001001ca6124$dc9268e0$95b73aa0$@cc> Message-ID: <87f2f3b90911091032h423890a9oea02c23674e2f3f4@mail.gmail.com> 2009/11/9 Dimitar Dechev > Dear All, > > > > I couldn?t find much information about how to monitor Freeswitch via SNMP > like how many calls/legs I have, how many CAPs, and etc. One of the thing I > do currently is to make simple bash script which in general runs ?fs_cli -x > ?show calls count?? or some other command and call that script via > snmpd.conf. > > > > I would appreciate if somebody tell me if they is snmp module in > Freeswitch, or provide with link/method from where I can read some > information. > > > > Thanks! > Sorry, no SNMP. However, there is something with a little more power: the event socket. You can use the ESL (event socket library) to write your own monitor script. More info: http://wiki.freeswitch.org/wiki/Event_Socket Also, if you really wanted to, you could do the "fs_cli -x 'foo bar'" stuff but there are more elegant and powerful ways of handing this. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/41e5692d/attachment-0002.html From msc at freeswitch.org Mon Nov 9 10:32:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 10:32:32 -0800 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <649eaa470911091028s202aad27x351d09b0f6925b0f@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> <649eaa470911090948v34ee9239l8eb98ba964d42ad7@mail.gmail.com> <87f2f3b90911091013i1f74fa87s2b9f79112e67bc48@mail.gmail.com> <649eaa470911091028s202aad27x351d09b0f6925b0f@mail.gmail.com> Message-ID: <87f2f3b90911091032j4fbec27fi8d1e2dc3aa212d45@mail.gmail.com> On Mon, Nov 9, 2009 at 10:28 AM, shouldbe q931 wrote: > I was sure I'd seen more, but > http://wiki.freeswitch.org/wiki/Installation_Guide search for "There > are also higher bitrate sounds available for download and installation > with:" > > Cheers > > Arne > > Thanks! I'll clean that up a bit. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/7ed1e993/attachment-0002.html From codecomplete at free.fr Mon Nov 9 10:39:45 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 9 Nov 2009 10:39:45 -0800 (PST) Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> Message-ID: <26271417.post@talk.nabble.com> mercutioviz wrote: > > I recommend you just do this: make cd-sounds-install && make > cd-moh-install > Will do. Thanks. -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26271417.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jmesquita at freeswitch.org Mon Nov 9 10:43:36 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 9 Nov 2009 16:43:36 -0200 Subject: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6 In-Reply-To: <191c3a030911090850j4d9e7972m15fa4661a2da7926@mail.gmail.com> References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> <2742E007-4C51-4E1A-96C6-B047F82174F2@nani.ca> <191c3a030911090850j4d9e7972m15fa4661a2da7926@mail.gmail.com> Message-ID: Or write one for Mac specifically since PA is fine for all the rest (I think)? JM On Mon, Nov 9, 2009 at 2:50 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > maybe we should write a new audio abstraction lib =D > > > On Mon, Nov 9, 2009 at 2:07 AM, Bruce Fletcher wrote: > >> The patch from the PortAudio site does get the library to build, but >> it still fails with the same assertion when I try to play MOH. The >> patch I'm talking about is this one: >> >> http://www.portaudio.com/trac/changeset/1418 >> >> If the same build problem applies to other 64 bit systems, it might be >> a good idea to incorporate this patch. It looks clean and reasonable >> to me, at least. >> >> I've managed to work around the problem entirely by building >> FreeSWITCH for i386, but I'll go ask the PortAudio folks what the >> status is of their 64 bit support. They are clearly assuming 32 bit >> long integers in some places, which is hopefully on a to-fix list >> somewhere. >> >> Thanks, >> - Bruce >> >> >> On 2009-11-08, at 12:25 PM, Michael Jerris wrote: >> >> > If you can figure out a clean way for us to do this with proper ifdefs >> > in tree in a way that will not break others that would be the most >> > preferred. >> > >> > Mike >> > >> > On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote: >> > >> >> OK, I'll ignore that MacPorts patch for now and try to find a better >> >> approach. >> >> >> >> I'll look into this further tonight, but this morning I found a more >> >> recent promising patch on the PortAudio site: >> >> >> >> http://www.portaudio.com/trac/changeset/1418 >> >> >> >> It seems to push some data types to 32 bit regardless of platform, >> >> which might work better than the MacPorts approach of migrating some >> >> data structures to 64 bit. At any rate, this patch being on the >> >> PortAudio site suggests it might be a more approved fix. >> >> >> >> I'll keep plugging at this in my free time and report any significant >> >> progress back to the list. >> >> >> >> Thanks, >> >> - Bruce >> >> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/b7faf0f2/attachment-0002.html From stevendt at primrosebank.net Mon Nov 9 10:44:27 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 18:44:27 -0000 Subject: [Freeswitch-users] Cordless VOIP Phones References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> Message-ID: <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Hi, has anyone any good results to share with using cordless phones for VOIP with FreeSwitch ? I have seen a few around that appear to operate with wireless networks and make SIP connections to VOIP PBXs. I have seen various models from Engenius, Prestige, DORO and Siemens as well as Snom. regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/4db387aa/attachment-0002.html From jmesquita at freeswitch.org Mon Nov 9 10:50:02 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 9 Nov 2009 16:50:02 -0200 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: I have Siemens A58IP and Snom M3. Both work very well with pros and cons. Nonetheless, both lack HD .... JM On Mon, Nov 9, 2009 at 4:44 PM, Dave Stevenson wrote: > Hi, > > has anyone any good results to share with using cordless phones for VOIP > with FreeSwitch ? > > I have seen a few around that appear to operate with wireless networks and > make SIP connections to VOIP PBXs. > > I have seen various models from Engenius, Prestige, DORO and Siemens as > well as Snom. > > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/1eb79f3a/attachment-0002.html From djbinter at yahoo.com Mon Nov 9 11:05:43 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 9 Nov 2009 11:05:43 -0800 (PST) Subject: [Freeswitch-users] Another Concurrent Calls Monitor Question Message-ID: <158503.71880.qm@web37504.mail.mud.yahoo.com> I am also curious whether you can recommend how I can get the info if I want to see concurrent calls by account code. Let's say if I am running FS as SBC and I want to monitor concurrent calls per customer. I've looked at the HEARTBEAT, but it only gives me overall session-count. How safe is it if I have a cron job running every 5 minutes, and get the data from core.db in the calls tables. For instance, if I issue the following query: select count(*) from calls where substr(callee_chan_name,27,15)='$gw_ip_addr'; I don't want to query directly from core.db, so it would be great if I can use something from event socket to monitor per customer (account code) or ip address. Thank you. From codecomplete at free.fr Mon Nov 9 11:16:27 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 9 Nov 2009 11:16:27 -0800 (PST) Subject: [Freeswitch-users] Right way to start FS on CentOS at boot-time? Message-ID: <26272066.post@talk.nabble.com> Hello For those of you running FS on CentOS (5.4) who compiled FS from SVN, I'd like to make sure I'm doing it right to have FS start automatically at boot-time: 1. cp /usr/src/freeswitch/build/freeswitch.init.redhat /etc/init.d/freeswitch 2. vi /etc/init.d/freeswitch: PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} 3. chmod 755 /etc/init.d/freeswitch 4. chkconfig --level 345 freeswitch on 5. chkconfig --list freeswitch 6. (why needed in addition to chkconfig?) ln -s /etc/init.d/freeswitch /usr/sbin/rcfreeswitch To manually start the server: Launch the server through the rc.d script: /etc/init.d/freeswitch start Is the above correct? Thank you. -- View this message in context: http://old.nabble.com/Right-way-to-start-FS-on-CentOS-at-boot-time--tp26272066p26272066.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From bruce at nani.ca Mon Nov 9 11:23:25 2009 From: bruce at nani.ca (Bruce Fletcher) Date: Mon, 9 Nov 2009 11:23:25 -0800 Subject: [Freeswitch-users] 64 bit PortAudio status In-Reply-To: References: <7405E1CF-32C7-47D4-9711-CDC74A105CE8@nani.ca> <1129D667-E08C-40D9-B11F-052F6AA13AB0@freeswitch.org> <0AA4FC95-9FF4-4600-9B60-310FD7E0BC3F@nani.ca> <545ECBBD-0FC9-4B65-83E4-8D1305D5E14E@jerris.com> <2742E007-4C51-4E1A-96C6-B047F82174F2@nani.ca> <191c3a030911090850j4d9e7972m15fa4661a2da7926@mail.gmail.com> Message-ID: I just want to clarify the status of PortAudio on 64 bit architectures. There is a compile-time problem in pa_dither.c (and .h) that comes from the code not being 64 bit ready. This problem has been patched cleanly here: http://www.portaudio.com/trac/changeset/1418 I think this patch should go into FreeSWITCH because it makes PortAudio compilable and minimally useful on 64 bit platforms. There is a separate 64 bit issue involving PortAudio's Mac interface which shows up at runtime in the function ringBufferIOProc(). Someone marked it as a to-fix issue for 64 bit compiles with the following assertion: assert( sizeof( UInt32 ) == sizeof( long ) ); There is a discussion that just started this weekend about fixing this on the PortAudio mailing list here: http://music.columbia.edu/pipermail/portaudio/2009-November/009581.html It sounds like someone there may fix up this problem soon. I am going to keep tracking that and see if I can help it along. In any event, this problem is unrelated to the pa_dither problem. I just wanted to clarify this so the existing PortAudio patch for pa_dither could possibly be included before 1.05 is released. Thanks, - Bruce From andrew at hijacked.us Mon Nov 9 11:29:05 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 9 Nov 2009 14:29:05 -0500 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> Message-ID: <20091109192904.GI9418@hijacked.us> On Mon, Nov 09, 2009 at 08:59:54PM +0800, mark morreny wrote: > Hi, > > Does anyone know how to playback based on files from hadoop storage. > > There is a libhdcp, and java api. Is there anyway to put together a sample > middle piece to move files from hadoop to freeswitch using memory space, so > there is no disk I/O? > > Any feedback or suggestion will be greatly appreciated. > mod_shell_stream might work, if you can just spit out the raw audio to the shell. Or write another stream module that works with libhdcp. Andrew From stevendt at primrosebank.net Mon Nov 9 11:38:03 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 19:38:03 -0000 Subject: [Freeswitch-users] Cordless VOIP Phones References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: Joao, thanks for the note. The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ? Interesting about the HD, but do you notice the difference and find that you're disappointed with the quality of their sounds ? regards Dave ----- Original Message ----- From: Jo?o Mesquita To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 6:50 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones I have Siemens A58IP and Snom M3. Both work very well with pros and cons. Nonetheless, both lack HD .... JM On Mon, Nov 9, 2009 at 4:44 PM, Dave Stevenson wrote: Hi, has anyone any good results to share with using cordless phones for VOIP with FreeSwitch ? I have seen a few around that appear to operate with wireless networks and make SIP connections to VOIP PBXs. I have seen various models from Engenius, Prestige, DORO and Siemens as well as Snom. regards Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/34e10e68/attachment-0002.html From brian at freeswitch.org Mon Nov 9 12:05:07 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Nov 2009 14:05:07 -0600 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote: > Joao, > > thanks for the note. The Snom M3 is one of the ones that I was > looking at - I would be interested in the "Pro's & Cons" ? RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN > > Interesting about the HD, but do you notice the difference and find > that you're disappointed with the quality of their sounds ? Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements. > > regards > Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/98c02fb6/attachment-0002.html From stevendt at primrosebank.net Mon Nov 9 12:15:45 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 20:15:45 -0000 Subject: [Freeswitch-users] Cordless VOIP Phones References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> Message-ID: <4E593A21A8B9463FB86AAC3B5ED5A941@bp1.ad.bp.com> Hi, thanks Brian, that's interesting. I had a comment "off list" which suggested the same thing. It did not quite fit with my aspiration for an all VOIP solution, but it sounds like the technology is not quite there yet for hands-free. That's great feedback before I spend some cash on a hands-free VOIP phone regards Dave ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 8:05 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote: Joao, thanks for the note. The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ? RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN Interesting about the HD, but do you notice the difference and find that you're disappointed with the quality of their sounds ? Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements. regards Dave ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/0804c928/attachment-0002.html From rupa at rupa.com Mon Nov 9 12:19:42 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 9 Nov 2009 12:19:42 -0800 Subject: [Freeswitch-users] Another Concurrent Calls Monitor Question In-Reply-To: <158503.71880.qm@web37504.mail.mud.yahoo.com> References: <158503.71880.qm@web37504.mail.mud.yahoo.com> Message-ID: Use mod limit to do this. You can choose to use it in count only mode if you want (no limit). On Mon, Nov 9, 2009 at 11:05 AM, DJB wrote: > I am also curious whether you can recommend how I can get the info if I want to see concurrent calls by account code. ?Let's say if I am running FS as SBC and I want to monitor concurrent calls per customer. ?I've looked at the HEARTBEAT, but it only gives me overall session-count. > > How safe is it if I have a cron job running every 5 minutes, and get the data from core.db in the calls tables. ?For instance, if I issue the following query: > select count(*) from calls where substr(callee_chan_name,27,15)='$gw_ip_addr'; > > I don't want to query directly from core.db, so it would be great if I can use something from event socket to monitor per customer (account code) or ip address. > > Thank you. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From stevendt at primrosebank.net Mon Nov 9 12:22:15 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 20:22:15 -0000 Subject: [Freeswitch-users] RegEx Help References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> I **think** that the following will match any three character strings from 1xx to 399 I want to exclude 100 though, can anyone help me with the required RegEx please ? ^([1-3][0-9][0-9])$ I could (I think) do ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ But it does not "feel" elegant - is there a better way ? regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/e179c576/attachment-0002.html From rupa at rupa.com Mon Nov 9 12:23:05 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 9 Nov 2009 12:23:05 -0800 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> References: <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> Message-ID: I agree about the M3. I have the handset and it is not ergonomic at all. I also have a Siemens A580-IP. It does do G722 but has a few bugs related to G722 that I normally run it with G711 only. There is a quality difference between G722 and G711 when talking among the A580 handsets or the handset and my Polycom 450. Of the DECT handsets, I'd recommend the Siemens. It is also very inexpensive. Beware it has a 3 concurrent calls limit per base-station. On Mon, Nov 9, 2009 at 12:05 PM, Brian West wrote: > > On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote: > > Joao, > > thanks for the note. The Snom M3 is one of the ones that I was looking at - > I would be interested in the "Pro's & Cons" ? > > RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN > > > Interesting about the HD, but do you?notice the difference and find that > you're disappointed with the quality of their sounds ? > > Get an ATA with a Dect handset it works much better... the Snom M3 and the > Aastra are one in the same and they both do not live up to the quality or > usability requirements. > > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From hads at nice.net.nz Mon Nov 9 12:26:41 2009 From: hads at nice.net.nz (Hadley Rich) Date: Tue, 10 Nov 2009 09:26:41 +1300 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> Message-ID: <1257798401.10738.18.camel@sodium> On Mon, 2009-11-09 at 14:05 -0600, Brian West wrote: > Get an ATA with a Dect handset it works much better... the Snom M3 and > the Aastra are one in the same and they both do not live up to the > quality or usability requirements. That said, they are better than what else is around. I'd call them average. Nothing to write home about but you don't need to run away from them. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier From stevecrozz at gmail.com Mon Nov 9 12:37:15 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 9 Nov 2009 12:37:15 -0800 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> Message-ID: <11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> Would something like this work for you? --Stephen On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson wrote: > I **think** that the following will match any three character strings > from 1xx to 399 > > I want to exclude 100 though, can anyone help me with the required RegEx > please ? > > > ^([1-3][0-9][0-9])$ > > I could (I think) do > > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ > > But it does not "feel" elegant - is there a better way ? > > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/b97dab90/attachment-0002.html From msc at freeswitch.org Mon Nov 9 12:38:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 12:38:17 -0800 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> Message-ID: <87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson wrote: > I **think** that the following will match any three character strings > from 1xx to 399 > > I want to exclude 100 though, can anyone help me with the required RegEx > please ? > > > ^([1-3][0-9][0-9])$ > > I could (I think) do > > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ > > But it does not "feel" elegant - is there a better way ? > > expression="^[1-3]\d\d$" will match 100 to 399 -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/a48f1d30/attachment-0002.html From stevendt at primrosebank.net Mon Nov 9 12:55:30 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 20:55:30 -0000 Subject: [Freeswitch-users] RegEx Help References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> Message-ID: <6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com> Hi Stephen, thanks for the reply. I'm not sure , does the code below handle all number from 101 to 399 ? It would rely on the 100 code being picked up by the dialplan before the other extensions were processed so the order of the code in the dialplan is significant. Is that how people normally write their code, i.e., the extension processing is position dependant in the file ? regards Dave ----- Original Message ----- From: Stephen Crosby To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 8:37 PM Subject: Re: [Freeswitch-users] RegEx Help Would something like this work for you? --Stephen On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson wrote: I **think** that the following will match any three character strings from 1xx to 399 I want to exclude 100 though, can anyone help me with the required RegEx please ? ^([1-3][0-9][0-9])$ I could (I think) do ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ But it does not "feel" elegant - is there a better way ? regards Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/a3899507/attachment-0002.html From stevendt at primrosebank.net Mon Nov 9 12:55:34 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 20:55:34 -0000 Subject: [Freeswitch-users] RegEx Help References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> Message-ID: <56D7DCE6AE1844A3B47AE8E351733B9C@bp1.ad.bp.com> Thanks Michael, but I want to exclude 100 ? regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 8:38 PM Subject: Re: [Freeswitch-users] RegEx Help On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson wrote: I **think** that the following will match any three character strings from 1xx to 399 I want to exclude 100 though, can anyone help me with the required RegEx please ? ^([1-3][0-9][0-9])$ I could (I think) do ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ But it does not "feel" elegant - is there a better way ? expression="^[1-3]\d\d$" will match 100 to 399 -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/9d936bfc/attachment-0002.html From federico.omoto at gmail.com Mon Nov 9 11:50:14 2009 From: federico.omoto at gmail.com (Fede) Date: Mon, 9 Nov 2009 17:50:14 -0200 Subject: [Freeswitch-users] Doddle Web SIP phone Message-ID: <8b4221f20911091150k7f3d01eem3c5eae845158c050@mail.gmail.com> Hi! I'm trying the Doodle web SIP phone but for some reason I'm unable to register to my FreeSWITCH server. I've tried with other servers and it works ok. Did someone tried this web phone with FreeSWITCH? Any tips why it doesn't authenticate? Thank you! Federico Omoto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/cf7aa011/attachment-0002.html From frank at carmickle.com Mon Nov 9 13:03:43 2009 From: frank at carmickle.com (Frank Carmickle) Date: Mon, 9 Nov 2009 16:03:43 -0500 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> References: <4AF4AF73.8070804@tx.rr.com> <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> Message-ID: <20091109210343.GH11697@base.carmickle.com> On Mon, Nov 09, Michael Collins wrote: > On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson > wrote: > > > I **think** that the following will match any three character strings > > from 1xx to 399 > > > > I want to exclude 100 though, can anyone help me with the required RegEx > > please ? > > > > > > ^([1-3][0-9][0-9])$ > > > > I could (I think) do > > > > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ You mean (^1[0-9][1-9]$|^[2-3]\d\d$) --FC From stevendt at primrosebank.net Mon Nov 9 13:05:40 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 21:05:40 -0000 Subject: [Freeswitch-users] Cordless VOIP Phones References: <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> Message-ID: <90CFC4A7AD444A9CA63B3FAF00744E5C@bp1.ad.bp.com> Hi Rupa, thanks for the tip. I've had a look for the A580-PI - as you say, quite inexpensive and probably worth taking a chance on one. regards Dave ----- Original Message ----- From: "Rupa Schomaker" To: Sent: Monday, November 09, 2009 8:23 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones I agree about the M3. I have the handset and it is not ergonomic at all. I also have a Siemens A580-IP. It does do G722 but has a few bugs related to G722 that I normally run it with G711 only. There is a quality difference between G722 and G711 when talking among the A580 handsets or the handset and my Polycom 450. Of the DECT handsets, I'd recommend the Siemens. It is also very inexpensive. Beware it has a 3 concurrent calls limit per base-station. On Mon, Nov 9, 2009 at 12:05 PM, Brian West wrote: > > On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote: > > Joao, > > thanks for the note. The Snom M3 is one of the ones that I was looking > at - > I would be interested in the "Pro's & Cons" ? > > RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN > > > Interesting about the HD, but do you notice the difference and find that > you're disappointed with the quality of their sounds ? > > Get an ATA with a Dect handset it works much better... the Snom M3 and the > Aastra are one in the same and they both do not live up to the quality or > usability requirements. > > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From stevendt at primrosebank.net Mon Nov 9 13:09:15 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 21:09:15 -0000 Subject: [Freeswitch-users] RegEx Help References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com><87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> <20091109210343.GH11697@base.carmickle.com> Message-ID: <9D87DD1FCD5A45518D76F9A70F001F96@bp1.ad.bp.com> Hi Frank Yup ! That's what I mean :-) thanks a lot, regards Dave ----- Original Message ----- From: "Frank Carmickle" To: Sent: Monday, November 09, 2009 9:03 PM Subject: Re: [Freeswitch-users] RegEx Help > On Mon, Nov 09, Michael Collins wrote: >> On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson >> wrote: >> >> > I **think** that the following will match any three character strings >> > from 1xx to 399 >> > >> > I want to exclude 100 though, can anyone help me with the required >> > RegEx >> > please ? >> > >> > >> > ^([1-3][0-9][0-9])$ >> > >> > I could (I think) do >> > >> > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ > > You mean > > (^1[0-9][1-9]$|^[2-3]\d\d$) > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Russell.Mosemann at cune.org Mon Nov 9 13:10:25 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Mon, 9 Nov 2009 21:10:25 -0000 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> Message-ID: <20091109211025.ACC7B2E3DD4@mail.cune.org> Dave Stevenson said: > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ Another possibility. ^(1(0[1-9]|[1-9]\d)|[2-3]\d{2}) -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From stevecrozz at gmail.com Mon Nov 9 13:15:56 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 9 Nov 2009 13:15:56 -0800 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <20091109210343.GH11697@base.carmickle.com> References: <4AF4AF73.8070804@tx.rr.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <87f2f3b90911091238q1f4e83f6xcf4d8cf028b397e2@mail.gmail.com> <20091109210343.GH11697@base.carmickle.com> Message-ID: <11990ade0911091315l6db2eb04v4d1a472d2b9b8b10@mail.gmail.com> Dave, I think extensions are processed in order although I can't quickly find any documentation that says this, why don't you try it and see, it would take only a moment to find out for sure. --Stephen On Mon, Nov 9, 2009 at 1:03 PM, Frank Carmickle wrote: > On Mon, Nov 09, Michael Collins wrote: > > On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson > > wrote: > > > > > I **think** that the following will match any three character strings > > > from 1xx to 399 > > > > > > I want to exclude 100 though, can anyone help me with the required > RegEx > > > please ? > > > > > > > > > ^([1-3][0-9][0-9])$ > > > > > > I could (I think) do > > > > > > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ > > You mean > > (^1[0-9][1-9]$|^[2-3]\d\d$) > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/d85d1346/attachment-0002.html From msc at freeswitch.org Mon Nov 9 13:15:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 13:15:56 -0800 Subject: [Freeswitch-users] Doddle Web SIP phone In-Reply-To: <8b4221f20911091150k7f3d01eem3c5eae845158c050@mail.gmail.com> References: <8b4221f20911091150k7f3d01eem3c5eae845158c050@mail.gmail.com> Message-ID: <87f2f3b90911091315o722ae1c4xdc6728e251f4f7b2@mail.gmail.com> On Mon, Nov 9, 2009 at 11:50 AM, Fede wrote: > Hi! > > I'm trying the Doodle web SIP phone but for some reason I'm unable to > register to my FreeSWITCH server. I've tried with other servers and it works > ok. > Did someone tried this web phone with FreeSWITCH? Any tips why it doesn't > authenticate? > Can you capture the debug log from the command line? It would also be good to have a SIP trace. More information on gathering info and putting it in pastebin can be found here: http://wiki.freeswitch.org/wiki/Reporting_Bugs Also, be sure that you are using the latest version of FreeSWITCH, preferably SVN trunk. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/dc67cecd/attachment-0002.html From msc at freeswitch.org Mon Nov 9 13:20:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 13:20:59 -0800 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <20091109211025.ACC7B2E3DD4@mail.cune.org> References: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <20091109211025.ACC7B2E3DD4@mail.cune.org> Message-ID: <87f2f3b90911091320g79a6da8eo48b2b572322f8eb2@mail.gmail.com> On Mon, Nov 9, 2009 at 1:10 PM, wrote: > Dave Stevenson said: > > > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ > > Another possibility. > > ^(1(0[1-9]|[1-9]\d)|[2-3]\d{2}) > > Yep this is the one. I'm sorry I didn't read the OP correctly the first time. Skipping 100 and matching 101 is the tricky part, obviously. This regex should fit the bill. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/bd29c30b/attachment-0002.html From mattdfong at gmail.com Mon Nov 9 13:41:01 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 10 Nov 2009 04:41:01 +0700 Subject: [Freeswitch-users] Doddle Web SIP phone In-Reply-To: <87f2f3b90911091315o722ae1c4xdc6728e251f4f7b2@mail.gmail.com> References: <8b4221f20911091150k7f3d01eem3c5eae845158c050@mail.gmail.com> <87f2f3b90911091315o722ae1c4xdc6728e251f4f7b2@mail.gmail.com> Message-ID: <4256bf830911091341w1bfb2dafx818dd4d4f18248ec@mail.gmail.com> I just tried the webphone with my freeswitch server and it worked fine, making a call to my echo test w/o any issues...so it's probably a configuration issue with freeswitch. --matt http://www.hellohunter.com On Tue, Nov 10, 2009 at 4:15 AM, Michael Collins wrote: > > > On Mon, Nov 9, 2009 at 11:50 AM, Fede wrote: > >> Hi! >> >> I'm trying the Doodle web SIP phone but for some reason I'm unable to >> register to my FreeSWITCH server. I've tried with other servers and it works >> ok. >> Did someone tried this web phone with FreeSWITCH? Any tips why it doesn't >> authenticate? >> > > Can you capture the debug log from the command line? It would also be good > to have a SIP trace. More information on gathering info and putting it in > pastebin can be found here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Also, be sure that you are using the latest version of FreeSWITCH, > preferably SVN trunk. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/ed0488ee/attachment-0002.html From federico.omoto at gmail.com Mon Nov 9 13:46:30 2009 From: federico.omoto at gmail.com (Fede) Date: Mon, 9 Nov 2009 19:46:30 -0200 Subject: [Freeswitch-users] Doddle Web SIP phone In-Reply-To: <87f2f3b90911091315o722ae1c4xdc6728e251f4f7b2@mail.gmail.com> References: <8b4221f20911091150k7f3d01eem3c5eae845158c050@mail.gmail.com> <87f2f3b90911091315o722ae1c4xdc6728e251f4f7b2@mail.gmail.com> Message-ID: <8b4221f20911091346y6181f8b1o6e6c4d88ca81eb94@mail.gmail.com> Hi Michael! Thank you for your quicky answer. I'm using FreeSWITCH 1.0.5 pre5. The debug log from the command line plus the SIP trace are at: http://pastebin.freeswitch.org/11043 The Doddle web phone is at: http://www.doddlephone.com You can test this account at my FreeSWITCH server at: -server: 216.75.60.102 -username: doddle -password: doddle Thank you! Federico Omoto On Mon, Nov 9, 2009 at 7:15 PM, Michael Collins wrote: > > > On Mon, Nov 9, 2009 at 11:50 AM, Fede wrote: > >> Hi! >> >> I'm trying the Doodle web SIP phone but for some reason I'm unable to >> register to my FreeSWITCH server. I've tried with other servers and it works >> ok. >> Did someone tried this web phone with FreeSWITCH? Any tips why it doesn't >> authenticate? >> > > Can you capture the debug log from the command line? It would also be good > to have a SIP trace. More information on gathering info and putting it in > pastebin can be found here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Also, be sure that you are using the latest version of FreeSWITCH, > preferably SVN trunk. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/a8eeec25/attachment-0002.html From JCasale at activenetwerx.com Mon Nov 9 14:46:18 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 9 Nov 2009 22:46:18 +0000 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: >The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ? Worst POS I have ever used, from a sound quality to ergonomics pov, tech support was as bad... I have Aastra 480i CT's which work well. jlc From stevendt at primrosebank.net Mon Nov 9 14:56:10 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 22:56:10 -0000 Subject: [Freeswitch-users] DIalplan logic References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com><11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> <6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com> Message-ID: <77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox set the timeout for the call Added these lines - but don't know why - they are in the default extension code ????? then go to voice mail on 100 giving The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/78c113b9/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 9 15:01:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 17:01:13 -0600 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com> <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> asstra has one issue where if you look at them wrong they start telling the server that the media ip is 0.0.0.0 which we have never identified but they indeed seem to work better than snom m3 On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale wrote: > >The Snom M3 is one of the ones that I was looking at - I would be > interested in the "Pro's & Cons" ? > > Worst POS I have ever used, from a sound quality to ergonomics pov, tech > support was as bad... > > I have Aastra 480i CT's which work well. > > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/9fca3e43/attachment-0002.html From stevendt at primrosebank.net Mon Nov 9 15:01:13 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 23:01:13 -0000 Subject: [Freeswitch-users] Cordless VOIP Phones References: <4AF4AF73.8070804@tx.rr.com><5C69DE1704EC4BE8AA4D26CC116F0B55@bp1.ad.bp.com><6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org><2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> Message-ID: <908AF2F7F29D44CEA03C12A60C6CB298@bp1.ad.bp.com> Thanks - pretty unambiguous reply ! I won't go down that route then :-) ----- Original Message ----- From: "Joseph L. Casale" To: Sent: Monday, November 09, 2009 10:46 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones > >The Snom M3 is one of the ones that I was looking at - I would be > >interested in the "Pro's & Cons" ? > > Worst POS I have ever used, from a sound quality to ergonomics pov, tech > support was as bad... > > I have Aastra 480i CT's which work well. > > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Mon Nov 9 15:04:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 17:04:13 -0600 Subject: [Freeswitch-users] RegEx Help In-Reply-To: <87f2f3b90911091320g79a6da8eo48b2b572322f8eb2@mail.gmail.com> References: <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <20091109211025.ACC7B2E3DD4@mail.cune.org> <87f2f3b90911091320g79a6da8eo48b2b572322f8eb2@mail.gmail.com> Message-ID: <191c3a030911091504s675f8e79vafffea36130e20c5@mail.gmail.com> If the global var "auto_hunt" is "true" the xml dialplan will try to find an extension where the "name" param matches the destination number. This is not the default The default is to try them in order from top to bottom. On Mon, Nov 9, 2009 at 3:20 PM, Michael Collins wrote: > > > On Mon, Nov 9, 2009 at 1:10 PM, wrote: > >> Dave Stevenson said: >> >> > ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ >> >> Another possibility. >> >> ^(1(0[1-9]|[1-9]\d)|[2-3]\d{2}) >> >> > Yep this is the one. I'm sorry I didn't read the OP correctly the first > time. Skipping 100 and matching 101 is the tricky part, obviously. This > regex should fit the bill. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/77405a75/attachment-0002.html From jmesquita at freeswitch.org Mon Nov 9 15:16:37 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 9 Nov 2009 21:16:37 -0200 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> References: <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> Message-ID: Beat me with a dead cat all you want but I rather the snom m3 than the Siemens A580IP.... Siemens has very low volume which makes its call quality suck despite of being ergonomic and all... That gigaset application sucks and the base station is slow as hell... Maybe I have a bad unit? The snom m3 has its downsides, but all and all, I am happy with the phone if you consider its price tag here in South America where a Polycom can easily cost over 200USD the cheapest unit. Regards, JM On Mon, Nov 9, 2009 at 9:01 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > asstra has one issue where if you look at them wrong they start telling the > server that the media ip is 0.0.0.0 which we have never identified but they > indeed seem to work better than snom m3 > > > > On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale < > JCasale at activenetwerx.com> wrote: > >> >The Snom M3 is one of the ones that I was looking at - I would be >> interested in the "Pro's & Cons" ? >> >> Worst POS I have ever used, from a sound quality to ergonomics pov, tech >> support was as bad... >> >> I have Aastra 480i CT's which work well. >> >> jlc >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/88edc5fd/attachment-0002.html From msc at freeswitch.org Mon Nov 9 15:26:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 15:26:25 -0800 Subject: [Freeswitch-users] DIalplan logic In-Reply-To: <77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> References: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> <6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com> <77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> Message-ID: <87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com> See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: > Hi Guys, > > OK, with the RegEx help that you gave me, I have separated out the > processing of extension 100 from 101 to 399 as I wanted. > > I have created a group (100) which contains a number of phones - 101 to 105 > at the moment. > > When the PSTN line rings, I want all the extensions in the group to ring - > that's the easy bit (I think - it's a copy of extension 2000 code) > > > > > > > > That's fine and the nominated phones all ring. > > I'm struggling to get it to do what I want when some doesn't pick up > though. > > All extensions ring as required, but their own dialplan entries (copies of > the 1001 to 1005 code in the default dialplan) don't answer the call. That's > fine, as you would not want every extension's voice mail to kick in. > > What I want to happen is for extension 100's voice mail to kick in after a > time delay. > > So, get the dialed exetension number so that I can point at the right > mailbox > > > set the timeout for the call > > > Added these lines - but don't know why - they are in the default extension > code ????? > > > > > then go to voice mail on 100 > > > > > giving > > > The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. > > > > > > > > > > I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" > > > > > The voicemail kicks in, and prompts are correct (although the extension > name is not spoken) but the wav file is saved in the 1001 directory not 100 > and neither extension 100 or 1001 think that have any voice mail messages. > > Can someone help please ? > > Where am i going wrong ? > > Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/3389c076/attachment-0002.html From srinivas.ksvreddy at gmail.com Mon Nov 9 15:36:14 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Mon, 9 Nov 2009 18:36:14 -0500 Subject: [Freeswitch-users] Request: Notify sip messages from Freeswitch to UserAgent Message-ID: Hi, >From Freeswitch there is continuously Request: Notify (Messages-waiting) requests are comming, i didnt subscribe from Freeswith and pjsip(ua). any body know how to stop those requests from Freeswitch. Thanks-- Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/6254aa14/attachment-0002.html From stevendt at primrosebank.net Mon Nov 9 15:52:44 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 9 Nov 2009 23:52:44 -0000 Subject: [Freeswitch-users] DIalplan logic References: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com><11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com><6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com><77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> <87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com> Message-ID: Michael, thanks a lot - it's fixed...... you spotted exactly what the problem was ! regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 11:26 PM Subject: Re: [Freeswitch-users] DIalplan logic See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox set the timeout for the call Added these lines - but don't know why - they are in the default extension code ????? then go to voice mail on 100 giving The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/61cbd198/attachment-0002.html From brian at freeswitch.org Mon Nov 9 16:03:14 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Nov 2009 18:03:14 -0600 Subject: [Freeswitch-users] Request: Notify sip messages from Freeswitch to UserAgent In-Reply-To: References: Message-ID: <3EBF2A1F-122E-4116-B0AE-989C9268B1D5@freeswitch.org> gratuitous notifies are what they are called and I think their is a patch on jira with that functionality. I would have to dig thru jira to double check... I think Moc wrote the patch. /b On Nov 9, 2009, at 5:36 PM, srinivasula reddy wrote: > Hi, > > From Freeswitch there is continuously Request: Notify (Messages- > waiting) requests are comming, i didnt subscribe from Freeswith and > pjsip(ua). > any body know how to stop those requests from Freeswitch. > > Thanks-- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Nov 9 16:09:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 18:09:07 -0600 Subject: [Freeswitch-users] Request: Notify sip messages from Freeswitch to UserAgent In-Reply-To: References: Message-ID: <191c3a030911091609u4e809fe6r5858ec3d1f917adf@mail.gmail.com> Add that to your sofia profile. You must be new to SIP, you will soon learn that almost every SIP device just stupidly expects you to send this and never does it the correct way by subscribing to it which is why this option is the default. On Mon, Nov 9, 2009 at 5:36 PM, srinivasula reddy < srinivas.ksvreddy at gmail.com> wrote: > Hi, > > From Freeswitch there is continuously Request: Notify (Messages-waiting) > requests are comming, i didnt subscribe from Freeswith and pjsip(ua). > any body know how to stop those requests from Freeswitch. > > Thanks-- > Srinivasula Reddy K > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/031a5e96/attachment-0002.html From stevendt at primrosebank.net Mon Nov 9 16:18:15 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 10 Nov 2009 00:18:15 -0000 Subject: [Freeswitch-users] DIalplan logic References: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com><0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com><2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com><11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com><6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com><77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com><87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com> Message-ID: <054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> Well, I thought it was fixed - it is more or less working, with one more stumbling block. I have just posted a dump to the pastebin - from Dave (stevendt) The voice mail works - but too well. If the call is answered by a someone at this end - everything is fine until the user hangs up, then the remote party gets the voicemail messages.# Is there something else wrong with the dialplan logic below ? regards Dave ----- Original Message ----- From: Dave Stevenson To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 11:52 PM Subject: Re: [Freeswitch-users] DIalplan logic Michael, thanks a lot - it's fixed...... you spotted exactly what the problem was ! regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 11:26 PM Subject: Re: [Freeswitch-users] DIalplan logic See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox set the timeout for the call Added these lines - but don't know why - they are in the default extension code ????? then go to voice mail on 100 giving The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/a4515cf4/attachment-0002.html From mike at jerris.com Mon Nov 9 16:19:10 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Nov 2009 19:19:10 -0500 Subject: [Freeswitch-users] Request: Notify sip messages from Freeswitch to UserAgent In-Reply-To: References: Message-ID: I have asked you before to please not cross post to both mailing lists. Please refrain from this in the future. Mike On Nov 9, 2009, at 6:36 PM, srinivasula reddy wrote: > Hi, > > From Freeswitch there is continuously Request: Notify (Messages- > waiting) requests are comming, i didnt subscribe from Freeswith and > pjsip(ua). > any body know how to stop those requests from Freeswitch. > > Thanks-- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Nov 9 16:29:24 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Nov 2009 18:29:24 -0600 Subject: [Freeswitch-users] DIalplan logic In-Reply-To: <054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> References: <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> <6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com> <77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> <87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com> <054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> Message-ID: <191c3a030911091629g584b4519nb8e555bbd38ff7b3@mail.gmail.com> You set both hangup_after_bridge and continue_on_fail after you already called bridge. Try setting it *before* Seems to be a running theme here that things will be parsed in a linear fashion that you may want to take note of. On Mon, Nov 9, 2009 at 6:18 PM, Dave Stevenson wrote: > Well, > > I thought it was fixed - it is more or less working, with one more > stumbling block. > > I have just posted a dump to the pastebin - from Dave (stevendt) > > The voice mail works - but too well. > > If the call is answered by a someone at this end - everything is fine until > the user hangs up, then the remote party gets the voicemail messages.# > > Is there something else wrong with the dialplan logic below ? > > regards > Dave > > > > ----- Original Message ----- > *From:* Dave Stevenson > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, November 09, 2009 11:52 PM > *Subject:* Re: [Freeswitch-users] DIalplan logic > > Michael, > > thanks a lot - it's fixed...... > > > you spotted exactly what the problem was ! > > > > > > > regards > Dave > > > > > ----- Original Message ----- > *From:* Michael Collins > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, November 09, 2009 11:26 PM > *Subject:* Re: [Freeswitch-users] DIalplan logic > > See comment inline > > On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: > >> Hi Guys, >> >> OK, with the RegEx help that you gave me, I have separated out the >> processing of extension 100 from 101 to 399 as I wanted. >> >> I have created a group (100) which contains a number of phones - 101 to >> 105 at the moment. >> >> When the PSTN line rings, I want all the extensions in the group to ring - >> that's the easy bit (I think - it's a copy of extension 2000 code) >> >> >> >> > >> >> >> That's fine and the nominated phones all ring. >> >> I'm struggling to get it to do what I want when some doesn't pick up >> though. >> >> All extensions ring as required, but their own dialplan entries (copies of >> the 1001 to 1005 code in the default dialplan) don't answer the call. That's >> fine, as you would not want every extension's voice mail to kick in. >> >> What I want to happen is for extension 100's voice mail to kick in after a >> time delay. >> >> So, get the dialed exetension number so that I can point at the right >> mailbox >> >> >> set the timeout for the call >> >> >> Added these lines - but don't know why - they are in the default extension >> code ????? >> >> >> >> >> then go to voice mail on 100 >> >> >> >> >> giving >> >> >> > > The following line needs to have 100 in parens like this: "^(100)$" because > that's how you get $1 to be populated. > >> >> >> >> > >> >> >> >> >> > > I think this might be a typo? Shouldn't this next line be ... data="default > ${domain_name} ${dialed_extension}" > >> >> >> >> >> The voicemail kicks in, and prompts are correct (although the extension >> name is not spoken) but the wav file is saved in the 1001 directory not 100 >> and neither extension 100 or 1001 think that have any voice mail messages. >> >> Can someone help please ? >> >> Where am i going wrong ? >> >> > > Make those changes, reloadxml, and then try again. Be sure to capture a > debug log if it doesn't work and put that log in pastebin.freeswitch.org. > -MC > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/5a7a0b28/attachment-0002.html From msc at freeswitch.org Mon Nov 9 16:32:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Nov 2009 16:32:14 -0800 Subject: [Freeswitch-users] DIalplan logic In-Reply-To: <054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> References: <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com> <11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com> <6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com> <77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com> <87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com> <054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> Message-ID: <87f2f3b90911091632v66c5678dg3c3120115bda63dc@mail.gmail.com> Oops, you've got some lines that are in the wrong place: Those lines need to come prior to the bridge call or they'll never be applied. :) -MC On Mon, Nov 9, 2009 at 4:18 PM, Dave Stevenson wrote: > Well, > > I thought it was fixed - it is more or less working, with one more > stumbling block. > > I have just posted a dump to the pastebin - from Dave (stevendt) > > The voice mail works - but too well. > > If the call is answered by a someone at this end - everything is fine until > the user hangs up, then the remote party gets the voicemail messages.# > > Is there something else wrong with the dialplan logic below ? > > regards > Dave > > > > ----- Original Message ----- > *From:* Dave Stevenson > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, November 09, 2009 11:52 PM > *Subject:* Re: [Freeswitch-users] DIalplan logic > > Michael, > > thanks a lot - it's fixed...... > > > you spotted exactly what the problem was ! > > > > > > > regards > Dave > > > > > ----- Original Message ----- > *From:* Michael Collins > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, November 09, 2009 11:26 PM > *Subject:* Re: [Freeswitch-users] DIalplan logic > > See comment inline > > On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: > >> Hi Guys, >> >> OK, with the RegEx help that you gave me, I have separated out the >> processing of extension 100 from 101 to 399 as I wanted. >> >> I have created a group (100) which contains a number of phones - 101 to >> 105 at the moment. >> >> When the PSTN line rings, I want all the extensions in the group to ring - >> that's the easy bit (I think - it's a copy of extension 2000 code) >> >> >> >> > >> >> >> That's fine and the nominated phones all ring. >> >> I'm struggling to get it to do what I want when some doesn't pick up >> though. >> >> All extensions ring as required, but their own dialplan entries (copies of >> the 1001 to 1005 code in the default dialplan) don't answer the call. That's >> fine, as you would not want every extension's voice mail to kick in. >> >> What I want to happen is for extension 100's voice mail to kick in after a >> time delay. >> >> So, get the dialed exetension number so that I can point at the right >> mailbox >> >> >> set the timeout for the call >> >> >> Added these lines - but don't know why - they are in the default extension >> code ????? >> >> >> >> >> then go to voice mail on 100 >> >> >> >> >> giving >> >> >> > > The following line needs to have 100 in parens like this: "^(100)$" because > that's how you get $1 to be populated. > >> >> >> >> > >> >> >> >> >> > > I think this might be a typo? Shouldn't this next line be ... data="default > ${domain_name} ${dialed_extension}" > >> >> >> >> >> The voicemail kicks in, and prompts are correct (although the extension >> name is not spoken) but the wav file is saved in the 1001 directory not 100 >> and neither extension 100 or 1001 think that have any voice mail messages. >> >> Can someone help please ? >> >> Where am i going wrong ? >> >> > > Make those changes, reloadxml, and then try again. Be sure to capture a > debug log if it doesn't work and put that log in pastebin.freeswitch.org. > -MC > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/513dd73c/attachment-0002.html From stevendt at primrosebank.net Mon Nov 9 16:42:14 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 10 Nov 2009 00:42:14 -0000 Subject: [Freeswitch-users] DIalplan logic References: <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com><6E5741081C4040DD9E5A3A8DC5408F35@bp1.ad.bp.com><11990ade0911091237n2b1ea4d2ke06921f21438d6ad@mail.gmail.com><6309D1D0245B430BA571F625B7FF1444@bp1.ad.bp.com><77B398DF7E35442DBA91AB12C5391DE1@bp1.ad.bp.com><87f2f3b90911091526o61b37c35o39832666fb06f48d@mail.gmail.com><054F4B4D36A6490F9806668CE50E72EF@bp1.ad.bp.com> <191c3a030911091629g584b4519nb8e555bbd38ff7b3@mail.gmail.com> Message-ID: <160223E6251F4D24B6D9A50362204BA3@bp1.ad.bp.com> Thanks Anthony that did the trick ! Excuse my ignorance - this is all new to me . . . It would help if I knew what I was doing, as I commented below, I copied the from the code for the default extensions (in the wrong order though obviously), without understanding what they were there for ! I've just stumbled across the Wiki page http://wiki.freeswitch.org/wiki/Extension_Status_Example - hopefully, I understand now regards Dave ----- Original Message ----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 10, 2009 12:29 AM Subject: Re: [Freeswitch-users] DIalplan logic You set both hangup_after_bridge and continue_on_fail after you already called bridge. Try setting it *before* Seems to be a running theme here that things will be parsed in a linear fashion that you may want to take note of. On Mon, Nov 9, 2009 at 6:18 PM, Dave Stevenson wrote: Well, I thought it was fixed - it is more or less working, with one more stumbling block. I have just posted a dump to the pastebin - from Dave (stevendt) The voice mail works - but too well. If the call is answered by a someone at this end - everything is fine until the user hangs up, then the remote party gets the voicemail messages.# Is there something else wrong with the dialplan logic below ? regards Dave ----- Original Message ----- From: Dave Stevenson To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 11:52 PM Subject: Re: [Freeswitch-users] DIalplan logic Michael, thanks a lot - it's fixed...... you spotted exactly what the problem was ! regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Monday, November 09, 2009 11:26 PM Subject: Re: [Freeswitch-users] DIalplan logic See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson wrote: Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox set the timeout for the call Added these lines - but don't know why - they are in the default extension code ????? then go to voice mail on 100 giving The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC ------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/63e062d0/attachment-0002.html From gshfreesw at gmail.com Mon Nov 9 18:54:15 2009 From: gshfreesw at gmail.com (Shameem Shiek) Date: Mon, 9 Nov 2009 21:54:15 -0500 Subject: [Freeswitch-users] Mod Voicemail, Is it just for registered users? Message-ID: <5070fcbd0911091854t3d359deam6170d27fa426d9ab@mail.gmail.com> Dear Freeswitch users, I am building an app where the extensions map to external callers and there are no registered users. For example, the extension 1001 would map to an external number. In that case, does it make sense to use the Mod voicemail or should I build a voicemail solution using dialplan/commands ? I need a voicemail solution to email voice messages. Thanks in advance for your input. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/bca3fdeb/attachment-0002.html From mike at jerris.com Mon Nov 9 19:07:37 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Nov 2009 22:07:37 -0500 Subject: [Freeswitch-users] Mod Voicemail, Is it just for registered users? In-Reply-To: <5070fcbd0911091854t3d359deam6170d27fa426d9ab@mail.gmail.com> References: <5070fcbd0911091854t3d359deam6170d27fa426d9ab@mail.gmail.com> Message-ID: <06537308-F9FE-4A16-8F0F-E1491AA48C40@jerris.com> registration has nothing at all to do with mod_voicemail. It should work fine. On Nov 9, 2009, at 9:54 PM, Shameem Shiek wrote: > Dear Freeswitch users, > > I am building an app where the extensions map to external callers > and there are no registered users. For example, the extension 1001 > would map to an external number. In that case, does it make sense to > use the Mod voicemail or should I build a voicemail solution using > dialplan/commands ? I need a voicemail solution to email voice > messages. > From ujjval at simplesignal.com Mon Nov 9 21:08:56 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Mon, 9 Nov 2009 21:08:56 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> , <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com> <3C04B27FC880044F8FCD735D0D952FF71702E84BF7@EXMBXCLUS01.citservers.local> <28FF3073-BFC0-4DD1-9AE8-3ACCD94B12DA@freeswitch.org> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175C650F1D@EXMBXCLUS01.citservers.local> OK, I may have solved this mystery, if I use application=answer and answer the call before the IVR which then flows into the Conference app, DTMF works from the AT&T phone.. So, if you face issues with Conferencing/IVR, answer the call before you invoke those apps... Problem I have now is that a Polycom old phone like 501s are not doing DTMF 2833 to the Freeswitch server...has anyone seen this..Call is not going thru PSTN...its IP to IP Polycom 501 through our SBC to the Freeswitch Server. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, November 02, 2009 9:08 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator you know I have heard this before... It seems to ONLY be AT&T /b On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote: > Yes, I think I did. However here is what furthur testing revelas. If > I dial in from AT&T cell phone, I do not see any DTMF using Don's > IVR.xml.conf to call my conf app. But when I dial the same number > using a Verizon Cell, it works. > > When I dial a number that is provisioned to call the Conf App > directly from the public.xml dialplan...it works even with the same > AT&T cell phone... > > Strange behaviour _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From paul.thirumalai at gmail.com Mon Nov 9 21:37:41 2009 From: paul.thirumalai at gmail.com (Paul Thirumalai) Date: Mon, 9 Nov 2009 21:37:41 -0800 Subject: [Freeswitch-users] Configuring freeswitch with voicepulse Message-ID: <900c9adf0911092137vf45ec94ie7473d2c08e5ae12@mail.gmail.com> Hello All I am trying to configure freeswitch so that it sends outgoing calls to the PSTN through voicepulse My configuration is as follows. I created a file $PREFIX/conf/sip_profiles/external/voicepulse.xml I also have a dial plan defined as follows When I dial an external number using extension 1000 I get the following message on the CLI ] freeswitch at ubuntu> 2009-11-10 00:35:44.365614 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1000 at 74.207.249.79[e4301180-cdba-11de-a864-8927fe94a9f0] 2009-11-10 00:35:44.366623 [INFO] mod_dialplan_xml.c:315 Processing Paul->5555555555 in context default 2009-11-10 00:35:44.368645 [NOTICE] switch_channel.c:602 New Channel sofia/external/5555555555 [e43092f4-cdba-11de-a864-8927fe94a9f0] 2009-11-10 00:35:47.59221 [NOTICE] sofia_glue.c:2698 Pre-Answer sofia/external/5555555555! 2009-11-10 00:35:47.59221 [INFO] switch_ivr_originate.c:2017 Sending early media 2009-11-10 00:35:47.60524 [INFO] mod_sofia.c:1506 Ring SDP: v=0 o=FreeSWITCH 1257800805 1257800806 IN IP4 74.207.249.79 s=FreeSWITCH c=IN IP4 74.207.249.79 t=0 0 m=audio 30542 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-11-10 00:35:47.60524 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1000 at 74.207.249.79! 2009-11-10 00:35:51.449542 [NOTICE] sofia.c:3849 Hangup sofia/external/5555555555 [CS_EXCHANGE_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2009-11-10 00:35:51.452539 [NOTICE] switch_ivr_bridge.c:419 Hangup sofia/internal/1000 at 74.207.249.79 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/internal/1000 at 74.207.249.79) Ended 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1000 at 74.207.249.79 [CS_DESTROY] 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/external/5555555555) Ended 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/5555555555 [CS_DESTROY] I am really new to VOIP and having a hard time with this. I am really not sure how to proceed. Any help would be really appreciated. Thanks Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091109/b37a55a1/attachment-0002.html From jason at jasonjgw.net Mon Nov 9 22:06:46 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 10 Nov 2009 17:06:46 +1100 Subject: [Freeswitch-users] Configuring freeswitch with voicepulse In-Reply-To: <900c9adf0911092137vf45ec94ie7473d2c08e5ae12@mail.gmail.com> References: <900c9adf0911092137vf45ec94ie7473d2c08e5ae12@mail.gmail.com> Message-ID: <20091110060646.GA24954@jdc.jasonjgw.net> Paul Thirumalai wrote: > I am really new to VOIP and having a hard time with this. I am really not > sure how to proceed. Any help would be really appreciated. First, turn on debug logging (in fs_cli, it's /log debug) to obtain more information. The proxy variables in your configuration could be complicating the situation unnecessarily - try removing them and specifying only the realm. I don't think you want two proxy variables here. If you're just new to FreeSWITCH, leave the debug logging level on and read the logs in /opt/freeswitch/log/freeswitch.log to track down problems. From codecomplete at free.fr Tue Nov 10 01:45:01 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 10 Nov 2009 01:45:01 -0800 (PST) Subject: [Freeswitch-users] Displaying caller ID on LED? Message-ID: <26280730.post@talk.nabble.com> Hello I was wondering if someone had succesfully configured FS to display caller ID on a LED like this? http://usb.brando.com/prod_detail.php?prod_id=00575 That would be a nice alternative to displaying CID information on the user's PC screen when users need to see who's calling where they're not in front of their computer (doctors, auto mechanics, etc.) Thank you. -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280730.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Nov 10 02:06:58 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 10 Nov 2009 02:06:58 -0800 (PST) Subject: [Freeswitch-users] Displaying caller ID on LED? In-Reply-To: <26280730.post@talk.nabble.com> References: <26280730.post@talk.nabble.com> Message-ID: <26280912.post@talk.nabble.com> ... or alternatively, on one of those USB digital picture frames? www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B000087BHC -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From lakindia89 at gmail.com Tue Nov 10 02:51:14 2009 From: lakindia89 at gmail.com (lakshmanan) Date: Tue, 10 Nov 2009 02:51:14 -0800 (PST) Subject: [Freeswitch-users] Flushing the Event buffer in Perl Event Socket In-Reply-To: <191c3a030910300650w7b80568eu4c41c805b9372acc@mail.gmail.com> References: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com> <191c3a030910300650w7b80568eu4c41c805b9372acc@mail.gmail.com> Message-ID: <26281493.post@talk.nabble.com> Hi anthony, I was in a need of flushing the events buffer without reading it.I've done the following ESL(Async) program to flush the events. First I register for events. I answered the call and playback some message. Now the events would have been queued. I, then send "noevents". After sending that, I again register for events, and when I receive the events, I've not got the old events. I got only new events. But I don't know whether it is exactly a way to flush the events or not. I just need your suggestions or your thoughts on this. Here is the script: use lib "/usr/local/freeswitch/scripts/esl"; require ESL; use IO::Socket::INET; use Data::Dumper; my $ip = "192.168..0.0"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; my $con; for(;;) { my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { close($new_sock); next; } my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); print "Host name is $host\n"; $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s to %s\n", $uuid, $info->getHeader("caller-caller-id-number"), $info->getHeader("caller-destination-number"); $con->filter("Unique-Id", $uuid); $con->events("plain", "all"); $con->execute("answer"); $con->setEventLock("true"); $con->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); $con->send("noevents"); sleep(5); $con->events("plain", "all"); while(my $e = $con->recvEvent()) { print $e->serialize(); } } Anthony Minessale-2 wrote: > > read them in a timed loop of some small number of MS until you get a > timeout > meaning you have flushed them all. > > > On Fri, Oct 30, 2009 at 1:57 AM, velusamy velu > wrote: > >> Dear All, >> I receiving the events in while loop by using recvEventTimed method >> in ESL.pm. I have to flush that Event buffer after some particular time. >> How >> can I do it? >> >> Thanks, >> Velusamy >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Flushing-the-Event-buffer-in-Perl-Event-Socket-tp26125824p26281493.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From info at daccii.it Tue Nov 10 05:00:23 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Tue, 10 Nov 2009 14:00:23 +0100 Subject: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it Message-ID: <4AF963E7.2080104@daccii.it> Hi, yesterday i started to fix pronuce in mod_say_it for numbers, dates and times. I needed to add some sound files because these was necessary for a correct italian pronunce. I've patched these three functions: - play_group - it_say_time - it_say_general_count I've diff it against revision 15396 (i've updated freeswitch tree yesterday morning) Can you take a look to the patch? # Modification to play_group function In italian we pronunce 123 as "cento venti tre" and not "uno cento venti tre" so, if a is 1 just doesn't play the digit # Modification to it_say_time Our long date format is something like WDAY_NAME, WDAY_NUMBER MONTH_NAME YEAR so i converted the date pronunce to this. I've dropped am/pm logic, because we have 24h standard, and minutes related logic because, we don't have it. # Modification to it_say_general_count I rewrote number to string conversion to make it more readable (using just two math operations, a module and a division) and to drop the 999 milions limit (1*)(however more code should be changed to fully drop this limit). Changes are mainly related to millions and thousands pronunce: in italian, if you need to say 1 milion you doesn't say "uno milione" but "un milione" but to say 3 millions you say "tre milioni", while for thousand you doesn't pronunce "un" at all. 1*: i've noticed a little bug in xx_say_money in mod_say_xx ... it get up to 12 digits but in xx_say_general_count manage up to 9 digits, so the first three digits wouldn't get never pronunced Thank you -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: mod_say_it.c.fix-pronunce.patch Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/e68393dc/attachment-0002.pl -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/e68393dc/attachment-0002.vcf From brian at freeswitch.org Tue Nov 10 06:10:16 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 08:10:16 -0600 Subject: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it In-Reply-To: <4AF963E7.2080104@daccii.it> References: <4AF963E7.2080104@daccii.it> Message-ID: <5BB9DB8A-8309-463C-B06F-BDA2A2EDF97E@freeswitch.org> Can you post your patch to jira.freeswitch.org please. /b On Nov 10, 2009, at 7:00 AM, Albano Daniele Salvatore - Lavoro wrote: > Hi, > > yesterday i started to fix pronuce in mod_say_it for numbers, dates > and times. I needed to add some sound files because these was > necessary for a correct italian pronunce. > > I've patched these three functions: > - play_group > - it_say_time > - it_say_general_count > > I've diff it against revision 15396 (i've updated freeswitch tree > yesterday morning) > > Can you take a look to the patch? > > > > # Modification to play_group function > > In italian we pronunce 123 as "cento venti tre" and not "uno cento > venti tre" so, if a is 1 just doesn't play the digit > > > > # Modification to it_say_time > > Our long date format is something like > > WDAY_NAME, WDAY_NUMBER MONTH_NAME YEAR > > so i converted the date pronunce to this. > > I've dropped am/pm logic, because we have 24h standard, and minutes > related logic because, we don't have it. > > > > # Modification to it_say_general_count > > I rewrote number to string conversion to make it more readable > (using just two math operations, a module and a division) and to > drop the 999 milions limit (1*)(however more code should be changed > to fully drop this limit). > > Changes are mainly related to millions and thousands pronunce: in > italian, if you need to say 1 milion you doesn't say "uno milione" > but "un milione" but to say 3 millions you say "tre milioni", while > for thousand you doesn't pronunce "un" at all. > > > 1*: i've noticed a little bug in xx_say_money in mod_say_xx ... it > get up to 12 digits but in xx_say_general_count manage up to 9 > digits, so the first three digits wouldn't get never pronunced > > Thank you > > Index: src/mod/say/mod_say_it/mod_say_it.c > =================================================================== > --- src/mod/say/mod_say_it/mod_say_it.c (revisione 15396) > +++ src/mod/say/mod_say_it/mod_say_it.c (copia locale) > @@ -95,7 +95,9 @@ > { > > if (a) { > - say_file("digits/%d.wav", a); > + if (a != 1) { > + say_file("digits/%d.wav", a); > + } > say_file("digits/hundred.wav"); > } > > @@ -170,7 +172,7 @@ > char *tosay, switch_say_type_t type, switch_say_method_t > method, switch_input_args_t *args) > { > int in; > - int x = 0; > + int places_count = 0; > int places[9] = { 0 }; > char sbuf[13] = ""; > switch_status_t status; > @@ -179,26 +181,64 @@ > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Parse > Error!\n"); > return SWITCH_STATUS_GENERR; > } > - > + > + // Get in > in = atoi(tosay); > + > + // Check if number too big > + if (in > 999999999) { > + // Fail > + return SWITCH_STATUS_FALSE; > + } > > + // Check if number isin't zero > if (in != 0) { > - for (x = 8; x >= 0; x--) { > - int num = (int) pow(10, x); > - if ((places[(uint32_t) x] = in / num)) { > - in -= places[(uint32_t) x] * num; > - } > - } > - > + > + // Init x to 0 > + places_count = 0; > + > + // Loop until in is greater than zero > + do { > + // Get last digit > + places[places_count] = in % 10; > + > + // Drop last digit > + in = in / 10; > + } > + while(in > 0 && ++places_count > 0 /** fake check to put in > while */); > + > switch (method) { > case SSM_COUNTED: > case SSM_PRONOUNCED: > - if ((status = play_group(SSM_PRONOUNCED, places[8], places[7], > places[6], "digits/million.wav", session, args)) != > SWITCH_STATUS_SUCCESS) { > - return status; > - } > - if ((status = play_group(SSM_PRONOUNCED, places[5], places[4], > places[3], "digits/thousand.wav", session, args)) != > SWITCH_STATUS_SUCCESS) { > - return status; > - } > + > + // Check for milions > + if (places_count > 5) { > + // Check if the millions digit is one (digit 6 = 1, > digit 7 and 8 = 0) > + if (places[6] == 1 && places[7] == 0 && places[8] > == 0) { > + say_file("digits/un.wav"); > + say_file("digits/million.wav"); > + } else { > + // Play millions group (digits/million.wav > should be digits/millions.wav) > + if ((status = play_group(SSM_PRONOUNCED, places > [8], places[7], places[6], "digits/million.wav", session, args)) != > SWITCH_STATUS_SUCCESS) { > + return status; > + } > + } > + > + } > + > + // Check for thousands > + if (places_count > 2) { > + if (places[3] == 1 && places[4] == 0 && places[5] > == 0) { > + say_file("digits/thousand.wav"); > + } else { > + // Play thousand group > + if ((status = play_group(SSM_PRONOUNCED, places > [5], places[4], places[3], "digits/thousands.wav", session, args)) ! > = SWITCH_STATUS_SUCCESS) { > + return status; > + } > + } > + } > + > + // Play last group > if ((status = play_group(method, places[2], places[1], places[0], > NULL, session, args)) != SWITCH_STATUS_SUCCESS) { > return status; > } > @@ -370,36 +410,19 @@ > > if (say_date) { > say_file("time/day-%d.wav", tm.tm_wday); > + say_num(tm.tm_mday, SSM_PRONOUNCED); > say_file("time/mon-%d.wav", tm.tm_mon); > - say_num(tm.tm_mday, SSM_COUNTED); > say_num(tm.tm_year + 1900, SSM_PRONOUNCED); > } > > if (say_time) { > - int32_t hour = tm.tm_hour, pm = 0; > + say_file("time/hours.wav"); > + say_num(tm.tm_hour, SSM_PRONOUNCED); > > - if (hour > 12) { > - hour -= 12; > - pm = 1; > - } else if (hour == 12) { > - pm = 1; > - } else if (hour == 0) { > - hour = 12; > - pm = 0; > - } > - > - say_num(hour, SSM_PRONOUNCED); > - > - if (tm.tm_min > 9) { > + if (tm.tm_min) { > + say_file("time/and.wav"); > say_num(tm.tm_min, SSM_PRONOUNCED); > - } else if (tm.tm_min) { > - say_file("time/oh.wav"); > - say_num(tm.tm_min, SSM_PRONOUNCED); > - } else { > - say_file("time/oclock.wav"); > } > - > - say_file("time/%s.wav", pm ? "p-m" : "a-m"); > } > > return SWITCH_STATUS_SUCCESS; > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Tue Nov 10 06:11:11 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 08:11:11 -0600 Subject: [Freeswitch-users] Flushing the Event buffer in Perl Event Socket In-Reply-To: <26281493.post@talk.nabble.com> References: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com> <191c3a030910300650w7b80568eu4c41c805b9372acc@mail.gmail.com> <26281493.post@talk.nabble.com> Message-ID: <2CD889D1-934A-47CE-A938-1EB9D4325DD2@freeswitch.org> $| = 1; I think that is what you're lookin for. /b On Nov 10, 2009, at 4:51 AM, lakshmanan wrote: > I was in a need of flushing the events buffer without reading > it.I've done > the following ESL(Async) program to flush the events. From codecomplete at free.fr Tue Nov 10 06:13:17 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 10 Nov 2009 06:13:17 -0800 (PST) Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> Message-ID: <26284109.post@talk.nabble.com> Are non-English sound files available in the SVN version of the code? I just tried installing the French sound files, but got an error: Unknown target cd-sounds-fr-install Unknown target cd-moh-fr-install make[1]: *** [cd-sounds-fr-install] Error 1 make: *** [cd-sounds-fr-install] Error 2 make[1]: *** [cd-moh-fr-install] Error 1 make: *** [cd-moh-fr-install] Error 2 [1]+ Exit 2 make cd-sounds-fr-install -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26284109.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From info at daccii.it Tue Nov 10 06:38:33 2009 From: info at daccii.it (Daniele Salvatore Albano) Date: Tue, 10 Nov 2009 15:38:33 +0100 Subject: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it In-Reply-To: <5BB9DB8A-8309-463C-B06F-BDA2A2EDF97E@freeswitch.org> References: <4AF963E7.2080104@daccii.it> <5BB9DB8A-8309-463C-B06F-BDA2A2EDF97E@freeswitch.org> Message-ID: <4AF97AE9.8090103@daccii.it> Hi, patch posted to http://jira.freeswitch.org/browse/MODAPP-362 Best Regards, Daniele Brian West ha scritto: > Can you post your patch to jira.freeswitch.org please. > > /b -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 307 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/938fd236/attachment-0002.vcf From rupa at rupa.com Tue Nov 10 06:56:23 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 10 Nov 2009 06:56:23 -0800 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> Message-ID: 2009/11/9 Jo?o Mesquita : > Beat me with a dead cat all you want but I rather the snom m3 than the > Siemens A580IP.... Siemens has very low volume which makes its call quality > suck despite of being ergonomic and all... Did you flip hte option in the base station that tells it to make the audio louder? > That gigaset application sucks and the base station is slow as hell... Maybe > I have a bad unit? I didn't play with any of the gigaset specific stuff, I've disabled any screen savers. Maybe if I had the more "fancy" handsets the apps would be more useful, but when using the base handset they are not. Oh, and the weather is in C rather than F -- good for the rest of the world but not for us in the US. The base station is very slow with firefox but when I use chrome isn't so bad. Dunno if it was a combination of extensions or what. Oh, and it looks like a new firmware came out for the Siemens today. Wonder what it fixes (and breaks). Hmm.. wonder where I can find a list of whats new. > The snom m3 has its downsides, but all and all, I am happy with the phone if > you consider its price tag here in South America where a Polycom can easily > cost over 200USD the cheapest unit. > > Regards, > > JM > > On Mon, Nov 9, 2009 at 9:01 PM, Anthony Minessale > wrote: >> >> asstra has one issue where if you look at them wrong they start telling >> the server that the media ip is 0.0.0.0 which we have never identified but >> they indeed seem to work better than snom m3 >> >> >> On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale >> wrote: >>> >>> >The Snom M3 is one of the ones that I was looking at - I would be >>> > interested in the "Pro's & Cons" ? >>> >>> Worst POS I have ever used, from a sound quality to ergonomics pov, tech >>> support was as bad... >>> >>> I have Aastra 480i CT's which work well. >>> >>> jlc >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From brian at freeswitch.org Tue Nov 10 07:03:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 09:03:58 -0600 Subject: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it In-Reply-To: <4AF97AE9.8090103@daccii.it> References: <4AF963E7.2080104@daccii.it> <5BB9DB8A-8309-463C-B06F-BDA2A2EDF97E@freeswitch.org> <4AF97AE9.8090103@daccii.it> Message-ID: Patch applied. // comments aren't allowed in .c files in our tree we try hard to weed them out... anyway its committed now. And thanks for your contribution. /b On Nov 10, 2009, at 8:38 AM, Daniele Salvatore Albano wrote: > Hi, > > patch posted to http://jira.freeswitch.org/browse/MODAPP-362 > > > Best Regards, > Daniele > > Brian West ha scritto: >> Can you post your patch to jira.freeswitch.org please. >> >> /b > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From piotr_zurek at biprotech.com Tue Nov 10 07:01:17 2009 From: piotr_zurek at biprotech.com (=?UTF-8?B?UGlvdHIgxbt1cmVr?=) Date: Tue, 10 Nov 2009 16:01:17 +0100 Subject: [Freeswitch-users] How to pick up someone's phone remotely. Message-ID: <4AF9803D.9050806@biprotech.com> Hello. Thank You developers for Freeswitch. I have installed it lately and it's working quite nicely, but I have one problem: I need to mimic behavior of my current analogue PBX installation using Freeswitch. This is the scenario: In the office with a few desks (extensions 1000-1010) and only one person behind one of desks (whatever extension - in example 1000). 1. There's incoming call on _one_ of extensions 1001-1010 2. The person on extension 1000 wants to answer this call on his phone so dials #37 and this call is redirected to his phone. That's how it works on my office on analogue PBX system. Anyone can answer a call from any other phone as long as it hasn't been answered already. I tried to use the intercept action (with global example in default config) but it's not what I need because it intercepts the call even if it's already answered. I need to intercept all but only unanswered calls. I tried to use Redirect but it does not work on other's extensions call's (or does it?). Please help. Peter ?urek -------------- next part -------------- A non-text attachment was scrubbed... Name: piotr_zurek.vcf Type: text/x-vcard Size: 414 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/76bc8e42/attachment-0002.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3678 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/76bc8e42/attachment-0002.bin From rupa at rupa.com Tue Nov 10 07:09:18 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 10 Nov 2009 07:09:18 -0800 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> Message-ID: On Tue, Nov 10, 2009 at 6:56 AM, Rupa Schomaker wrote: > Oh, and it looks like a new firmware came out for the Siemens today. > Wonder what it fixes (and breaks). ?Hmm.. wonder where I can find a > list of whats new. Well, it seems to totally break g722. I haven't had a chance to narrow it down further, but beware if g722 is important don't update... -- -Rupa From brian at freeswitch.org Tue Nov 10 07:16:08 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 09:16:08 -0600 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <4AF9803D.9050806@biprotech.com> References: <4AF9803D.9050806@biprotech.com> Message-ID: <7B15EB87-5EDD-4234-8512-B1536B25DBEA@freeswitch.org> Please see the global-intercept example in the default config. /b On Nov 10, 2009, at 9:01 AM, Piotr ?urek wrote: > Hello. > > Thank You developers for Freeswitch. > I have installed it lately and it's working quite nicely, but I have > one problem: > > I need to mimic behavior of my current analogue PBX installation > using Freeswitch. > > This is the scenario: > In the office with a few desks (extensions 1000-1010) and only one > person behind one of desks (whatever extension - in example 1000). > 1. There's incoming call on _one_ of extensions 1001-1010 > 2. The person on extension 1000 wants to answer this call on his > phone so dials #37 and this call is redirected to his phone. > > That's how it works on my office on analogue PBX system. Anyone can > answer a call from any other phone as long as it hasn't been > answered already. > > I tried to use the intercept action (with global example in default > config) but it's not what I need because it intercepts the call even > if it's already answered. I need to intercept all but only > unanswered calls. I tried to use Redirect but it does not work on > other's extensions call's (or does it?). > > Please help. > Peter ?urek > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From markmorreny at gmail.com Tue Nov 10 07:29:37 2009 From: markmorreny at gmail.com (mark morreny) Date: Tue, 10 Nov 2009 23:29:37 +0800 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20091109192904.GI9418@hijacked.us> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> <20091109192904.GI9418@hijacked.us> Message-ID: <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> Hi Thanks for the tips. May I ask how to split the file from hadoop to the shell? Is it like copying the file to certain dir? I can't find any mod_shell_stream related info from the wiki. Does anyone know how to use it? thx, mark On Tue, Nov 10, 2009 at 3:29 AM, Andrew Thompson wrote: > On Mon, Nov 09, 2009 at 08:59:54PM +0800, mark morreny wrote: > > Hi, > > > > Does anyone know how to playback based on files from hadoop storage. > > > > There is a libhdcp, and java api. Is there anyway to put together a > sample > > middle piece to move files from hadoop to freeswitch using memory space, > so > > there is no disk I/O? > > > > Any feedback or suggestion will be greatly appreciated. > > > > mod_shell_stream might work, if you can just spit out the raw audio to > the shell. Or write another stream module that works with libhdcp. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/3290a455/attachment-0002.html From andrew at hijacked.us Tue Nov 10 07:56:33 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 10 Nov 2009 10:56:33 -0500 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> <20091109192904.GI9418@hijacked.us> <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> Message-ID: <20091110155633.GA194@hijacked.us> On Tue, Nov 10, 2009 at 11:29:37PM +0800, mark morreny wrote: > Hi > > Thanks for the tips. May I ask how to split the file from hadoop to the > shell? Is it like copying the file to certain dir? > > I can't find any mod_shell_stream related info from the wiki. Does anyone > know how to use it? > mod_shell_stream is undocumented, but from reading the code I gather it works like this: Module calls fork() and in the child process it runs an arbitrary shell command (specified in its config file?). The parent process then reads raw audio data from the child process and uses it as an audio source. So basicially you could write the shell command in anything, so long as it outputs raw audio to FS. Or maybe I read the code wrong when I skimmed over it. If you do get it working, please contribute some documentation to the wiki. Andrew From oseslija at gmail.com Tue Nov 10 08:06:22 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Tue, 10 Nov 2009 17:06:22 +0100 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <4AF9803D.9050806@biprotech.com> References: <4AF9803D.9050806@biprotech.com> Message-ID: <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> Add the following: . after in local extensions default example, or change it globally previously than this extension. You can join us on IRC if you can any more questions (sekil). Regards, Ognjen On Tue, Nov 10, 2009 at 4:01 PM, Piotr ?urek wrote: > Hello. > > Thank You developers for Freeswitch. > I have installed it lately and it's working quite nicely, but I have one > problem: > > I need to mimic behavior of my current analogue PBX installation using > Freeswitch. > > This is the scenario: > In the office with a few desks (extensions 1000-1010) and only one person > behind one of desks (whatever extension - in example 1000). > 1. There's incoming call on _one_ of extensions 1001-1010 > 2. The person on extension 1000 wants to answer this call on his phone so > dials #37 and this call is redirected to his phone. > > That's how it works on my office on analogue PBX system. Anyone can answer > a call from any other phone as long as it hasn't been answered already. > > I tried to use the intercept action (with global example in default config) > but it's not what I need because it intercepts the call even if it's already > answered. I need to intercept all but only unanswered calls. I tried to use > Redirect but it does not work on other's extensions call's (or does it?). > > Please help. > Peter ?urek > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/28693819/attachment-0002.html From stevendt at primrosebank.net Tue Nov 10 08:24:02 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 10 Nov 2009 16:24:02 -0000 Subject: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) Message-ID: <9E5323D6B69B489384D2E89358CC5EC5@bp1.ad.bp.com> I'm close to getting my SPA3102 working - he says hopefully .. . . . Making and receiving calls seems to be OK, but the SAP3102 doesn't seem to want to let go of the phone line once it's got it. Example I can receive a call, nobody answers and it goes to voicemail - working so far. FreeSwitch processes the normal VoiceMail system playing the prompts and recording the call. At the end, it says "Goodbye" and hear a "click" (from the remote end) and I see the console message that 2009-11-10 15:52:52.625000 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/1000 at 192.168.1.181 [CS_EXECUTE] [NORMAL_CLEARING] 2009-11-10 15:52:52.625000 [NOTICE] switch_core_session.c:1086 Session 362 (sofia/internal/1000 at 192.168.1.181) Ended 2009-11-10 15:52:52.625000 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1000 at 192.168.1.181 [CS_DESTROY] The above messages would suggest to me that FreeSwitch is doing its stuff right, but I have posted a dump in the pastebin just in case. The SPA3102 does not want to relinquish the line until the remote caller hangs up. Has anyone had similar problems with the SPA3102 or has any ideas where I can look to get to the bottom of the problem. (I have just upgraded the SPA3102 to the latest 5.1.0 firmware) regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/bd825583/attachment-0002.html From jpitcher at nuvio.com Tue Nov 10 08:03:00 2009 From: jpitcher at nuvio.com (Jonathan Pitcher) Date: Tue, 10 Nov 2009 08:03:00 -0800 Subject: [Freeswitch-users] Dialplans and XML_CURL Message-ID: Good morning everyone. I have a question regarding using MOD XML_CURL and returning a dial plan. I have my system setup to respond with the following dialplan.
My question is this. Can extension one, use extension two and three without XML_CURL making another dialplan request? Thanks, Jonathan Pitcher -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/08c48564/attachment-0002.html From mrene_lists at avgs.ca Tue Nov 10 08:55:58 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 10 Nov 2009 08:55:58 -0800 Subject: [Freeswitch-users] Dialplans and XML_CURL In-Reply-To: References: Message-ID: <7319C8C2-08D6-4E8F-AFAE-F9318700F6BD@avgs.ca> You'll get a single xml curl request, unless you use the transfer application, which will trigger another one. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 10-Nov-09, at 8:03 AM, Jonathan Pitcher wrote: > Good morning everyone. I have a question regarding using MOD > XML_CURL and returning a dial plan. > > I have my system setup to respond with the following dialplan. > > > > >
> > > > > > > > > > > > > > > > > > > > > >
> > > My question is this. Can extension one, use extension two and three > without XML_CURL making another dialplan request? > > Thanks, > > Jonathan Pitcher > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/f9f2e4c0/attachment-0002.html From msc at freeswitch.org Tue Nov 10 09:32:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Nov 2009 09:32:38 -0800 Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <26284109.post@talk.nabble.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> <26284109.post@talk.nabble.com> Message-ID: <87f2f3b90911100932i19c7c971y5fae90f6bb9f4dc0@mail.gmail.com> I believe that French and Spanish sounds are in the works by the community. The only other sounds I'm aware of are the Russian ones. -MC On Tue, Nov 10, 2009 at 6:13 AM, Fred-145 wrote: > > Are non-English sound files available in the SVN version of the code? > > I just tried installing the French sound files, but got an error: > > Unknown target cd-sounds-fr-install > Unknown target cd-moh-fr-install > make[1]: *** [cd-sounds-fr-install] Error 1 > make: *** [cd-sounds-fr-install] Error 2 > make[1]: *** [cd-moh-fr-install] Error 1 > make: *** [cd-moh-fr-install] Error 2 > [1]+ Exit 2 make cd-sounds-fr-install > -- > View this message in context: > http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26284109.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/370738d2/attachment-0002.html From msc at freeswitch.org Tue Nov 10 09:44:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Nov 2009 09:44:17 -0800 Subject: [Freeswitch-users] Dialplans and XML_CURL In-Reply-To: <7319C8C2-08D6-4E8F-AFAE-F9318700F6BD@avgs.ca> References: <7319C8C2-08D6-4E8F-AFAE-F9318700F6BD@avgs.ca> Message-ID: <87f2f3b90911100944i55a75b5u44f46bd3f906a4b6@mail.gmail.com> On Tue, Nov 10, 2009 at 8:55 AM, Mathieu Rene wrote: > You'll get a single xml curl request, unless you use the transfer > application, which will trigger another one. > Just curious: what about execute_extension? Does that cause a new XML CURL request also? I didn't see anything on the wiki about that. I'll update the wiki mod_xml_curl page accordingly. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/5e8f7438/attachment-0002.html From msc at freeswitch.org Tue Nov 10 09:53:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Nov 2009 09:53:11 -0800 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20091110155633.GA194@hijacked.us> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> <20091109192904.GI9418@hijacked.us> <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> <20091110155633.GA194@hijacked.us> Message-ID: <87f2f3b90911100953n78ae7a54kd253840217188827@mail.gmail.com> On Tue, Nov 10, 2009 at 7:56 AM, Andrew Thompson wrote: > On Tue, Nov 10, 2009 at 11:29:37PM +0800, mark morreny wrote: > > Hi > > > > Thanks for the tips. May I ask how to split the file from hadoop to the > > shell? Is it like copying the file to certain dir? > > > > I can't find any mod_shell_stream related info from the wiki. Does > anyone > > know how to use it? > > > > mod_shell_stream is undocumented, but from reading the code I gather it > works like this: > > Module calls fork() and in the child process it runs an arbitrary shell > command (specified in its config file?). The parent process then reads > raw audio data from the child process and uses it as an audio source. > > So basicially you could write the shell command in anything, so long as > it outputs raw audio to FS. > > Or maybe I read the code wrong when I skimmed over it. If you do get it > working, please contribute some documentation to the wiki. > > Andrew > > Andrew, Thanks for poking around in there. I made a stub for this mod on the wiki: http://wiki.freeswitch.org/wiki/Mod_shell_stream If anyone is familiar with it and could throw an example up there that would be much appreciated. Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/e29fbe92/attachment-0002.html From info at daccii.it Tue Nov 10 10:05:00 2009 From: info at daccii.it (Daniele Salvatore Albano) Date: Tue, 10 Nov 2009 19:05:00 +0100 Subject: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it In-Reply-To: References: <4AF963E7.2080104@daccii.it> <5BB9DB8A-8309-463C-B06F-BDA2A2EDF97E@freeswitch.org> <4AF97AE9.8090103@daccii.it> Message-ID: <4AF9AB4C.6030507@daccii.it> Hi, thank you and for your work! Where i can find coding style rules? Best Regards, Daniele Brian West ha scritto: > Patch applied. // comments aren't allowed in .c files in our tree we > try hard to weed them out... anyway its committed now. And thanks for > your contribution. > > /b -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 307 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/5cdceb49/attachment-0002.vcf From jmesquita at freeswitch.org Tue Nov 10 11:04:16 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 10 Nov 2009 17:04:16 -0200 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: References: <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <191c3a030911091501j14512c97l5dc3078a9970115e@mail.gmail.com> Message-ID: Rupa, I tried flipping, yes, but no, it works bad... Thanks for the heads up on G722! Regards, JM On Tue, Nov 10, 2009 at 1:09 PM, Rupa Schomaker wrote: > On Tue, Nov 10, 2009 at 6:56 AM, Rupa Schomaker wrote: > > Oh, and it looks like a new firmware came out for the Siemens today. > > Wonder what it fixes (and breaks). Hmm.. wonder where I can find a > > list of whats new. > > Well, it seems to totally break g722. I haven't had a chance to > narrow it down further, but beware if g722 is important don't > update... > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/f41665d9/attachment-0002.html From oseslija at gmail.com Tue Nov 10 11:56:38 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Tue, 10 Nov 2009 20:56:38 +0100 Subject: [Freeswitch-users] Cordless VOIP Phones In-Reply-To: <1257798401.10738.18.camel@sodium> References: <6B46BB75-C396-4426-86EF-DC7CE28BA8AE@freeswitch.org> <2498C810567A4F01B22119318B6803F2@bp1.ad.bp.com> <0C3195A85F8543D09019FDB14E88280A@bp1.ad.bp.com> <2815B65B0C704F638BEA0122AFF6EEE2@bp1.ad.bp.com> <659847D6-10B0-4E2B-A4B4-352D9401077A@freeswitch.org> <1257798401.10738.18.camel@sodium> Message-ID: <4468a6770911101156q3083747bl77976f98e930c045@mail.gmail.com> Hey Hadley, jump up on irc sometimes. Regards, Ognjen On Mon, Nov 9, 2009 at 9:26 PM, Hadley Rich wrote: > On Mon, 2009-11-09 at 14:05 -0600, Brian West wrote: > > Get an ATA with a Dect handset it works much better... the Snom M3 and > > the Aastra are one in the same and they both do not live up to the > > quality or usability requirements. > > That said, they are better than what else is around. > > I'd call them average. Nothing to write home about but you don't need to > run away from them. > > hads > > -- > http://nicegear.co.nz > New Zealand's Open Source Hardware Supplier > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/6d1dfbea/attachment-0002.html From sergey.kobzar at mail.ru Tue Nov 10 13:27:20 2009 From: sergey.kobzar at mail.ru (Sergey Kobzar) Date: Tue, 10 Nov 2009 23:27:20 +0200 Subject: [Freeswitch-users] SIP trunk without authentication Message-ID: <1352396721.20091110232720@mail.ru> Hello. I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP. I've searched through FS documentation and didn't find clear answer. Could you help me or maybe give a link to a doc which can help? Thanks. -- Sergey From mrene_lists at avgs.ca Tue Nov 10 13:43:04 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 10 Nov 2009 13:43:04 -0800 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: <1352396721.20091110232720@mail.ru> References: <1352396721.20091110232720@mail.ru> Message-ID: As easy as: in your dialplan. If you want to make a gateway out of it, you can enter whatever you want in username and password since they won't be used. (SIP works using challenge authentication which means the remote UA has to send you a packet requesting the credentials). Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote: > Hello. > > I'm FS newbie and want connect it to SIP provider which does not > require authentication - it make authentication using my IP. > > I've searched through FS documentation and didn't find clear answer. > > Could you help me or maybe give a link to a doc which can help? > > Thanks. > > > -- > Sergey > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nandy1925 at gmail.com Tue Nov 10 13:48:15 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 11 Nov 2009 05:48:15 +0800 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> Message-ID: <7d0bfd8c0911101348n5d7dfd20p224d972d68a1299d@mail.gmail.com> just change the dialplan/default.xml as mentioned by brian but i think you can't use # as the first key 'cuz it normally used as a Send key. you may change # to * (star key). On Wed, Nov 11, 2009 at 12:06 AM, Ognjen Seslija wrote: > Add the following: > > . > > after > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > > in local extensions default example, or change it globally previously than > this extension. You can join us on IRC if you can any more questions > (sekil). > > Regards, > Ognjen > > > > On Tue, Nov 10, 2009 at 4:01 PM, Piotr ?urek wrote: > >> Hello. >> >> Thank You developers for Freeswitch. >> I have installed it lately and it's working quite nicely, but I have one >> problem: >> >> I need to mimic behavior of my current analogue PBX installation using >> Freeswitch. >> >> This is the scenario: >> In the office with a few desks (extensions 1000-1010) and only one person >> behind one of desks (whatever extension - in example 1000). >> 1. There's incoming call on _one_ of extensions 1001-1010 >> 2. The person on extension 1000 wants to answer this call on his phone so >> dials #37 and this call is redirected to his phone. >> >> That's how it works on my office on analogue PBX system. Anyone can answer >> a call from any other phone as long as it hasn't been answered already. >> >> I tried to use the intercept action (with global example in default >> config) but it's not what I need because it intercepts the call even if it's >> already answered. I need to intercept all but only unanswered calls. I tried >> to use Redirect but it does not work on other's extensions call's (or does >> it?). >> >> Please help. >> Peter ?urek >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/45b24bf1/attachment-0002.html From brian at freeswitch.org Tue Nov 10 14:23:43 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 16:23:43 -0600 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <7d0bfd8c0911101348n5d7dfd20p224d972d68a1299d@mail.gmail.com> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> <7d0bfd8c0911101348n5d7dfd20p224d972d68a1299d@mail.gmail.com> Message-ID: <030D9DFF-7AFE-4942-8BEF-B374F8600396@freeswitch.org> That depends on the phone... some let you do it.. some don't... WELCOME TO VOIP!!! /b On Nov 10, 2009, at 3:48 PM, Nandy Dagondon wrote: > just change the dialplan/default.xml as mentioned by brian but i > think you can't use # as the first key 'cuz it normally used as a > Send key. you may change # to * (star key). From ujjval at simplesignal.com Tue Nov 10 14:39:33 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Tue, 10 Nov 2009 14:39:33 -0800 Subject: [Freeswitch-users] Simple Conference Setup issue Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175CF5087E@EXMBXCLUS01.citservers.local> I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch and when I have 2 people dial in, looks like the Music on Hold never stops. Here is what my public.xml looks like: Help appreciated -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/cf1e9e40/attachment-0002.html From brian at freeswitch.org Tue Nov 10 14:49:48 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 16:49:48 -0600 Subject: [Freeswitch-users] Simple Conference Setup issue In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175CF5087E@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175CF5087E@EXMBXCLUS01.citservers.local> Message-ID: <0B5A83F9-9754-44ED-A5C7-447B7F050255@freeswitch.org> What does your config look like? /b On Nov 10, 2009, at 4:39 PM, Ujjval Karihaloo wrote: > I am trying to call into a DID that is pointed to a Conf Bridge on > Freeswitch and when I have 2 people dial in, looks like the Music on > Hold never stops. > > Here is what my public.xml looks like: > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/bc2edeef/attachment-0002.html From ujjval at simplesignal.com Tue Nov 10 15:09:30 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Tue, 10 Nov 2009 15:09:30 -0800 Subject: [Freeswitch-users] Simple Conference Setup issue In-Reply-To: <0B5A83F9-9754-44ED-A5C7-447B7F050255@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175CF5087E@EXMBXCLUS01.citservers.local> <0B5A83F9-9754-44ED-A5C7-447B7F050255@freeswitch.org> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175CF508A9@EXMBXCLUS01.citservers.local> My mistake , it picked the default profile and was waiting for moderator in the conference.cof.xml file that is provided with the install. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, November 10, 2009 3:50 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Simple Conference Setup issue What does your config look like? /b On Nov 10, 2009, at 4:39 PM, Ujjval Karihaloo wrote: I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch and when I have 2 people dial in, looks like the Music on Hold never stops. Here is what my public.xml looks like: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/30b34810/attachment-0002.html From malay.thakershi at continuityhealth.com Tue Nov 10 15:25:24 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Tue, 10 Nov 2009 17:25:24 -0600 Subject: [Freeswitch-users] Help with dynamic IVR Message-ID: <006101ca625d$14257480$3c705d80$@thakershi@continuityhealth.com> Hello. I am very new to FreeSwitch, Telephony and IVR. My goal is to prepare a student assessment IVR system as a college project. But this IVR is going to be dynamic. So for each student assessment may be different (number of questions, possible responses, flow of prompts, etc). Is it possible to achieve something like this with FreeSwitch? Most IVR we see are static (like a bank IVR system that flows always in same way). That is why I am confused. Please share your views. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/56a56a9c/attachment-0002.html From lei.tlfly at gmail.com Tue Nov 10 17:55:15 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 11 Nov 2009 09:55:15 +0800 Subject: [Freeswitch-users] Help with dynamic IVR In-Reply-To: <-5075485054665589342@unknownmsgid> References: <-5075485054665589342@unknownmsgid> Message-ID: <50c41b4e0911101755j6e99a539pc0053befa9086ff9@mail.gmail.com> As I opinion, it's not necessary write ivr script for each student. A "static" ivr script load question and response dynamic is what you need. 2009/11/11 Malay Thakershi > Hello. I am very new to FreeSwitch, Telephony and IVR. > > > > My goal is to prepare a student assessment IVR system as a college project. > But this IVR is going to be dynamic. So for each student assessment may be > different (number of questions, possible responses, flow of prompts, etc). > Is it possible to achieve something like this with FreeSwitch? Most IVR we > see are static (like a bank IVR system that flows always in same way). That > is why I am confused. Please share your views. > > > > Malay Thakershi > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/19742746/attachment-0002.html From markmorreny at gmail.com Tue Nov 10 19:02:10 2009 From: markmorreny at gmail.com (mark morreny) Date: Wed, 11 Nov 2009 11:02:10 +0800 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20091110155633.GA194@hijacked.us> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> <20091109192904.GI9418@hijacked.us> <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> <20091110155633.GA194@hijacked.us> Message-ID: <20ad6b920911101902w52165681tb47fe8f3aa2ae76e@mail.gmail.com> Hi Sorry to ask again. I know the command to copy file from hadoop file system to somewhere else. But how do I make a shell command to output raw audio? What command is it like? Is it like play()? I am confused. Thx, mark On Tue, Nov 10, 2009 at 11:56 PM, Andrew Thompson wrote: > On Tue, Nov 10, 2009 at 11:29:37PM +0800, mark morreny wrote: > > Hi > > > > Thanks for the tips. May I ask how to split the file from hadoop to the > > shell? Is it like copying the file to certain dir? > > > > I can't find any mod_shell_stream related info from the wiki. Does > anyone > > know how to use it? > > > > mod_shell_stream is undocumented, but from reading the code I gather it > works like this: > > Module calls fork() and in the child process it runs an arbitrary shell > command (specified in its config file?). The parent process then reads > raw audio data from the child process and uses it as an audio source. > > So basicially you could write the shell command in anything, so long as > it outputs raw audio to FS. > > Or maybe I read the code wrong when I skimmed over it. If you do get it > working, please contribute some documentation to the wiki. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/2f4d6193/attachment-0002.html From quentusrex at gmail.com Tue Nov 10 19:22:34 2009 From: quentusrex at gmail.com (William King) Date: Tue, 10 Nov 2009 19:22:34 -0800 Subject: [Freeswitch-users] Help with dynamic IVR In-Reply-To: <50c41b4e0911101755j6e99a539pc0053befa9086ff9@mail.gmail.com> References: <-5075485054665589342@unknownmsgid> <50c41b4e0911101755j6e99a539pc0053befa9086ff9@mail.gmail.com> Message-ID: <4AFA2DFA.3070205@gmail.com> What you will probably want, if you are looking to go 'thicker' with this would be one of the IVR scripting languages and a database connection. For instance lua, and the database connection(either mysql or postgresql or sqlite). ' From there you have users, questions, and answers mapped in the database. Feel free to e-mail me about this off list for more assistance. -William King Lei Tang wrote: > As I opinion, it's not necessary write ivr script for each student. A > "static" ivr script load question and response dynamic is what you need. > > 2009/11/11 Malay Thakershi > > > Hello. I am very new to FreeSwitch, Telephony and IVR. > > > > My goal is to prepare a student assessment IVR system as a college > project. But this IVR is going to be dynamic. So for each student > assessment may be different (number of questions, possible > responses, flow of prompts, etc). Is it possible to achieve > something like this with FreeSwitch? Most IVR we see are static > (like a bank IVR system that flows always in same way). That is > why I am confused. Please share your views. > > > > Malay Thakershi > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Lei.Tang > lei.tlfly at gmail.com > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mitch.capper at gmail.com Tue Nov 10 20:01:44 2009 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 10 Nov 2009 23:01:44 -0500 Subject: [Freeswitch-users] Displaying caller ID on LED? In-Reply-To: <26280912.post@talk.nabble.com> References: <26280730.post@talk.nabble.com> <26280912.post@talk.nabble.com> Message-ID: I did something like this recently. From the dial plan it is easy to execute an external application on an incoming call with the caller's info. At that point if you can just push it down to the LCD panel all the better, but if your FS server is remote, and has no direct access to the client to render the caller ID, you will have to setup a fake push to get instant responses. You can do this through apache, or a simple tcp server but the idea being the client connects up to the server, and the server blocks until an incoming call comes in, it then responds to the client, and you have the caller id fairly instantly showing up. You could also use the event socket, heck even maybe use the event socket remotely if you wanted to, and then avoid some of the server side complexity too. ~Mitch On Tue, Nov 10, 2009 at 5:06 AM, Fred-145 wrote: > > ... or alternatively, on one of those USB digital picture frames? > > www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B000087BHC > -- > View this message in context: > http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/91f8f996/attachment-0002.html From lakindia89 at gmail.com Tue Nov 10 20:10:52 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Wed, 11 Nov 2009 09:40:52 +0530 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events In-Reply-To: <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> References: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> Message-ID: <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> Here is the required detail. http://pastebin.freeswitch.org/11049 On Mon, Nov 9, 2009 at 10:04 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > 1) install gdb > 2) run support_d/fscore_db in the tree from the working directory of the > core. > 3) if you are not on svn trunk, "make current" and start over. > > > On Mon, Nov 9, 2009 at 5:53 AM, lakshmanan ganapathy > wrote: > >> Dear all, >> I did the below code, to callback a function when CHANNEL_EXECUTE_COMPLETE >> event comes. >> I executed the script for the 1st time and I got nothing. >> When I executed the script for the 2nd time, it ended with Sedmentation >> fault with core dumped. >> >> I was unable to attach the core dump file with this mail. >> Please specify how to send files to freeswitch user mailing list if need >> be. >> >> The freeswitch log is here: >> http://pastebin.freeswitch.org/11038 >> >> #!/usr/bin/perl >> use strict; >> use Data::Dumper; >> our $session; >> $session->answer(); >> my $events=new freeswitch::EventConsumer("CHANNEL_EXECUTE_COMPLETE"); >> $events->pop(1); >> $events->swig_e_callback_set("playvoice"); >> sub playvoice() >> { >> freeswitch::consoleLog("INFO","Call back function called\n"); >> } >> return 1; >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/8c7938e0/attachment-0002.html From jmesquita at freeswitch.org Tue Nov 10 20:12:03 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 11 Nov 2009 01:12:03 -0300 Subject: [Freeswitch-users] Displaying caller ID on LED? In-Reply-To: References: <26280730.post@talk.nabble.com> <26280912.post@talk.nabble.com> Message-ID: If you donate one to the FsGui project, I can make it happen for you. Contact me off list if you are interested. Regards, JM On Wed, Nov 11, 2009 at 1:01 AM, Mitch Capper wrote: > I did something like this recently. From the dial plan it is easy to > execute an external application on an incoming call with the caller's info. > At that point if you can just push it down to the LCD panel all the better, > but if your FS server is remote, and has no direct access to the client to > render the caller ID, you will have to setup a fake push to get instant > responses. You can do this through apache, or a simple tcp server but the > idea being the client connects up to the server, and the server blocks until > an incoming call comes in, it then responds to the client, and you have the > caller id fairly instantly showing up. You could also use the event > socket, heck even maybe use the event socket remotely if you wanted to, and > then avoid some of the server side complexity too. > > ~Mitch > > > On Tue, Nov 10, 2009 at 5:06 AM, Fred-145 wrote: > >> >> ... or alternatively, on one of those USB digital picture frames? >> >> www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B000087BHC >> -- >> View this message in context: >> http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/6c164975/attachment-0002.html From brian at freeswitch.org Tue Nov 10 20:39:01 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Nov 2009 22:39:01 -0600 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events In-Reply-To: <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> References: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> Message-ID: <2E2AE3DF-1EA7-45AA-9C64-359C5D7585B7@freeswitch.org> You need to install the debug packages so you the symbols because that backtrace is useless. /b On Nov 10, 2009, at 10:10 PM, lakshmanan ganapathy wrote: > Here is the required detail. > > http://pastebin.freeswitch.org/11049 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/b2b63900/attachment-0002.html From mrene_lists at avgs.ca Tue Nov 10 21:18:40 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 10 Nov 2009 21:18:40 -0800 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events In-Reply-To: <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> References: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> Message-ID: It doesn't look like its trying to look for symbols inside freeswitch gdb /path/to/freeswitch/here /path/to/core/here bt thread apply all bt Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 10-Nov-09, at 8:10 PM, lakshmanan ganapathy wrote: > Here is the required detail. > > http://pastebin.freeswitch.org/11049 > > On Mon, Nov 9, 2009 at 10:04 PM, Anthony Minessale > wrote: > 1) install gdb > 2) run support_d/fscore_db in the tree from the working directory of > the core. > 3) if you are not on svn trunk, "make current" and start over. > > > On Mon, Nov 9, 2009 at 5:53 AM, lakshmanan ganapathy > wrote: > Dear all, > I did the below code, to callback a function when > CHANNEL_EXECUTE_COMPLETE event comes. > I executed the script for the 1st time and I got nothing. > When I executed the script for the 2nd time, it ended with > Sedmentation fault with core dumped. > > I was unable to attach the core dump file with this mail. > Please specify how to send files to freeswitch user mailing list if > need be. > > The freeswitch log is here: > http://pastebin.freeswitch.org/11038 > > #!/usr/bin/perl > use strict; > use Data::Dumper; > our $session; > $session->answer(); > my $events=new freeswitch::EventConsumer("CHANNEL_EXECUTE_COMPLETE"); > $events->pop(1); > $events->swig_e_callback_set("playvoice"); > sub playvoice() > { > freeswitch::consoleLog("INFO","Call back function called\n"); > } > return 1; > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091110/2309380e/attachment-0002.html From lakindia89 at gmail.com Tue Nov 10 21:22:49 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Wed, 11 Nov 2009 10:52:49 +0530 Subject: [Freeswitch-users] Freeswitch core dumped, when setting callback to events In-Reply-To: <2E2AE3DF-1EA7-45AA-9C64-359C5D7585B7@freeswitch.org> References: <7d79b3930911090353n17d64c45id9e9501f13a2bdce@mail.gmail.com> <191c3a030911090834lefa55v5a66ec2982e080b0@mail.gmail.com> <7d79b3930911102010u2777fc6epee2cab59a4f8dfa2@mail.gmail.com> <2E2AE3DF-1EA7-45AA-9C64-359C5D7585B7@freeswitch.org> Message-ID: <7d79b3930911102122jf3f44t7bc035960980678d@mail.gmail.com> What is meant by debug packages. Kindly specify where it is available. On Wed, Nov 11, 2009 at 10:09 AM, Brian West wrote: > You need to install the debug packages so you the symbols because that > backtrace is useless. > /b > > On Nov 10, 2009, at 10:10 PM, lakshmanan ganapathy wrote: > > Here is the required detail. > > http://pastebin.freeswitch.org/11049 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/5f8269d5/attachment-0002.html From lakindia89 at gmail.com Tue Nov 10 21:26:34 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Wed, 11 Nov 2009 10:56:34 +0530 Subject: [Freeswitch-users] Flushing the Event buffer in Perl Event Socket In-Reply-To: <2CD889D1-934A-47CE-A938-1EB9D4325DD2@freeswitch.org> References: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com> <191c3a030910300650w7b80568eu4c41c805b9372acc@mail.gmail.com> <26281493.post@talk.nabble.com> <2CD889D1-934A-47CE-A938-1EB9D4325DD2@freeswitch.org> Message-ID: <7d79b3930911102126u63ba7e8rce1f15ab6371ca2c@mail.gmail.com> That doesn't seems to work for me. Here is my need. I'm using Async in the Event socket outbound. I'll register for "events plain all" I'll answer the call. I'll playback a message. I'll sleep for 5 seconds. After that, I'll receive the events. I don't need the events that are for answer and playback. That action is completed and don't want to receive events for those application. I set $|=1 in my ESL script. But it doesn't seems to solve the above issue. Any help!!!!pls!!! On Tue, Nov 10, 2009 at 7:41 PM, Brian West wrote: > $| = 1; > > I think that is what you're lookin for. > > /b > > On Nov 10, 2009, at 4:51 AM, lakshmanan wrote: > > > I was in a need of flushing the events buffer without reading > > it.I've done > > the following ESL(Async) program to flush the events. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/ef17b7db/attachment-0002.html From andrew at hijacked.us Tue Nov 10 21:38:17 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Wed, 11 Nov 2009 00:38:17 -0500 Subject: [Freeswitch-users] playback from hadoop In-Reply-To: <20ad6b920911101902w52165681tb47fe8f3aa2ae76e@mail.gmail.com> References: <20ad6b920911090459h3e3d02ffv1230800a13f5c06d@mail.gmail.com> <20091109192904.GI9418@hijacked.us> <20ad6b920911100729i23e1f3d4i7a8ced7b2fc526ec@mail.gmail.com> <20091110155633.GA194@hijacked.us> <20ad6b920911101902w52165681tb47fe8f3aa2ae76e@mail.gmail.com> Message-ID: <20091111053816.GA19599@hijacked.us> On Wed, Nov 11, 2009 at 11:02:10AM +0800, mark morreny wrote: > Hi > > Sorry to ask again. > > I know the command to copy file from hadoop file system to somewhere else. > But how do I make a shell command to output raw audio? > What command is it like? Is it like play()? I am confused. > I was very nice and wrote up some documentation (and 2 examples) on the wiki page at http://wiki.freeswitch.org/wiki/Mod_shell_stream Now you know everything I know about using this module (which is a very cool module, by the way - thanks Tony). Andrew From lei.tlfly at gmail.com Wed Nov 11 05:43:58 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 11 Nov 2009 21:43:58 +0800 Subject: [Freeswitch-users] How to test FS rtp packet lost rate? Message-ID: <50c41b4e0911110543m7b5431ecu173d8386073fdb32@mail.gmail.com> Hi all, I'm testing a FS server using sipp, I found that sipp only show the retrans of sip packet, Does someone known is there a tool to test FS rtp packet lost rate in high concurrent call env? -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/9441f535/attachment-0002.html From dome at tel.co.th Wed Nov 11 07:49:09 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 11 Nov 2009 22:49:09 +0700 Subject: [Freeswitch-users] How to test mod_distributor ? Message-ID: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> I found mod_distributor in SVN. I want to know how does it work ? BG Dome C. From kristian.kielhofner at gmail.com Wed Nov 11 08:49:33 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 11:49:33 -0500 Subject: [Freeswitch-users] How to test FS rtp packet lost rate? In-Reply-To: <50c41b4e0911110543m7b5431ecu173d8386073fdb32@mail.gmail.com> References: <50c41b4e0911110543m7b5431ecu173d8386073fdb32@mail.gmail.com> Message-ID: <2d9149cd0911110849u74c2d8d1jb90de8c20cacde9a@mail.gmail.com> The simplest way I know of is to bring up another call from a local phone and listen to the audio. At the same time run tcpdump/etc with a strict filter to capture the rtp to/from that phone. You can then run RTP stream analysis and the like in Wireshark to identify any lost packets. While this obviously won't identify any/all potential lost packets it will be a lot more practical than any of the alternatives: - Capturing all media streams for RTP analysis - Implementing RTCP to identify lost packets - Commercial hardware/software If FreeSWITCH, your machine, or your network are pushed to the max and falling apart you're most likely going to see audio problems on your single (captured) call. On Wed, Nov 11, 2009 at 8:43 AM, Lei Tang wrote: > Hi all, I'm testing a FS server using sipp, I found that sipp only show the > retrans of sip packet, Does someone known is there a tool to test FS rtp > packet lost rate in high concurrent call env? > > -- > Lei.Tang > lei.tlfly at gmail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From juanbackson at gmail.com Wed Nov 11 09:08:45 2009 From: juanbackson at gmail.com (Juan Backson) Date: Thu, 12 Nov 2009 01:08:45 +0800 Subject: [Freeswitch-users] how to rewrite freeswitch SDP Message-ID: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> Hi, I am using 1.0.4 version of freeswitch and I am doing proxy_media for all calls. Basically, I just proxy all media from one gateway to another with freeswitch serving as a middleman. In the outgoing invite, I found that the owner line ( o= ) in SDP is showing the originator's IP which I would like to avoid. Is there anyway to rewirte part of the SDP for the outgoing invite? thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/83d0e05a/attachment-0002.html From kristian.kielhofner at gmail.com Wed Nov 11 09:15:16 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 12:15:16 -0500 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <030D9DFF-7AFE-4942-8BEF-B374F8600396@freeswitch.org> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> <7d0bfd8c0911101348n5d7dfd20p224d972d68a1299d@mail.gmail.com> <030D9DFF-7AFE-4942-8BEF-B374F8600396@freeswitch.org> Message-ID: <2d9149cd0911110915m9468422yc7da8d460e68335@mail.gmail.com> It's also configurable on some phones... As Brian said, welcome to VoIP! ;) On Tue, Nov 10, 2009 at 5:23 PM, Brian West wrote: > That depends on the phone... some let you do it.. some don't... > WELCOME TO VOIP!!! > > /b -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From kristian.kielhofner at gmail.com Wed Nov 11 09:23:07 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 12:23:07 -0500 Subject: [Freeswitch-users] how to rewrite freeswitch SDP In-Reply-To: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> References: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> Message-ID: <2d9149cd0911110923q2cf30d9ehddea6f9d1f96662a@mail.gmail.com> This might be a bit too obvious but unless you have a specific reason to use proxy_media (handling goofy codecs is a big one) you could just set proxy_media=false and FreeSWITH will proxy the media (effectively) and rewrite the entire SDP by default. On Wed, Nov 11, 2009 at 12:08 PM, Juan Backson wrote: > Hi, > > I am using 1.0.4 version of freeswitch and I am doing proxy_media for all > calls.? Basically, I just proxy all media from one gateway to another with > freeswitch serving as a middleman. > > In the outgoing invite, I found that the owner line ( o= ) in SDP is showing > the originator's IP which I would like to avoid. > > Is there anyway to rewirte part of the SDP for the outgoing invite? > > thanks, > jb > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Wed Nov 11 09:29:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Nov 2009 11:29:14 -0600 Subject: [Freeswitch-users] how to rewrite freeswitch SDP In-Reply-To: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> References: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> Message-ID: You use OpenSER /b On Nov 11, 2009, at 11:08 AM, Juan Backson wrote: > Hi, > > I am using 1.0.4 version of freeswitch and I am doing proxy_media > for all calls. Basically, I just proxy all media from one gateway > to another with freeswitch serving as a middleman. > > In the outgoing invite, I found that the owner line ( o= ) in SDP is > showing the originator's IP which I would like to avoid. > > Is there anyway to rewirte part of the SDP for the outgoing invite? > > thanks, > jb From msc at freeswitch.org Wed Nov 11 09:45:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Nov 2009 09:45:21 -0800 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> Message-ID: <87f2f3b90911110945p56104c19m5f7f06dc431a352d@mail.gmail.com> On Wed, Nov 11, 2009 at 7:49 AM, Dome Charoenyost wrote: > I found mod_distributor in SVN. I want to know how does it work ? > It's brand new - I haven't even seen it yet. I will start documenting it shortly. In the meantime if anyone else has started playing with it please let me know. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/85c7058c/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 11 10:32:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Nov 2009 12:32:27 -0600 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> Message-ID: <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> see conf/autoload_configs/distributor.conf.xml in your dialplan you can use ${distributor(test)} which will cycle expanding to foo1 1/10 times and foo2 the other 9 so imagine if foo1 or foo2 were the names of gateways, or hosts of a remote box basic jist is to set total-weight to a number of arbitrary units then set several nodes with weight elements that add up to that number to break down how many times that node text should be returned out of the total. Remember to use the most simplified reduced value for your fractions to get the most variety. Setting total weight to 1000 and then 2 nodes with 100 and 900 would result in foo1 100 times in a row, then foo2 900 times in a row. On Wed, Nov 11, 2009 at 9:49 AM, Dome Charoenyost wrote: > I found mod_distributor in SVN. I want to know how does it work ? > > BG > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/a826214b/attachment-0002.html From christian.loeschenkohl at xpirio.com Wed Nov 11 10:50:52 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Wed, 11 Nov 2009 19:50:52 +0100 Subject: [Freeswitch-users] how to rewrite freeswitch SDP In-Reply-To: <2d9149cd0911110923q2cf30d9ehddea6f9d1f96662a@mail.gmail.com> References: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> <2d9149cd0911110923q2cf30d9ehddea6f9d1f96662a@mail.gmail.com> Message-ID: <4AFB078C.9040008@xpirio.com> hi but this wouldn't work for larger volumens, g729 and t.38 or am i wrong on this? br On 2009-11-11 18:23, Kristian Kielhofner wrote: > This might be a bit too obvious but unless you have a specific reason > to use proxy_media (handling goofy codecs is a big one) you could just > set proxy_media=false and FreeSWITH will proxy the media (effectively) > and rewrite the entire SDP by default. > > On Wed, Nov 11, 2009 at 12:08 PM, Juan Backson wrote: >> Hi, >> >> I am using 1.0.4 version of freeswitch and I am doing proxy_media for all >> calls. Basically, I just proxy all media from one gateway to another with >> freeswitch serving as a middleman. >> >> In the outgoing invite, I found that the owner line ( o= ) in SDP is showing >> the originator's IP which I would like to avoid. >> >> Is there anyway to rewirte part of the SDP for the outgoing invite? >> >> thanks, >> jb >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From djbinter at yahoo.com Wed Nov 11 11:19:12 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 11 Nov 2009 11:19:12 -0800 (PST) Subject: [Freeswitch-users] mod_distributor for bridge Message-ID: <571698.13794.qm@web37508.mail.mud.yahoo.com> Anthony, Would this configuration work if we want to do load sharing 50/50: #distributor.conf.xml ---------------------------------------------------- #sip_profiles/external/carrier1.xml --------------------------------------------------- #dialplan/defalut/01_outbound_routes.xml Thank you, Dorn B. From anthony.minessale at gmail.com Wed Nov 11 11:37:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Nov 2009 13:37:31 -0600 Subject: [Freeswitch-users] mod_distributor for bridge In-Reply-To: <571698.13794.qm@web37508.mail.mud.yahoo.com> References: <571698.13794.qm@web37508.mail.mud.yahoo.com> Message-ID: <191c3a030911111137v232db254v15ffbee1b350c7ab@mail.gmail.com> you got 2 out of 3, the dialplan would look like this: On Wed, Nov 11, 2009 at 1:19 PM, DJB wrote: > Anthony, > > Would this configuration work if we want to do load sharing 50/50: > > > > #distributor.conf.xml > > > > > > > > > > > > ---------------------------------------------------- > #sip_profiles/external/carrier1.xml > > > > > > > > > > > > > > > > --------------------------------------------------- > #dialplan/defalut/01_outbound_routes.xml > > > > > data="${distributor(carrier1)}sofia/gateway/gateway1/$1"/> > > > > > data="${distributor(carrier1)}sofia/gateway/gateway2/$1"/> > > > > > > Thank you, > Dorn B. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/b7e8979d/attachment-0002.html From kristian.kielhofner at gmail.com Wed Nov 11 11:56:05 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 14:56:05 -0500 Subject: [Freeswitch-users] how to rewrite freeswitch SDP In-Reply-To: References: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> Message-ID: <2d9149cd0911111156k3857d644jafd89d687f4dc1aa@mail.gmail.com> This is correct. The nathelper module and RTPProxy have an option to rewrite o= as well as c=. On Wed, Nov 11, 2009 at 12:29 PM, Brian West wrote: > You use OpenSER > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From djbinter at yahoo.com Wed Nov 11 11:57:29 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 11 Nov 2009 11:57:29 -0800 (PST) Subject: [Freeswitch-users] mod_distributor for bridge In-Reply-To: <191c3a030911111137v232db254v15ffbee1b350c7ab@mail.gmail.com> References: <571698.13794.qm@web37508.mail.mud.yahoo.com> <191c3a030911111137v232db254v15ffbee1b350c7ab@mail.gmail.com> Message-ID: <183302.38522.qm@web37501.mail.mud.yahoo.com> Thank you. I will test with the latest SVN; however, can you please advise how do I add it in modules.conf since I don't see the item in there so that I can rebuild the source. Regards, Dorn B. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 11:37:31 AM Subject: Re: [Freeswitch-users] mod_distributor for bridge you got 2 out of 3, the dialplan would look like this: On Wed, Nov 11, 2009 at 1:19 PM, DJB wrote: Anthony, > >>Would this configuration work if we want to do load sharing 50/50: > > > >>#distributor.conf.xml > > >> >> >> >> >> >> >> >> > >>---------------------------------------------------- >>#sip_profiles/external/carrier1.xml > >> >> >> >> >> >> >> >> >> >> >> >> >> > >>--------------------------------------------------- >>#dialplan/defalut/01_outbound_routes.xml > >> >> >> >> >> >> >> >> >> >> >> >> > > >>Thank you, >>Dorn B. > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/40e5b946/attachment-0002.html From kristian.kielhofner at gmail.com Wed Nov 11 11:58:12 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 14:58:12 -0500 Subject: [Freeswitch-users] how to rewrite freeswitch SDP In-Reply-To: <4AFB078C.9040008@xpirio.com> References: <27c25bc40911110908v36b98a42tf3884514a0eed94d@mail.gmail.com> <2d9149cd0911110923q2cf30d9ehddea6f9d1f96662a@mail.gmail.com> <4AFB078C.9040008@xpirio.com> Message-ID: <2d9149cd0911111158g72a2107mf11803a880ba3c69@mail.gmail.com> Brian once told me (at ClueCon) that proxy_media isn't really that much lighter on the CPU. At least that's what I think he said. Anyone care to clarify/quantify? Anyways it would work for G729 (pass through is no problem) but not T.38 (it isn't a recognized codec at all). 2009/11/11 Christian L?schenkohl : > hi > > but this wouldn't work for larger volumens, g729 and t.38 > or am i wrong on this? > > br > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From msc at freeswitch.org Wed Nov 11 12:44:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Nov 2009 12:44:49 -0800 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> Message-ID: <87f2f3b90911111244r130945b7w7813c71fbbd12738@mail.gmail.com> Perfect. I'll have it documented by the end of the day. -MC On Wed, Nov 11, 2009 at 10:32 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > see conf/autoload_configs/distributor.conf.xml > > > > > > > > > > > > > > in your dialplan you can use > > ${distributor(test)} which will cycle expanding to foo1 1/10 times and foo2 > the other 9 > > so imagine if foo1 or foo2 were the names of gateways, or hosts of a remote > box > > basic jist is to set total-weight to a number of arbitrary units then set > several nodes with weight elements that add up to that number to break down > how many times that node text should be returned out of the total. > > Remember to use the most simplified reduced value for your fractions to get > the most variety. > > Setting total weight to 1000 and then 2 nodes with 100 and 900 would result > in foo1 100 times in a row, then foo2 900 times in a row. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/c4f3cea7/attachment-0002.html From djbinter at yahoo.com Wed Nov 11 13:06:39 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 11 Nov 2009 13:06:39 -0800 (PST) Subject: [Freeswitch-users] mod_distributor for bridge In-Reply-To: <183302.38522.qm@web37501.mail.mud.yahoo.com> References: <571698.13794.qm@web37508.mail.mud.yahoo.com> <191c3a030911111137v232db254v15ffbee1b350c7ab@mail.gmail.com> <183302.38522.qm@web37501.mail.mud.yahoo.com> Message-ID: <878657.77487.qm@web37503.mail.mud.yahoo.com> Actually, I got it. I've added: applications/mod_distributor in the modules.conf I will start testing now. Thank you, Dorn B. ________________________________ From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 11:57:29 AM Subject: Re: [Freeswitch-users] mod_distributor for bridge Thank you. I will test with the latest SVN; however, can you please advise how do I add it in modules.conf since I don't see the item in there so that I can rebuild the source. Regards, Dorn B. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 11:37:31 AM Subject: Re: [Freeswitch-users] mod_distributor for bridge you got 2 out of 3, the dialplan would look like this: On Wed, Nov 11, 2009 at 1:19 PM, DJB wrote: Anthony, > >>Would this configuration work if we want to do load sharing 50/50: > > > >>#distributor.conf.xml > > >> >> >> >> >> >> >> >> > >>---------------------------------------------------- >>#sip_profiles/external/carrier1.xml > >> >> >> >> >> >> >> >> >> >> >> >> >> > >>--------------------------------------------------- >>#dialplan/defalut/01_outbound_routes.xml > >> >> >> >> >> >> >> >> >> >> >> >> > > >>Thank you, >>Dorn B. > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/82a39ba5/attachment-0002.html From kristian.kielhofner at gmail.com Wed Nov 11 13:19:47 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 16:19:47 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted Message-ID: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> Full log and trace here: http://pastebin.freeswitch.org/11062 Pretty standard situation. User calls another user (same profile) and tries to place the call on hold (RFC 3264/sendonly). FS places call on hold and tries to start music but ends with: [local_stream://moh] already broadcasting...broadcast aborted ... and we don't get music. Any ideas why? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From djbinter at yahoo.com Wed Nov 11 13:35:28 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 11 Nov 2009 13:35:28 -0800 (PST) Subject: [Freeswitch-users] mod_distributor for bridge In-Reply-To: <878657.77487.qm@web37503.mail.mud.yahoo.com> References: <571698.13794.qm@web37508.mail.mud.yahoo.com> <191c3a030911111137v232db254v15ffbee1b350c7ab@mail.gmail.com> <183302.38522.qm@web37501.mail.mud.yahoo.com> <878657.77487.qm@web37503.mail.mud.yahoo.com> Message-ID: <164694.95836.qm@web37504.mail.mud.yahoo.com> Anthony, I did the test and the load sharing works great. However, I tried to test by failing the first gateway and the load sharing is working correctly, but is there a way that it would fail over and continue to the next one if any of the gateways failed within the list. I've tried with continue_on_fail=true, but it did not work. Thank you, Dorn B. ________________________________ From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 1:06:39 PM Subject: Re: [Freeswitch-users] mod_distributor for bridge Actually, I got it. I've added: applications/mod_distributor in the modules.conf I will start testing now. Thank you, Dorn B. ________________________________ From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 11:57:29 AM Subject: Re: [Freeswitch-users] mod_distributor for bridge Thank you. I will test with the latest SVN; however, can you please advise how do I add it in modules.conf since I don't see the item in there so that I can rebuild the source. Regards, Dorn B. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, November 11, 2009 11:37:31 AM Subject: Re: [Freeswitch-users] mod_distributor for bridge you got 2 out of 3, the dialplan would look like this: On Wed, Nov 11, 2009 at 1:19 PM, DJB wrote: Anthony, > >>Would this configuration work if we want to do load sharing 50/50: > > > >>#distributor.conf.xml > > >> >> >> >> >> >> >> >> > >>---------------------------------------------------- >>#sip_profiles/external/carrier1.xml > >> >> >> >> >> >> >> >> >> >> >> >> >> > >>--------------------------------------------------- >>#dialplan/defalut/01_outbound_routes.xml > >> >> >> >> >> >> >> >> >> >> >> >> > > >>Thank you, >>Dorn B. > > > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/45ddd594/attachment-0002.html From brian at freeswitch.org Wed Nov 11 13:52:27 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Nov 2009 15:52:27 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> Message-ID: I noticed you are sending the call to a socket... what did you do to the call in the socket? /b On Nov 11, 2009, at 3:19 PM, Kristian Kielhofner wrote: > Full log and trace here: > > http://pastebin.freeswitch.org/11062 > > Pretty standard situation. User calls another user (same profile) and > tries to place the call on hold (RFC 3264/sendonly). FS places call > on hold and tries to start music but ends with: > > [local_stream://moh] already broadcasting...broadcast aborted > > ... and we don't get music. Any ideas why? > > Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/577cc858/attachment-0002.html From federico.omoto at gmail.com Wed Nov 11 13:58:28 2009 From: federico.omoto at gmail.com (Fede) Date: Wed, 11 Nov 2009 19:58:28 -0200 Subject: [Freeswitch-users] Unable to register UA Message-ID: <8b4221f20911111358u10502dcdva9382bed13cb81a6@mail.gmail.com> Hi! I'm trying to register a SIP UA to my FreeSWITCH server and for some reason I always get a "401 Unauthorized" response. I've tried with other UA (X-Lite and Ekiga) and they do work. The UA is: http://www.doddlephone.com My user configuration is: Can someone help me and tell me what I'm doing wrong? Here's the FreeSWITCH trace if it's useful: freeswitch at fc1160102.aspadmin.net> tport_wakeup_pri(0xae7056b0): events IN tport_recv_event(0xae7056b0) tport_recv_iovec(0xae7056b0) msg 0xae703f88 from (udp/216.75.60.102:5060) has 444 bytes, veclen = 1 recv 444 bytes from udp/[190.179.3.18]:4375 at 21:51:10.194004: ------------------------------------------------------------------------ REGISTER sip:216.75.60.102 SIP/2.0 From: sip:doddle at 216.75.60.102 ;tag=633f3915 To: sip:doddle at 216.75.60.102 Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 101 REGISTER Contact: Expires: 3600 Date: Wed, 11 Nov 2009 21:51:51 GMT Max-Forwards: 70 User-Agent: Doddle WebPhone Supported: replaces Via: SIP/2.0/UDP 192.168.0.1:4375;branch=z9hG4bK-0f9263f10caa;rport Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0xae7056b0): msg 0xae703f88 (444 bytes) from udp/ 190.179.3.18:5060/sip next=(nil) nta: received REGISTER sip:216.75.60.102 SIP/2.0 (CSeq 101) nta: Via check: received=190.179.3.18 nta: canonizing sip:216.75.60.102 with contact nta: REGISTER (101) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0xae7098a0, 0xae704890, 0xae718fc8) called soa_set_params(static::0xae719398, ...) called nua: nua_application_event: entering nua: nua_respond: entering nua(0xae718fc8): sent signal r_respond nua: nua_handle_destroy: entering nua(0xae718fc8): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering soa_set_params(static::0xae719398, ...) called tport_tsend(0xae7056b0) tpn = UDP/190.179.3.18:4375 tport_resolve addrinfo = 190.179.3.18:4375 tport_by_addrinfo(0xae7056b0): not found by name UDP/190.179.3.18:4375 tport_vsend returned 631 send 631 bytes to udp/[190.179.3.18]:4375 at 21:51:10.198733: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.1:4375 ;branch=z9hG4bK-0f9263f10caa;rport=4375;received=190.179.3.18 From: sip:doddle at 216.75.60.102 ;tag=633f3915 To: >;tag=U8QpcZyvrQ3Fg Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 101 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.5pre5-15326M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="216.75.60.102", nonce="132239a5-e37e-4698-af61-5df9b3b67de8", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ nta: sent 401 Unauthorized for REGISTER (101) nta: timer set to 32000 ms nta_leg_destroy((nil)) soa_destroy(static::0xae719398) called tport_wakeup_pri(0xae7056b0): events IN tport_recv_event(0xae7056b0) tport_recv_iovec(0xae7056b0) msg 0xae716af8 from (udp/216.75.60.102:5060) has 706 bytes, veclen = 1 recv 706 bytes from udp/[190.179.3.18]:4375 at 21:51:10.453646: ------------------------------------------------------------------------ REGISTER sip:216.75.60.102 SIP/2.0 From: sip:doddle at 216.75.60.102 ;tag=633f3915 To: sip:doddle at 216.75.60.102 Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 102 REGISTER Expires: 3600 Date: Wed, 11 Nov 2009 21:51:51 GMT Max-Forwards: 70 User-Agent: Doddle WebPhone Supported: replaces Authorization: Digest username="doddle", realm="216.75.60.102", nonce="132239a5-e37e-4698-af61-5df9b3b67de8", uri="sip:216.75.60.102", response="686ec180c04fc70be22aaca9eb21f5e9", algorithm=MD5, cnonce="155fb1307a89ffc0eaed1e0e94958e6e", qop=auth, nc=00000033 Via: SIP/2.0/UDP 192.168.0.1:4375;branch=z9hG4bK-36a1b6039a42;rport Contact: Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0xae7056b0): msg 0xae716af8 (706 bytes) from udp/ 190.179.3.18:5060/sip next=(nil) nta: received REGISTER sip:216.75.60.102 SIP/2.0 (CSeq 102) nta: Via check: received=190.179.3.18 nta: canonizing sip:216.75.60.102 with contact nta: REGISTER (102) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0xae7098a0, 0xae704890, 0xae71ae30) called soa_set_params(static::0xae71b340, ...) called nua: nua_application_event: entering nua: nua_respond: entering nua(0xae71ae30): sent signal r_respond nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0xae71ae30): sent signal r_destroy nua: nua_handle_magic: entering soa_set_params(static::0xae71b340, ...) called nua: nua_handle_destroy: entering tport_tsend(0xae7056b0) tpn = UDP/190.179.3.18:4375 tport_resolve addrinfo = 190.179.3.18:4375 tport_by_addrinfo(0xae7056b0): not found by name UDP/190.179.3.18:4375 tport_vsend returned 645 send 645 bytes to udp/[190.179.3.18]:4375 at 21:51:10.462570: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.1:4375 ;branch=z9hG4bK-36a1b6039a42;rport=4375;received=190.179.3.18 From: sip:doddle at 216.75.60.102 ;tag=633f3915 To: >;tag=vHHFetF0N0S2B Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 102 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.5pre5-15326M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="216.75.60.102", nonce="5a5ebe31-baf6-429c-a184-f835a22d7c24", stale="true", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ nta: sent 401 Unauthorized for REGISTER (102) nta_leg_destroy((nil)) soa_destroy(static::0xae71b340) called Thank you in advacen, Federico Omoto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/c670d981/attachment-0002.html From kristian.kielhofner at gmail.com Wed Nov 11 14:20:40 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 17:20:40 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> Message-ID: <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> >From the trace: # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr.c:540 sofia/pjsip/nobody at 192.168.4.192 Command Execute set(sip_h_X-voalte-call-id=9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec) # EXECUTE sofia/pjsip/nobody at 192.168.4.192 set(sip_h_X-voalte-call-id=9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec) # 2009-11-11 11:23:58.909804 [DEBUG] mod_dptools.c:766 sofia/pjsip/nobody at 192.168.4.192 SET [sip_h_X-voalte-call-id]=[9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr.c:540 sofia/pjsip/nobody at 192.168.4.192 Command Execute ring_ready() # EXECUTE sofia/pjsip/nobody at 192.168.4.192 ring_ready() # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr.c:540 sofia/pjsip/nobody at 192.168.4.192 Command Execute bridge({originate_timeout=30,bypass_media=false,origination_caller_id_number=1001,origination_caller_id_name=Danielle Reed}sofia/voalte/huttoj at 192.168.4.180) # EXECUTE sofia/pjsip/nobody at 192.168.4.192 bridge({originate_timeout=30,bypass_media=false,origination_caller_id_number=1001,origination_caller_id_name=Danielle Reed}sofia/voalte/huttoj at 192.168.4.180) # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 0 = [originate_timeout=30] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 1 = [bypass_media=false] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 2 = [origination_caller_id_number=1001] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 3 = [origination_caller_id_name=Danielle Reed] # 2009-11-11 11:23:58.909804 [NOTICE] switch_channel.c:613 New Channel sofia/pjsip/huttoj at 192.168.4.180 [66e0041f-feff-49c1-baf8-0e5aa1ae99fa] 1) Set a SIP header (this is being removed, actually). 2) Indicate ring_ready (it's a long story). 3) Execute bridge to call the other device. On Wed, Nov 11, 2009 at 4:52 PM, Brian West wrote: > I noticed you are sending the call to a socket... what did you do to the > call in the socket? > /b -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From Prometheus001 at gmx.net Wed Nov 11 14:27:09 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 11 Nov 2009 23:27:09 +0100 Subject: [Freeswitch-users] att_xfer and Loopback Message-ID: <4AFB3A3D.1050602@gmx.net> Hello, I have some problems with attended transfer and loopback Scenario how id work - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped How it work until here: - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, B and C are connected (A still MOH) The dial string for C is dynamic and dependent on certain parameters, therefore C must be called via Loopback in our scenario. Here are the configs: In dialplan for calling B: Dialplan for executing the att_xfer: So this is pretty standard, except the loopback. SVN is 15322. Anybody has a solution for this? Best regards Peter From msc at freeswitch.org Wed Nov 11 14:31:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Nov 2009 14:31:57 -0800 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <87f2f3b90911111244r130945b7w7813c71fbbd12738@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> <87f2f3b90911111244r130945b7w7813c71fbbd12738@mail.gmail.com> Message-ID: <87f2f3b90911111431y7b68856cwfe8a97ec0754f86d@mail.gmail.com> FYI, I added some docs here: http://wiki.freeswitch.org/wiki/Mod_distributor Please feel free to add to it if you are doing anything interesting or creative that hasn't been covered. -MC On Wed, Nov 11, 2009 at 12:44 PM, Michael Collins wrote: > Perfect. I'll have it documented by the end of the day. > -MC > > > On Wed, Nov 11, 2009 at 10:32 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> see conf/autoload_configs/distributor.conf.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> in your dialplan you can use >> >> ${distributor(test)} which will cycle expanding to foo1 1/10 times and >> foo2 the other 9 >> >> so imagine if foo1 or foo2 were the names of gateways, or hosts of a >> remote box >> >> basic jist is to set total-weight to a number of arbitrary units then set >> several nodes with weight elements that add up to that number to break down >> how many times that node text should be returned out of the total. >> >> Remember to use the most simplified reduced value for your fractions to >> get the most variety. >> >> Setting total weight to 1000 and then 2 nodes with 100 and 900 would >> result in foo1 100 times in a row, then foo2 900 times in a row. >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/86b3835a/attachment-0002.html From kristian.kielhofner at gmail.com Wed Nov 11 14:33:15 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Nov 2009 17:33:15 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> Message-ID: <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64. On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner wrote: > From the trace: > ..snip.. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From sergey.kobzar at mail.ru Wed Nov 11 14:33:52 2009 From: sergey.kobzar at mail.ru (Sergey Kobzar) Date: Thu, 12 Nov 2009 00:33:52 +0200 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: References: <1352396721.20091110232720@mail.ru> Message-ID: <633434680.20091112003352@mail.ru> Mathieu, thanks for the help. I got external oubound calls working. The things are simpler then I expected. This is my configuration: I still have 2 questions: 1. Users must type '9' at the beginning, which means this is external call and it must go out through VoIP provider. My config: ... But I see that 9 still exists. 2. Ideally each internal number must have external one. In other words ${outbound_caller_id_number} must be mapped to int. number. Where can I do this? P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use FS default configuration for testing. Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote: > As easy as: > > in your dialplan. If you want to make a gateway out of it, you can > enter whatever you want in username and password since they won't be > used. (SIP works using challenge authentication which means the remote > UA has to send you a packet requesting the credentials). > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote: >> Hello. >> >> I'm FS newbie and want connect it to SIP provider which does not >> require authentication - it make authentication using my IP. >> >> I've searched through FS documentation and didn't find clear answer. >> >> Could you help me or maybe give a link to a doc which can help? >> >> Thanks. >> >> >> -- >> Sergey >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sergey From mrene_lists at avgs.ca Wed Nov 11 14:41:22 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 11 Nov 2009 14:41:22 -0800 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: <633434680.20091112003352@mail.ru> References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> Message-ID: $1 gives you the content of the first regex capture group, so the first ( ) group. ^9(\d{7,})$ would put it in $1 Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Nov-09, at 2:33 PM, Sergey Kobzar wrote: > Mathieu, thanks for the help. I got external oubound calls working. > The things are simpler then I expected. > > This is my configuration: > > > > > > > > > > > I still have 2 questions: > > 1. Users must type '9' at the beginning, which means this is external > call and it must go out through VoIP provider. My config: > > ... > > > But I see that 9 still exists. > > > 2. Ideally each internal number must have external one. In other words > ${outbound_caller_id_number} must be mapped to int. number. Where > can I do this? > > > P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use FS > default configuration for testing. > > > > Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote: > >> As easy as: >> > >> in your dialplan. If you want to make a gateway out of it, you can >> enter whatever you want in username and password since they won't be >> used. (SIP works using challenge authentication which means the >> remote >> UA has to send you a packet requesting the credentials). > >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca > > > > >> On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote: > >>> Hello. >>> >>> I'm FS newbie and want connect it to SIP provider which does not >>> require authentication - it make authentication using my IP. >>> >>> I've searched through FS documentation and didn't find clear answer. >>> >>> Could you help me or maybe give a link to a doc which can help? >>> >>> Thanks. >>> >>> >>> -- >>> Sergey >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Sergey > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Russell.Mosemann at cune.org Wed Nov 11 14:43:20 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Wed, 11 Nov 2009 22:43:20 -0000 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: <633434680.20091112003352@mail.ru> Message-ID: <20091111224320.1A44337F22E@mail.cune.org> > > ... > But I see that 9 still exists. Put the parentheses around the portion you want to capture. http://wiki.freeswitch.org/wiki/Regular_Expression -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From sergey.kobzar at mail.ru Wed Nov 11 15:04:12 2009 From: sergey.kobzar at mail.ru (Sergey Kobzar) Date: Thu, 12 Nov 2009 01:04:12 +0200 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> Message-ID: <1952719808.20091112010412@mail.ru> Ah, right. I was inattentive :) What about my 2nd question? Each user must have unique outbound number which is mapped to his internal number. How can I set ${outbound_caller_id_number} depending on calling internal number? Thursday, November 12, 2009, 12:41:22 AM, Mathieu wrote: > $1 gives you the content of the first regex capture group, so the > first ( ) group. > ^9(\d{7,})$ would put it in $1 > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > On 11-Nov-09, at 2:33 PM, Sergey Kobzar wrote: >> Mathieu, thanks for the help. I got external oubound calls working. >> The things are simpler then I expected. >> >> This is my configuration: >> >> >> >> >> >> >> >> >> >> >> I still have 2 questions: >> >> 1. Users must type '9' at the beginning, which means this is external >> call and it must go out through VoIP provider. My config: >> >> ... >> >> >> But I see that 9 still exists. >> >> >> 2. Ideally each internal number must have external one. In other words >> ${outbound_caller_id_number} must be mapped to int. number. Where >> can I do this? >> >> >> P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use FS >> default configuration for testing. >> >> >> >> Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote: >> >>> As easy as: >>> >> >>> in your dialplan. If you want to make a gateway out of it, you can >>> enter whatever you want in username and password since they won't be >>> used. (SIP works using challenge authentication which means the >>> remote >>> UA has to send you a packet requesting the credentials). >> >>> Mathieu Rene >>> Avant-Garde Solutions Inc >>> Office: + 1 (514) 664-1044 x100 >>> Cell: +1 (514) 664-1044 x200 >>> mrene at avgs.ca >> >> >> >> >>> On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote: >> >>>> Hello. >>>> >>>> I'm FS newbie and want connect it to SIP provider which does not >>>> require authentication - it make authentication using my IP. >>>> >>>> I've searched through FS documentation and didn't find clear answer. >>>> >>>> Could you help me or maybe give a link to a doc which can help? >>>> >>>> Thanks. >>>> >>>> >>>> -- >>>> Sergey >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >> >> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> >> -- >> Sergey >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sergey From mrene_lists at avgs.ca Wed Nov 11 15:09:30 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 11 Nov 2009 15:09:30 -0800 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: <1952719808.20091112010412@mail.ru> References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> Message-ID: Set it in the user directory entry. All variables all loaded whenever the user is authenticated (before the call hits the dialplan) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Nov-09, at 3:04 PM, Sergey Kobzar wrote: > Ah, right. I was inattentive :) > > What about my 2nd question? Each user must have unique outbound number > which is mapped to his internal number. > > How can I set ${outbound_caller_id_number} depending on calling > internal number? > > > > Thursday, November 12, 2009, 12:41:22 AM, Mathieu wrote: > >> $1 gives you the content of the first regex capture group, so the >> first ( ) group. > >> ^9(\d{7,})$ would put it in $1 > >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca > > > > >> On 11-Nov-09, at 2:33 PM, Sergey Kobzar wrote: > >>> Mathieu, thanks for the help. I got external oubound calls working. >>> The things are simpler then I expected. >>> >>> This is my configuration: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I still have 2 questions: >>> >>> 1. Users must type '9' at the beginning, which means this is >>> external >>> call and it must go out through VoIP provider. My config: >>> >>> ... >>> >>> >>> But I see that 9 still exists. >>> >>> >>> 2. Ideally each internal number must have external one. In other >>> words >>> ${outbound_caller_id_number} must be mapped to int. number. Where >>> can I do this? >>> >>> >>> P.S. I try to move from Asterisk + Cisco CME to FreeSWITCH and use >>> FS >>> default configuration for testing. >>> >>> >>> >>> Tuesday, November 10, 2009, 11:43:04 PM, Mathieu wrote: >>> >>>> As easy as: >>>> >>> >>>> in your dialplan. If you want to make a gateway out of it, you can >>>> enter whatever you want in username and password since they won't >>>> be >>>> used. (SIP works using challenge authentication which means the >>>> remote >>>> UA has to send you a packet requesting the credentials). >>> >>>> Mathieu Rene >>>> Avant-Garde Solutions Inc >>>> Office: + 1 (514) 664-1044 x100 >>>> Cell: +1 (514) 664-1044 x200 >>>> mrene at avgs.ca >>> >>> >>> >>> >>>> On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote: >>> >>>>> Hello. >>>>> >>>>> I'm FS newbie and want connect it to SIP provider which does not >>>>> require authentication - it make authentication using my IP. >>>>> >>>>> I've searched through FS documentation and didn't find clear >>>>> answer. >>>>> >>>>> Could you help me or maybe give a link to a doc which can help? >>>>> >>>>> Thanks. >>>>> >>>>> >>>>> -- >>>>> Sergey >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>> >>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Sergey >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Sergey > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris at cloudtel.com Wed Nov 11 17:16:50 2009 From: chris at cloudtel.com (Chris Burns) Date: Wed, 11 Nov 2009 20:16:50 -0500 Subject: [Freeswitch-users] Unable to register UA In-Reply-To: <8b4221f20911111358u10502dcdva9382bed13cb81a6@mail.gmail.com> References: <8b4221f20911111358u10502dcdva9382bed13cb81a6@mail.gmail.com> Message-ID: <200911112016.50824.chris@cloudtel.com> Your SIP UA needs to take the info in the 401 and use it to digest authenticate. If you trace a SIP UA that supports authentication you will see that they also get the 401/407 and only then are able to authenticate. This is just a fact of how digest auth works in SIP ... see section 22.4 The Digest Authentication Scheme: http://www.ietf.org/rfc/rfc3261.txt On November 11, 2009 04:58:28 pm Fede wrote: > Hi! > > I'm trying to register a SIP UA to my FreeSWITCH server and for some reason > I always get a "401 Unauthorized" response. I've tried with other UA > (X-Lite and Ekiga) and they do work. The UA is: http://www.doddlephone.com > > My user configuration is: > > > > > > > > > > > > > > > > > > Can someone help me and tell me what I'm doing wrong? > > Here's the FreeSWITCH trace if it's useful: > > freeswitch at fc1160102.aspadmin.net> tport_wakeup_pri(0xae7056b0): events IN > tport_recv_event(0xae7056b0) > tport_recv_iovec(0xae7056b0) msg 0xae703f88 from (udp/216.75.60.102:5060) > has 444 bytes, veclen = 1 > recv 444 bytes from udp/[190.179.3.18]:4375 at 21:51:10.194004: > ------------------------------------------------------------------------ > REGISTER sip:216.75.60.102 SIP/2.0 > From: sip:doddle at 216.75.60.102 ;tag=633f3915 > To: sip:doddle at 216.75.60.102 > Call-Id: 186e708700bcf9a944855105fc3dce0e > Cseq: 101 REGISTER > Contact: > Expires: 3600 > Date: Wed, 11 Nov 2009 21:51:51 GMT > Max-Forwards: 70 > User-Agent: Doddle WebPhone > Supported: replaces > Via: SIP/2.0/UDP 192.168.0.1:4375;branch=z9hG4bK-0f9263f10caa;rport > Content-Length: 0 > > ------------------------------------------------------------------------ > tport_deliver(0xae7056b0): msg 0xae703f88 (444 bytes) from udp/ > 190.179.3.18:5060/sip next=(nil) > nta: received REGISTER sip:216.75.60.102 SIP/2.0 (CSeq 101) > nta: Via check: received=190.179.3.18 > nta: canonizing sip:216.75.60.102 with contact > nta: REGISTER (101) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > soa_clone(static::0xae7098a0, 0xae704890, 0xae718fc8) called > soa_set_params(static::0xae719398, ...) called > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0xae718fc8): sent signal r_respond > nua: nua_handle_destroy: entering > nua(0xae718fc8): sent signal r_destroy > nua: nua_stack_set_params: entering > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > soa_set_params(static::0xae719398, ...) called > tport_tsend(0xae7056b0) tpn = UDP/190.179.3.18:4375 > tport_resolve addrinfo = 190.179.3.18:4375 > tport_by_addrinfo(0xae7056b0): not found by name UDP/190.179.3.18:4375 > tport_vsend returned 631 > send 631 bytes to udp/[190.179.3.18]:4375 at 21:51:10.198733: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.0.1:4375 > ;branch=z9hG4bK-0f9263f10caa;rport=4375;received=190.179.3.18 > From: sip:doddle at 216.75.60.102 ;tag=633f3915 > To: > > >;tag=U8QpcZyvrQ3Fg > > Call-Id: 186e708700bcf9a944855105fc3dce0e > Cseq: 101 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.0.5pre5-15326M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="216.75.60.102", > nonce="132239a5-e37e-4698-af61-5df9b3b67de8", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > nta: sent 401 Unauthorized for REGISTER (101) > nta: timer set to 32000 ms > nta_leg_destroy((nil)) > soa_destroy(static::0xae719398) called > tport_wakeup_pri(0xae7056b0): events IN > tport_recv_event(0xae7056b0) > tport_recv_iovec(0xae7056b0) msg 0xae716af8 from (udp/216.75.60.102:5060) > has 706 bytes, veclen = 1 > recv 706 bytes from udp/[190.179.3.18]:4375 at 21:51:10.453646: > ------------------------------------------------------------------------ > REGISTER sip:216.75.60.102 SIP/2.0 > From: sip:doddle at 216.75.60.102 ;tag=633f3915 > To: sip:doddle at 216.75.60.102 > Call-Id: 186e708700bcf9a944855105fc3dce0e > Cseq: 102 REGISTER > Expires: 3600 > Date: Wed, 11 Nov 2009 21:51:51 GMT > Max-Forwards: 70 > User-Agent: Doddle WebPhone > Supported: replaces > Authorization: Digest username="doddle", realm="216.75.60.102", > nonce="132239a5-e37e-4698-af61-5df9b3b67de8", uri="sip:216.75.60.102", > response="686ec180c04fc70be22aaca9eb21f5e9", algorithm=MD5, > cnonce="155fb1307a89ffc0eaed1e0e94958e6e", qop=auth, nc=00000033 > Via: SIP/2.0/UDP 192.168.0.1:4375;branch=z9hG4bK-36a1b6039a42;rport > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > tport_deliver(0xae7056b0): msg 0xae716af8 (706 bytes) from udp/ > 190.179.3.18:5060/sip next=(nil) > nta: received REGISTER sip:216.75.60.102 SIP/2.0 (CSeq 102) > nta: Via check: received=190.179.3.18 > nta: canonizing sip:216.75.60.102 with contact > nta: REGISTER (102) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > soa_clone(static::0xae7098a0, 0xae704890, 0xae71ae30) called > soa_set_params(static::0xae71b340, ...) called > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0xae71ae30): sent signal r_respond > nua: nua_handle_destroy: entering > nua: nua_stack_set_params: entering > nua(0xae71ae30): sent signal r_destroy > nua: nua_handle_magic: entering > soa_set_params(static::0xae71b340, ...) called > nua: nua_handle_destroy: entering > tport_tsend(0xae7056b0) tpn = UDP/190.179.3.18:4375 > tport_resolve addrinfo = 190.179.3.18:4375 > tport_by_addrinfo(0xae7056b0): not found by name UDP/190.179.3.18:4375 > tport_vsend returned 645 > send 645 bytes to udp/[190.179.3.18]:4375 at 21:51:10.462570: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.0.1:4375 > ;branch=z9hG4bK-36a1b6039a42;rport=4375;received=190.179.3.18 > From: sip:doddle at 216.75.60.102 ;tag=633f3915 > To: > > >;tag=vHHFetF0N0S2B > > Call-Id: 186e708700bcf9a944855105fc3dce0e > Cseq: 102 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.0.5pre5-15326M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="216.75.60.102", > nonce="5a5ebe31-baf6-429c-a184-f835a22d7c24", stale="true", algorithm=MD5, > qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > nta: sent 401 Unauthorized for REGISTER (102) > nta_leg_destroy((nil)) > soa_destroy(static::0xae71b340) called > > > > Thank you in advacen, > > Federico Omoto From lists at redbonez.net Wed Nov 11 16:20:37 2009 From: lists at redbonez.net (Adam Ford) Date: Wed, 11 Nov 2009 17:20:37 -0700 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> Message-ID: <00eb01ca632d$f719ecf0$e54dc6d0$@net> Hi everybody, I have setup a FreeSWITCH IP-PBX for my office using a T1 and Redfone foneBridge2, which uses Openzap, for my connection to the PSTN. I am trying to figure out if it is possible to forward a call that comes in through the T1/Openzap, back out to a PSTN number. An example would be, I am going to be out of the office and need to forward my office line to my cell phone. I have read what I could find in the wiki about redirect and deflect, but that appears to only deal with SIP providers/gateways. Can I use these applications with Openzap? And if so, what is the syntax for doing so? I have also noted that I can simply bridge the call out another line on the T1 through Openzap. However, that seems to tie up 2 lines just to forward a call. This is not a desirable solution. Thanks to anyone who can help. -Adam From Russell.Mosemann at cune.org Wed Nov 11 17:36:21 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Wed, 11 Nov 2009 19:36:21 -0600 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: <00eb01ca632d$f719ecf0$e54dc6d0$@net> References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> <00eb01ca632d$f719ecf0$e54dc6d0$@net> Message-ID: Adam Ford wrote: > I have also noted that I can simply bridge the call out another line on > the T1 through Openzap. However, that seems to tie up 2 lines just to > forward a call. This is not a desirable solution. That's the way it has to work with any phone system, including your cell phone. If your cell phone provider received a call for you and you had forwarded cell phone calls to another number, your cell phone provider would have to route the incoming call out another line to the next destination. That takes two lines (or channels). That's what forwarding means. One incoming call bridged to one outgoing call. -- Russell Mosemann From lists at redbonez.net Wed Nov 11 17:43:22 2009 From: lists at redbonez.net (Adam Ford) Date: Wed, 11 Nov 2009 18:43:22 -0700 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> <00eb01ca632d$f719ecf0$e54dc6d0$@net> Message-ID: <010401ca6339$860f0f20$922d2d60$@net> Alright. Thank you for your answer. I just had hoped there might be something better that I didn't know about, after reading about deflect on the wiki. -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Russell Mosemann Sent: Wednesday, November 11, 2009 6:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP Adam Ford wrote: > I have also noted that I can simply bridge the call out another line on > the T1 through Openzap. However, that seems to tie up 2 lines just to > forward a call. This is not a desirable solution. That's the way it has to work with any phone system, including your cell phone. If your cell phone provider received a call for you and you had forwarded cell phone calls to another number, your cell phone provider would have to route the incoming call out another line to the next destination. That takes two lines (or channels). That's what forwarding means. One incoming call bridged to one outgoing call. -- Russell Mosemann _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mrene_lists at avgs.ca Wed Nov 11 17:45:49 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 11 Nov 2009 17:45:49 -0800 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: <010401ca6339$860f0f20$922d2d60$@net> References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> <00eb01ca632d$f719ecf0$e54dc6d0$@net> <010401ca6339$860f0f20$922d2d60$@net> Message-ID: <061C000F-5E5D-4C30-875D-860F7162F61D@avgs.ca> There is something called 2B Channel Transfer that can make 2 channels of the same span be released, but providers don't always implement it. Im not exactly sure what kind of Q931 message we need to send down the TDM circuit though. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Nov-09, at 5:43 PM, Adam Ford wrote: > Alright. Thank you for your answer. I just had hoped there might be > something better that I didn't know about, after reading about > deflect on > the wiki. > > -Adam > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Russell > Mosemann > Sent: Wednesday, November 11, 2009 6:36 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Forwarding calls to an outside > number - > OpenZAP > > Adam Ford wrote: > >> I have also noted that I can simply bridge the call out another >> line on >> the T1 through Openzap. However, that seems to tie up 2 lines just to >> forward a call. This is not a desirable solution. > > That's the way it has to work with any phone system, including your > cell > phone. If your cell phone provider received a call for you and you had > forwarded cell phone calls to another number, your cell phone > provider would > have to route the incoming call out another line to the next > destination. > That takes two lines (or channels). That's what forwarding means. One > incoming call bridged to one outgoing call. > > -- > Russell Mosemann > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Nov 11 17:49:48 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Nov 2009 19:49:48 -0600 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: <010401ca6339$860f0f20$922d2d60$@net> References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> <00eb01ca632d$f719ecf0$e54dc6d0$@net> <010401ca6339$860f0f20$922d2d60$@net> Message-ID: deflect would work if the stack and your provider supported TBCT /b On Nov 11, 2009, at 7:43 PM, Adam Ford wrote: > Alright. Thank you for your answer. I just had hoped there might be > something better that I didn't know about, after reading about > deflect on > the wiki. > > -Adam From peter at cindyandpeter.com Wed Nov 11 18:16:59 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Wed, 11 Nov 2009 21:16:59 -0500 Subject: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP In-Reply-To: References: <1352396721.20091110232720@mail.ru> <633434680.20091112003352@mail.ru> <1952719808.20091112010412@mail.ru> <00eb01ca632d$f719ecf0$e54dc6d0$@net> <010401ca6339$860f0f20$922d2d60$@net> Message-ID: <029b01ca633e$373828f0$a5a87ad0$@com> FWIW: If you're getting your T1 from Qwest, they have an option called TnR (Transfer and Release). You play a DTMF sequence followed by the destination number. They take the call back and bridge it to the new number. Of course, you pay to have the feature enabled and you pay per use. It's only SLIGHTLY cheaper than using two lines. Peter -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, November 11, 2009 8:50 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Forwarding calls to an outside number - OpenZAP deflect would work if the stack and your provider supported TBCT /b On Nov 11, 2009, at 7:43 PM, Adam Ford wrote: > Alright. Thank you for your answer. I just had hoped there might be > something better that I didn't know about, after reading about > deflect on > the wiki. > > -Adam _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dome at tel.co.th Wed Nov 11 18:28:52 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 12 Nov 2009 09:28:52 +0700 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <87f2f3b90911111431y7b68856cwfe8a97ec0754f86d@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> <87f2f3b90911111244r130945b7w7813c71fbbd12738@mail.gmail.com> <87f2f3b90911111431y7b68856cwfe8a97ec0754f86d@mail.gmail.com> Message-ID: <8ccbff060911111828q1aa92ff2kf279235983f25275@mail.gmail.com> Wow. we can use FS for sip dispatcher :) How to forward call in FS ? i mean 302 redirect not bridge ? BG Dome C. 2009/11/12 Michael Collins : > FYI, I added some docs here: > http://wiki.freeswitch.org/wiki/Mod_distributor > > Please feel free to add to it if you are doing anything interesting or > creative that hasn't been covered. > -MC > > On Wed, Nov 11, 2009 at 12:44 PM, Michael Collins > wrote: >> >> Perfect. I'll have it documented by the end of the day. >> -MC >> >> On Wed, Nov 11, 2009 at 10:32 AM, Anthony Minessale >> wrote: >>> >>> see conf/autoload_configs/distributor.conf.xml >>> >>> >>> ? >>> ??? >>> ??? >>> ??? >>> ????? >>> ????? >>> ??? >>> ? >>> >>> >>> >>> in your dialplan you can use >>> >>> ${distributor(test)} which will cycle expanding to foo1 1/10 times and >>> foo2 the other 9 >>> >>> so imagine if foo1 or foo2 were the names of gateways, or hosts of a >>> remote box >>> >>> basic jist is to set total-weight to a number of arbitrary units then set >>> several nodes with weight elements that add up to that number to break down >>> how many times that node text should be returned out of the total. >>> >>> Remember to use the most simplified reduced value for your fractions to >>> get the most variety. >>> >>> Setting total weight to 1000 and then 2 nodes with 100 and 900 would >>> result in foo1 100 times in a row, then foo2 900 times in a row. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dome at tel.co.th Wed Nov 11 18:42:42 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 12 Nov 2009 09:42:42 +0700 Subject: [Freeswitch-users] How to test mod_distributor ? In-Reply-To: <8ccbff060911111828q1aa92ff2kf279235983f25275@mail.gmail.com> References: <8ccbff060911110749j604d6e54v93b0caaa4329d8a@mail.gmail.com> <191c3a030911111032l1163f4efpcd8b462f319c83be@mail.gmail.com> <87f2f3b90911111244r130945b7w7813c71fbbd12738@mail.gmail.com> <87f2f3b90911111431y7b68856cwfe8a97ec0754f86d@mail.gmail.com> <8ccbff060911111828q1aa92ff2kf279235983f25275@mail.gmail.com> Message-ID: <8ccbff060911111842g538e1d2fxe6177e4f2d56c8aa@mail.gmail.com> Got it from wiki Dome C. 2009/11/12 Dome Charoenyost : > Wow. we can use FS for sip dispatcher :) > How to forward call in FS ? i mean 302 redirect not bridge ? > > > BG > > Dome C. > > > 2009/11/12 Michael Collins : >> FYI, I added some docs here: >> http://wiki.freeswitch.org/wiki/Mod_distributor >> >> Please feel free to add to it if you are doing anything interesting or >> creative that hasn't been covered. >> -MC >> >> On Wed, Nov 11, 2009 at 12:44 PM, Michael Collins >> wrote: >>> >>> Perfect. I'll have it documented by the end of the day. >>> -MC >>> >>> On Wed, Nov 11, 2009 at 10:32 AM, Anthony Minessale >>> wrote: >>>> >>>> see conf/autoload_configs/distributor.conf.xml >>>> >>>> >>>> ? >>>> ??? >>>> ??? >>>> ??? >>>> ????? >>>> ????? >>>> ??? >>>> ? >>>> >>>> >>>> >>>> in your dialplan you can use >>>> >>>> ${distributor(test)} which will cycle expanding to foo1 1/10 times and >>>> foo2 the other 9 >>>> >>>> so imagine if foo1 or foo2 were the names of gateways, or hosts of a >>>> remote box >>>> >>>> basic jist is to set total-weight to a number of arbitrary units then set >>>> several nodes with weight elements that add up to that number to break down >>>> how many times that node text should be returned out of the total. >>>> >>>> Remember to use the most simplified reduced value for your fractions to >>>> get the most variety. >>>> >>>> Setting total weight to 1000 and then 2 nodes with 100 and 900 would >>>> result in foo1 100 times in a row, then foo2 900 times in a row. >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From anthony.minessale at gmail.com Wed Nov 11 19:14:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Nov 2009 21:14:23 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> Message-ID: <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> dont execute bridge that way, your bridge itself is the other thing already broadcasting. api uuid_transfer bridge:sofia/myprofile/foo at bar.com inline if you want to do more after the bridge set the variable park_after_bridge=true to make it go back to idle On Wed, Nov 11, 2009 at 4:33 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64. > > On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner > wrote: > > From the trace: > > > ..snip.. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/20585d2c/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 11 20:11:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Nov 2009 22:11:59 -0600 Subject: [Freeswitch-users] att_xfer and Loopback In-Reply-To: <4AFB3A3D.1050602@gmx.net> References: <4AFB3A3D.1050602@gmx.net> Message-ID: <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> set/export the channel variable loopback_bowout=true so it's on the loopback leg On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX wrote: > Hello, > > I have some problems with attended transfer and loopback > > Scenario how id work > - A calls B > - B enters *4 gets an announcement and enter digits for C (A get MOH) > - C is called > - As soon as C picks up the call, A and C are connected and B is dropped > > How it work until here: > - A calls B > - B enters *4 gets an announcement and enter digits for C (A get MOH) > - C is called > - As soon as C picks up the call, B and C are connected (A still MOH) > > The dial string for C is dynamic and dependent on certain parameters, > therefore C must be called via Loopback in our scenario. > > > Here are the configs: > In dialplan for calling B: > > > Dialplan for executing the att_xfer: > > > > > > > > > > So this is pretty standard, except the loopback. SVN is 15322. > > Anybody has a solution for this? > > > Best regards > Peter > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/c1ad4625/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 11 20:12:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Nov 2009 22:12:53 -0600 Subject: [Freeswitch-users] att_xfer and Loopback In-Reply-To: <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> References: <4AFB3A3D.1050602@gmx.net> <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> Message-ID: <191c3a030911112012i63000f3j9867308057c5f318@mail.gmail.com> hit send too soon you want to set loopback_bowout=false This keeps loopback from trying to destroy itself when it sees a chance to cut out of the call path. On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > > set/export the channel variable loopback_bowout=true so it's on the > loopback leg > > > > > On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX wrote: > >> Hello, >> >> I have some problems with attended transfer and loopback >> >> Scenario how id work >> - A calls B >> - B enters *4 gets an announcement and enter digits for C (A get MOH) >> - C is called >> - As soon as C picks up the call, A and C are connected and B is dropped >> >> How it work until here: >> - A calls B >> - B enters *4 gets an announcement and enter digits for C (A get MOH) >> - C is called >> - As soon as C picks up the call, B and C are connected (A still MOH) >> >> The dial string for C is dynamic and dependent on certain parameters, >> therefore C must be called via Loopback in our scenario. >> >> >> Here are the configs: >> In dialplan for calling B: >> >> >> Dialplan for executing the att_xfer: >> >> >> >> >> >> >> >> >> >> So this is pretty standard, except the loopback. SVN is 15322. >> >> Anybody has a solution for this? >> >> >> Best regards >> Peter >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091111/b913d75a/attachment-0002.html From lei.tlfly at gmail.com Wed Nov 11 22:09:57 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Thu, 12 Nov 2009 14:09:57 +0800 Subject: [Freeswitch-users] How to test FS rtp packet lost rate? In-Reply-To: <2d9149cd0911110849u74c2d8d1jb90de8c20cacde9a@mail.gmail.com> References: <50c41b4e0911110543m7b5431ecu173d8386073fdb32@mail.gmail.com> <2d9149cd0911110849u74c2d8d1jb90de8c20cacde9a@mail.gmail.com> Message-ID: <50c41b4e0911112209n24011465v5fb7e8300462a58d@mail.gmail.com> Hi, thanks Kristian for your answer, it make sense 2009/11/12 Kristian Kielhofner > The simplest way I know of is to bring up another call from a local > phone and listen to the audio. At the same time run tcpdump/etc with > a strict filter to capture the rtp to/from that phone. You can then > run RTP stream analysis and the like in Wireshark to identify any lost > packets. While this obviously won't identify any/all potential lost > packets it will be a lot more practical than any of the alternatives: > > - Capturing all media streams for RTP analysis > - Implementing RTCP to identify lost packets > - Commercial hardware/software > > If FreeSWITCH, your machine, or your network are pushed to the max and > falling apart you're most likely going to see audio problems on your > single (captured) call. > > On Wed, Nov 11, 2009 at 8:43 AM, Lei Tang wrote: > > Hi all, I'm testing a FS server using sipp, I found that sipp only show > the > > retrans of sip packet, Does someone known is there a tool to test FS rtp > > packet lost rate in high concurrent call env? > > > > -- > > Lei.Tang > > lei.tlfly at gmail.com > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/6403a1e2/attachment-0002.html From Prometheus001 at gmx.net Thu Nov 12 00:38:36 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 12 Nov 2009 09:38:36 +0100 Subject: [Freeswitch-users] att_xfer and Loopback In-Reply-To: <191c3a030911112012i63000f3j9867308057c5f318@mail.gmail.com> References: <4AFB3A3D.1050602@gmx.net> <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> <191c3a030911112012i63000f3j9867308057c5f318@mail.gmail.com> Message-ID: <4AFBC98C.4070602@gmx.net> Thanks Anthony, however this rather deteriorated the situation. Now it works the following - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C both have no voice (and B is dropped) - When A hangs up, C hangs up Before it did: - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped - When A hangs up, C hangs up Best regards Peter Anthony Minessale schrieb: > hit send too soon > you want to set loopback_bowout=false > > This keeps loopback from trying to destroy itself when it sees a > chance to cut out of the call path. > > > On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale > > wrote: > > > set/export the channel variable loopback_bowout=true so it's on > the loopback leg > > > > > On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX > > wrote: > > Hello, > > I have some problems with attended transfer and loopback > > Scenario how id work > - A calls B > - B enters *4 gets an announcement and enter digits for C (A > get MOH) > - C is called > - As soon as C picks up the call, A and C are connected and B > is dropped > > How it work until here: > - A calls B > - B enters *4 gets an announcement and enter digits for C (A > get MOH) > - C is called > - As soon as C picks up the call, B and C are connected (A > still MOH) > > The dial string for C is dynamic and dependent on certain > parameters, > therefore C must be called via Loopback in our scenario. > > > Here are the configs: > In dialplan for calling B: > > > Dialplan for executing the att_xfer: > > expression="^attended_xfer$"> > > > > data="loopback/${attxfer_callthis}"/> > > > > So this is pretty standard, except the loopback. SVN is 15322. > > Anybody has a solution for this? > > > Best regards > Peter > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dschwartz at xconnect.net Thu Nov 12 02:03:09 2009 From: dschwartz at xconnect.net (David Schwartz) Date: Thu, 12 Nov 2009 12:03:09 +0200 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? Message-ID: <6EA53FAD386F9D46B97D49BFE148D51406359AED@ISR-JLM-MAIL1.xconnect.co.il> All of the example I see allow me to call FROM gtalk. Help? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/89985963/attachment-0002.html From mcampbellsmith at gmail.com Thu Nov 12 04:08:39 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 12 Nov 2009 23:08:39 +1100 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? In-Reply-To: <6EA53FAD386F9D46B97D49BFE148D51406359AED@ISR-JLM-MAIL1.xconnect.co.il> References: <6EA53FAD386F9D46B97D49BFE148D51406359AED@ISR-JLM-MAIL1.xconnect.co.il> Message-ID: <33c87fa30911120408v2d081e79ja50d2799a594ce91@mail.gmail.com> Check this page out... maybe the info should be put on the wiki... http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ On Thu, Nov 12, 2009 at 9:03 PM, David Schwartz wrote: > All of the example I see allow me to call FROM gtalk. > > > > Help? > > > > Thanks, > > > > David > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From piotr_zurek at biprotech.com Thu Nov 12 04:32:26 2009 From: piotr_zurek at biprotech.com (=?UTF-8?B?UGlvdHIgxbt1cmVr?=) Date: Thu, 12 Nov 2009 13:32:26 +0100 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> Message-ID: <4AFC005A.4090200@biprotech.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/3fd76688/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: piotr_zurek.vcf Type: text/x-vcard Size: 414 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/3fd76688/attachment-0002.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3678 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/3fd76688/attachment-0002.bin From codecomplete at free.fr Thu Nov 12 04:50:25 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 12 Nov 2009 04:50:25 -0800 (PST) Subject: [Freeswitch-users] Displaying caller ID on LED? In-Reply-To: References: <26280730.post@talk.nabble.com> <26280912.post@talk.nabble.com> Message-ID: <26318100.post@talk.nabble.com> Mitch Capper wrote: > I did something like this recently. Thanks for the feedback. I'll see how Linux can be made to send stuff to a USB display. -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26318100.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Nov 12 04:51:45 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 12 Nov 2009 04:51:45 -0800 (PST) Subject: [Freeswitch-users] cd-sounds vs. sounds? In-Reply-To: <87f2f3b90911100932i19c7c971y5fae90f6bb9f4dc0@mail.gmail.com> References: <26269842.post@talk.nabble.com> <87f2f3b90911090934p10d5fa9eh580cae19aab62eef@mail.gmail.com> <26284109.post@talk.nabble.com> <87f2f3b90911100932i19c7c971y5fae90f6bb9f4dc0@mail.gmail.com> Message-ID: <26318115.post@talk.nabble.com> mercutioviz wrote: > I believe that French and Spanish sounds are in the works by the > community. > The only other sounds I'm aware of are the Russian ones. Thanks for the tip. -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26318115.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Nov 12 04:59:18 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 12 Nov 2009 04:59:18 -0800 (PST) Subject: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) In-Reply-To: <9E5323D6B69B489384D2E89358CC5EC5@bp1.ad.bp.com> References: <9E5323D6B69B489384D2E89358CC5EC5@bp1.ad.bp.com> Message-ID: <26318213.post@talk.nabble.com> Dave Stevenson-4 wrote: > Has anyone had similar problems with the SPA3102 or has any ideas where I > can look to get to the bottom of the problem. (I have just upgraded the > SPA3102 to the latest 5.1.0 firmware) Before investigating further, you might want to ask in those forums to check that it's not an 3102-related issue instead of Freeswitch: http://forum.voxilla.com/linksys-sipura-voip-support-forum/ http://forums.whirlpool.net.au/forum/107?&g=100 -- View this message in context: http://old.nabble.com/SPA3102-Won%27t-drop-the-PSTN-line-%28UK%29-tp26286696p26318213.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dschwartz at xconnect.net Thu Nov 12 05:02:33 2009 From: dschwartz at xconnect.net (David Schwartz) Date: Thu, 12 Nov 2009 15:02:33 +0200 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? In-Reply-To: <33c87fa30911120408v2d081e79ja50d2799a594ce91@mail.gmail.com> Message-ID: <6EA53FAD386F9D46B97D49BFE148D51406359B4D@ISR-JLM-MAIL1.xconnect.co.il> Thanks Mark I read this and didn't find a dialplan (do I need one?) to make calls into gtalk? I mean how would I even dial in? via URI (e.g. someone at gmail.com)? Wouldn't that just send the call to gmail? What I am looking for is hard coding a number (e.g. 1010) that would enable me to call it and have it convert 1010 to someone at gmail.com who is NOT logged into FS and have the call goto gtalk via some other user (e.g. me at gmail.com) who IS logged into FS. Do you think this is possible? Thanks, David -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Campbell-Smith Sent: Thursday, November 12, 2009 2:09 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? Check this page out... maybe the info should be put on the wiki... http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ On Thu, Nov 12, 2009 at 9:03 PM, David Schwartz wrote: > All of the example I see allow me to call FROM gtalk. > > > > Help? > > > > Thanks, > > > > David > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From stevendt at primrosebank.net Thu Nov 12 05:32:31 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 12 Nov 2009 13:32:31 -0000 Subject: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) References: <9E5323D6B69B489384D2E89358CC5EC5@bp1.ad.bp.com> <26318213.post@talk.nabble.com> Message-ID: <135A6D0AA4AC476A841AF47564917926@bp1.ad.bp.com> Thanks for the pointers - I'll head off there now...... regards Dave ----- Original Message ----- From: "Fred-145" To: Sent: Thursday, November 12, 2009 12:59 PM Subject: Re: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) > > > Dave Stevenson-4 wrote: >> Has anyone had similar problems with the SPA3102 or has any ideas where I >> can look to get to the bottom of the problem. (I have just upgraded the >> SPA3102 to the latest 5.1.0 firmware) > > Before investigating further, you might want to ask in those forums to > check > that it's not an 3102-related issue instead of Freeswitch: > > http://forum.voxilla.com/linksys-sipura-voip-support-forum/ > http://forums.whirlpool.net.au/forum/107?&g=100 > > > -- > View this message in context: > http://old.nabble.com/SPA3102-Won%27t-drop-the-PSTN-line-%28UK%29-tp26286696p26318213.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From testeador01 at gmail.com Thu Nov 12 05:35:52 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 12 Nov 2009 08:35:52 -0500 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? In-Reply-To: <6EA53FAD386F9D46B97D49BFE148D51406359B4D@ISR-JLM-MAIL1.xconnect.co.il> References: <33c87fa30911120408v2d081e79ja50d2799a594ce91@mail.gmail.com> <6EA53FAD386F9D46B97D49BFE148D51406359B4D@ISR-JLM-MAIL1.xconnect.co.il> Message-ID: Hello, Obviously it is possible, next time try to search better, the answer is on the same blog Mark pointed you too: http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-jingle-all-the-way/ Good luck 2009/11/12 David Schwartz > Thanks Mark > > I read this and didn't find a dialplan (do I need one?) to make calls into > gtalk? I mean how would I even dial in? via URI (e.g. someone at gmail.com)? > Wouldn't that just send the call to gmail? > > What I am looking for is hard coding a number (e.g. 1010) that would enable > me to call it and have it convert 1010 to someone at gmail.com who is NOT > logged into FS and have the call goto gtalk via some other user (e.g. > me at gmail.com) who IS logged into FS. > > Do you think this is possible? > > Thanks, > > David > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark > Campbell-Smith > Sent: Thursday, November 12, 2009 2:09 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? > > Check this page out... maybe the info should be put on the wiki... > > http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ > > > On Thu, Nov 12, 2009 at 9:03 PM, David Schwartz > wrote: > > All of the example I see allow me to call FROM gtalk. > > > > > > > > Help? > > > > > > > > Thanks, > > > > > > > > David > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/cbd3bd39/attachment-0002.html From dschwartz at xconnect.net Thu Nov 12 05:53:24 2009 From: dschwartz at xconnect.net (David Schwartz) Date: Thu, 12 Nov 2009 15:53:24 +0200 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? In-Reply-To: Message-ID: <6EA53FAD386F9D46B97D49BFE148D51406359B65@ISR-JLM-MAIL1.xconnect.co.il> Thanks I overlooked that :) D. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena Sent: Thursday, November 12, 2009 3:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? Hello, Obviously it is possible, next time try to search better, the answer is on the same blog Mark pointed you too: http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-jingle-all-the-way/ Good luck 2009/11/12 David Schwartz > Thanks Mark I read this and didn't find a dialplan (do I need one?) to make calls into gtalk? I mean how would I even dial in? via URI (e.g. someone at gmail.com)? Wouldn't that just send the call to gmail? What I am looking for is hard coding a number (e.g. 1010) that would enable me to call it and have it convert 1010 to someone at gmail.com who is NOT logged into FS and have the call goto gtalk via some other user (e.g. me at gmail.com) who IS logged into FS. Do you think this is possible? Thanks, David -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Campbell-Smith Sent: Thursday, November 12, 2009 2:09 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? Check this page out... maybe the info should be put on the wiki... http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ On Thu, Nov 12, 2009 at 9:03 PM, David Schwartz > wrote: > All of the example I see allow me to call FROM gtalk. > > > > Help? > > > > Thanks, > > > > David > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/cc901004/attachment-0002.html From brian at freeswitch.org Thu Nov 12 05:59:29 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Nov 2009 07:59:29 -0600 Subject: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk? In-Reply-To: <6EA53FAD386F9D46B97D49BFE148D51406359B4D@ISR-JLM-MAIL1.xconnect.co.il> References: <6EA53FAD386F9D46B97D49BFE148D51406359B4D@ISR-JLM-MAIL1.xconnect.co.il> Message-ID: This is just basic freeswitch dialplan concepts. It has nothing to do specifically with gtalk. Seems like you need to step back and do some more reading on the dialplan. ;) /b On Nov 12, 2009, at 7:02 AM, David Schwartz wrote: > What I am looking for is hard coding a number (e.g. 1010) that would > enable me to call it and have it convert 1010 to someone at gmail.com > who is NOT logged into FS and have the call goto gtalk via some > other user (e.g. me at gmail.com) who IS logged into FS. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/b4fbe386/attachment-0002.html From peder at networkoblivion.com Thu Nov 12 06:25:48 2009 From: peder at networkoblivion.com (Peder) Date: Thu, 12 Nov 2009 08:25:48 -0600 Subject: [Freeswitch-users] Cisco 79x1 & Presence Message-ID: <034a01ca63a4$07c13790$1743a6b0$@com> Has anybody every figured out how to get presence working on a Cisco 79x1 w/ FreeSWITCH? I spent quite a bit of time 6+ months ago on it and could never get it to work. Peder From lei.tlfly at gmail.com Thu Nov 12 07:01:02 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Thu, 12 Nov 2009 23:01:02 +0800 Subject: [Freeswitch-users] hangup incoming call by Reason: Q.850; cause=1; text="Unallocated (unassigned) number" Message-ID: <50c41b4e0911120701r737ce492j5bf5f5be2fd15550@mail.gmail.com> Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal sip endpoint of FS. I added two dialplan in public dialplan xml file. as flow: Every thing is ok when call to number 88888. but when I call the second number "*114", fs hangup after accept and answer the call, I captured the sip packets and found FS sent a bye packet after answer the call. the cause is "Reason: Q.850;cause=1;text="Unallocated (unassigned) number"". But as the fs console log show, the call is answered and the correct ivr script is runned. Why FS hangup the call? Does somebody have any idea about this problem? ============sip packets=================== ********invite msg from softswitch INVITE sip:*114 at 10.37.143.6:5060;user=phone SIP/2.0 Contact: Content-Type: application/sdp To: From: xxxxxxxxx;tag=949132463135364198E42500 P-Asserted-Identity: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE Supported: 100rel,timer,replaces,diversion Expires: 155 Session-Expires: 1800 Min-SE: 90 Call-ID: 01FD10D1BD81400000010690 at sip-3 Max-Forwards: 70 CSeq: 1 INVITE Timestamp: 58520 Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK5C0F524645A70C943998751419749696 Content-Length: 150 v=0 o=- 54000602557 1258015146 IN IP4 10.4.35.59 s=SDP Data c=IN IP4 10.4.35.59 t=0 0 m=audio 30000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 ******FS ack SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK5C0F524645A70C943998751419749696 From: xxxxxxxxx ;tag=949132463135364198E42500 To: Call-ID: 01FD10D1BD81400000010690 at sip-3 CSeq: 1 INVITE Timestamp: 58520 0.000000 User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Content-Length: 0 *****FS answer the call (in lua script, I called session:answer() ) SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK5C0F524645A70C943998751419749696 From: xxxxxxxxx ;tag=949132463135364198E42500 To: ;tag=UjZcZUKZXjHcQ Call-ID: 01FD10D1BD81400000010690 at sip-3 CSeq: 1 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, refer Session-Expires: 1800;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 245 v=0 o=FreeSWITCH 1257988835 1257988836 IN IP4 10.37.143.6 s=FreeSWITCH c=IN IP4 10.37.143.6 t=0 0 m=audio 24890 RTP/AVP 8 120 a=rtpmap:8 PCMA/8000 a=rtpmap:120 telephone-event/8000 a=fmtp:120 0-16 a=silenceSupp:off - - - - a=ptime:20 ACK sip:*114 at 10.37.143.6:5060;transport=udp SIP/2.0 CSeq: 1 ACK To: ;tag=UjZcZUKZXjHcQ From: xxxxxxxxx;tag=949132463135364198E42500 Call-ID: 01FD10D1BD81400000010690 at sip-3 Max-Forwards: 70 Timestamp: 58520 Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bK0CC4AE6EE59CA15F69429CDB97848C21 Content-Length: 0 *******FS hangup the call BYE sip:*114 at 10.37.143.6:5060;transport=udp SIP/2.0 Reason: Q.850;cause=1;text="Unallocated (unassigned) number" To: ;tag=UjZcZUKZXjHcQ From: xxxxxxxxx;tag=949132463135364198E42500 Call-ID: 01FD10D1BD81400000010690 at sip-3 Max-Forwards: 70 CSeq: 2 BYE Timestamp: 58521 Via: SIP/2.0/UDP 10.4.35.17:5061 ;branch=z9hG4bKBE2D7D86B44CA171A5D374ECAA99A1DB Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/27853ad9/attachment-0002.html From codecomplete at free.fr Thu Nov 12 07:38:30 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 12 Nov 2009 07:38:30 -0800 (PST) Subject: [Freeswitch-users] Does OpenZap support CTR21? In-Reply-To: <20091106200458.AACDC3E5BEF@mail.cune.org> References: <26217371.post@talk.nabble.com> <7FD19B47-C121-48CD-98C2-2830BFDF1068@jerris.com> <26230864.post@talk.nabble.com> <20091106200458.AACDC3E5BEF@mail.cune.org> Message-ID: <26320837.post@talk.nabble.com> Russell.Mosemann wrote: > Yes, it should just work. I'd recommend Dahdi (complete), because Zaptel > is not being developed anymore. Thanks for the links. Turns out this card seems incompatible with the motherboard I have, so I'll concentrate on the Linksys 3102 instead. -- View this message in context: http://old.nabble.com/Does-OpenZap-support-CTR21--tp26217371p26320837.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Thu Nov 12 07:50:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Nov 2009 09:50:49 -0600 Subject: [Freeswitch-users] att_xfer and Loopback In-Reply-To: <4AFBC98C.4070602@gmx.net> References: <4AFB3A3D.1050602@gmx.net> <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> <191c3a030911112012i63000f3j9867308057c5f318@mail.gmail.com> <4AFBC98C.4070602@gmx.net> Message-ID: <191c3a030911120750p34d27d44u2fec4015caf2f367@mail.gmail.com> if you provide a console trace of both situations with console loglevel debug and put them on pastebin i can tell you what's happening. On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX wrote: > Thanks Anthony, > > however this rather deteriorated the situation. > Now it works the following > - A calls B > - B enters *4 gets an announcement and enters digits for C (A get MOH) > - C is called > - As soon as C picks up the call, A and C both have no voice (and B is > dropped) > - When A hangs up, C hangs up > > Before it did: > - A calls B > - B enters *4 gets an announcement and enters digits for C (A get MOH) > - C is called > - As soon as C picks up the call, A and C are connected and B is dropped > - When A hangs up, C hangs up > > Best regards > Peter > > Anthony Minessale schrieb: > > hit send too soon > > you want to set loopback_bowout=false > > > > This keeps loopback from trying to destroy itself when it sees a > > chance to cut out of the call path. > > > > > > On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale > > > > wrote: > > > > > > set/export the channel variable loopback_bowout=true so it's on > > the loopback leg > > > > > > > > > > On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX > > > wrote: > > > > Hello, > > > > I have some problems with attended transfer and loopback > > > > Scenario how id work > > - A calls B > > - B enters *4 gets an announcement and enter digits for C (A > > get MOH) > > - C is called > > - As soon as C picks up the call, A and C are connected and B > > is dropped > > > > How it work until here: > > - A calls B > > - B enters *4 gets an announcement and enter digits for C (A > > get MOH) > > - C is called > > - As soon as C picks up the call, B and C are connected (A > > still MOH) > > > > The dial string for C is dynamic and dependent on certain > > parameters, > > therefore C must be called via Loopback in our scenario. > > > > > > Here are the configs: > > In dialplan for calling B: > > > > > > Dialplan for executing the att_xfer: > > > > > expression="^attended_xfer$"> > > > > > > > > > data="loopback/${attxfer_callthis}"/> > > > > > > > > So this is pretty standard, except the loopback. SVN is 15322. > > > > Anybody has a solution for this? > > > > > > Best regards > > Peter > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/edb5436c/attachment-0002.html From mike at jerris.com Thu Nov 12 08:09:36 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 12 Nov 2009 11:09:36 -0500 Subject: [Freeswitch-users] hangup incoming call by Reason: Q.850; cause=1; text="Unallocated (unassigned) number" In-Reply-To: <50c41b4e0911120701r737ce492j5bf5f5be2fd15550@mail.gmail.com> References: <50c41b4e0911120701r737ce492j5bf5f5be2fd15550@mail.gmail.com> Message-ID: Take a look at the freeswitch debug log, it should tell you exactly why it hung up. Mike On Nov 12, 2009, at 10:01 AM, Lei Tang wrote: > Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal sip endpoint of FS. > I added two dialplan in public dialplan xml file. as flow: > > > > > > > > > > > > > Every thing is ok when call to number 88888. but when I call the second number "*114", fs hangup after accept and answer the call, I captured the sip packets and found FS sent a bye packet after answer the call. the cause is "Reason: Q.850;cause=1;text="Unallocated (unassigned) number"". But as the fs console log show, the call is answered and the correct ivr script is runned. Why FS hangup the call? Does somebody have any idea about this problem? > > > ============sip packets=================== > ********invite msg from softswitch > INVITE sip:*114 at 10.37.143.6:5060;user=phone SIP/2.0 > Contact: > Content-Type: application/sdp > To: > From: xxxxxxxxx;tag=949132463135364198E42500 > P-Asserted-Identity: > Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE > Supported: 100rel,timer,replaces,diversion > Expires: 155 > Session-Expires: 1800 > Min-SE: 90 > Call-ID: 01FD10D1BD81400000010690 at sip-3 > Max-Forwards: 70 > CSeq: 1 INVITE > Timestamp: 58520 > Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 > Content-Length: 150 > > v=0 > o=- 54000602557 1258015146 IN IP4 10.4.35.59 > s=SDP Data > c=IN IP4 10.4.35.59 > t=0 0 > m=audio 30000 RTP/AVP 8 > a=rtpmap:8 PCMA/8000 > a=ptime:20 > > > ******FS ack > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 > From: xxxxxxxxx ;tag=949132463135364198E42500 > To: > Call-ID: 01FD10D1BD81400000010690 at sip-3 > CSeq: 1 INVITE > Timestamp: 58520 0.000000 > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 > Content-Length: 0 > > *****FS answer the call (in lua script, I called session:answer() ) > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 > From: xxxxxxxxx ;tag=949132463135364198E42500 > To: ;tag=UjZcZUKZXjHcQ > Call-ID: 01FD10D1BD81400000010690 at sip-3 > CSeq: 1 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Require: timer > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Session-Expires: 1800;refresher=uac > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 245 > > v=0 > o=FreeSWITCH 1257988835 1257988836 IN IP4 10.37.143.6 > s=FreeSWITCH > c=IN IP4 10.37.143.6 > t=0 0 > m=audio 24890 RTP/AVP 8 120 > a=rtpmap:8 PCMA/8000 > a=rtpmap:120 telephone-event/8000 > a=fmtp:120 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:*114 at 10.37.143.6:5060;transport=udp SIP/2.0 > CSeq: 1 ACK > To: ;tag=UjZcZUKZXjHcQ > From: xxxxxxxxx;tag=949132463135364198E42500 > Call-ID: 01FD10D1BD81400000010690 at sip-3 > Max-Forwards: 70 > Timestamp: 58520 > Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK0CC4AE6EE59CA15F69429CDB97848C21 > Content-Length: 0 > > *******FS hangup the call > BYE sip:*114 at 10.37.143.6:5060;transport=udp SIP/2.0 > Reason: Q.850;cause=1;text="Unallocated (unassigned) number" > To: ;tag=UjZcZUKZXjHcQ > From: xxxxxxxxx;tag=949132463135364198E42500 > Call-ID: 01FD10D1BD81400000010690 at sip-3 > Max-Forwards: 70 > CSeq: 2 BYE > Timestamp: 58521 > Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bKBE2D7D86B44CA171A5D374ECAA99A1DB > Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/8029d908/attachment-0002.html From lists at redbonez.net Thu Nov 12 09:41:37 2009 From: lists at redbonez.net (Adam Ford) Date: Thu, 12 Nov 2009 10:41:37 -0700 Subject: [Freeswitch-users] Polycom SoundPoint IP501 In-Reply-To: <034a01ca63a4$07c13790$1743a6b0$@com> References: <034a01ca63a4$07c13790$1743a6b0$@com> Message-ID: <012901ca63bf$64152090$2c3f61b0$@net> Has anyone used a Polycom SoundPoint IP501 or similar hard phone with FreeSWITCH? I configured one to register with my FreeSWITCH server using one of the default sip profiles to test and I get "[DEBUG] sofia_reg.c:1688 SIP username 1001 does not match auth username" in the log file and the phone doesn't register. I have confirmed that the auth username and the display name are both 1001. Is there some additional configuration on the FreeSWITCH side to get these phones to register? Thanks for any help you can offer, -Adam From brian at freeswitch.org Thu Nov 12 09:50:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Nov 2009 11:50:28 -0600 Subject: [Freeswitch-users] Polycom SoundPoint IP501 In-Reply-To: <012901ca63bf$64152090$2c3f61b0$@net> References: <034a01ca63a4$07c13790$1743a6b0$@com> <012901ca63bf$64152090$2c3f61b0$@net> Message-ID: <50AD10F6-C050-40F7-A235-93E40213273D@freeswitch.org> Not sure what do you have in your config file for the polycom exactly? btw you hijacked the Cisco Presence thread by clicking reply.. and changing the subject please don't do that in the future. Click new message and input the address for the list. Thanks, Brian On Nov 12, 2009, at 11:41 AM, Adam Ford wrote: > Has anyone used a Polycom SoundPoint IP501 or similar hard phone with > FreeSWITCH? I configured one to register with my FreeSWITCH server > using one > of the default sip profiles to test and I get "[DEBUG] sofia_reg.c: > 1688 SIP > username 1001 does not match auth username" in the log file and the > phone > doesn't register. I have confirmed that the auth username and the > display > name are both 1001. Is there some additional configuration on the > FreeSWITCH > side to get these phones to register? > > Thanks for any help you can offer, > -Adam > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Thu Nov 12 09:50:49 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Nov 2009 11:50:49 -0600 Subject: [Freeswitch-users] Cisco 79x1 & Presence In-Reply-To: <034a01ca63a4$07c13790$1743a6b0$@com> References: <034a01ca63a4$07c13790$1743a6b0$@com> Message-ID: <859F9AF7-01A9-4621-B902-393E0B93DCDC@freeswitch.org> They do it in their own weird way... if you wanna track it down I know their are examples of it out there. /b On Nov 12, 2009, at 8:25 AM, Peder wrote: > Has anybody every figured out how to get presence working on a Cisco > 79x1 w/ > FreeSWITCH? I spent quite a bit of time 6+ months ago on it and > could never > get it to work. > > Peder > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From tina at a2unlimited.com Thu Nov 12 09:57:34 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Thu, 12 Nov 2009 12:57:34 -0500 Subject: [Freeswitch-users] Calls per second on FreeSWITCH Message-ID: <67387ff53247696360986ff66f5dc894.squirrel@emailmg.ipower.com> I'm trying to increase the number of calls per second that I can originate from FreeSWITCH, but I cannot seem to get more than two-per-second. (I am trying to use FS to initiate thousands of calls quickly) switch.conf.xml I beefed up the max-sessions and sessions-per-second in the switch.conf.xml file, but that did not seem to make any difference. Any thoughts? - Tina From mattdfong at gmail.com Thu Nov 12 10:17:12 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 13 Nov 2009 01:17:12 +0700 Subject: [Freeswitch-users] Calls per second on FreeSWITCH In-Reply-To: <67387ff53247696360986ff66f5dc894.squirrel@emailmg.ipower.com> References: <67387ff53247696360986ff66f5dc894.squirrel@emailmg.ipower.com> Message-ID: <4256bf830911121017u7e755453o987cd55359b21928@mail.gmail.com> Tina, How are you originating the calls? from the console? Try bgapi originate... --matt Voice Broadcasting - http://www.hellohunter.com/voice_blast.php On Fri, Nov 13, 2009 at 12:57 AM, wrote: > I'm trying to increase the number of calls per second that I can originate > from FreeSWITCH, but I cannot seem to get more than two-per-second. > > (I am trying to use FS to initiate thousands of calls quickly) > > switch.conf.xml > I beefed up the max-sessions and sessions-per-second in the > switch.conf.xml file, but that did not seem to make any difference. > > Any thoughts? > > - Tina > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/5828eb8e/attachment-0002.html From lfurrea at gmail.com Thu Nov 12 10:38:54 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Thu, 12 Nov 2009 12:38:54 -0600 Subject: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) In-Reply-To: <135A6D0AA4AC476A841AF47564917926@bp1.ad.bp.com> References: <9E5323D6B69B489384D2E89358CC5EC5@bp1.ad.bp.com> <26318213.post@talk.nabble.com> <135A6D0AA4AC476A841AF47564917926@bp1.ad.bp.com> Message-ID: Remember you have a plain old regular analog connection between the FXO port of the SPA and your "phone line". The FXO circuit is just an analog switch (open or closed) if no one answers on the IP side and the person on the PSTN side hangs up, then the FXO side should sense a change in the polarity of voltage, a lack of current for a certain period or at least detect a tone played by your Telco to be able to go on hook and wait for another call. Try to google "disconnect supervision issues" so that you can get a clearer explanation of what you are experiencing. On Thu, Nov 12, 2009 at 7:32 AM, Dave Stevenson wrote: > Thanks for the pointers - I'll head off there now...... > > regards > Dave > > > > ----- Original Message ----- > From: "Fred-145" > To: > Sent: Thursday, November 12, 2009 12:59 PM > Subject: Re: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) > > > > > > > > Dave Stevenson-4 wrote: > >> Has anyone had similar problems with the SPA3102 or has any ideas where > I > >> can look to get to the bottom of the problem. (I have just upgraded the > >> SPA3102 to the latest 5.1.0 firmware) > > > > Before investigating further, you might want to ask in those forums to > > check > > that it's not an 3102-related issue instead of Freeswitch: > > > > http://forum.voxilla.com/linksys-sipura-voip-support-forum/ > > http://forums.whirlpool.net.au/forum/107?&g=100 > > > > > > -- > > View this message in context: > > > http://old.nabble.com/SPA3102-Won%27t-drop-the-PSTN-line-%28UK%29-tp26286696p26318213.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- firma -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/27be958e/attachment-0002.html From tina at a2unlimited.com Thu Nov 12 12:08:03 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Thu, 12 Nov 2009 15:08:03 -0500 Subject: [Freeswitch-users] Calls per second on FreeSWITCH In-Reply-To: References: Message-ID: Matt, Thank you so much! bgapi did the trick. - Tina > Tina, > > How are you originating the calls? from the console? Try bgapi > originate... > > --matt > Voice Broadcasting - http://www.hellohunter.com/voice_blast.php > > On Fri, Nov 13, 2009 at 12:57 AM, wrote: > >> I'm trying to increase the number of calls per second that I can >> originate >> from FreeSWITCH, but I cannot seem to get more than two-per-second. >> >> (I am trying to use FS to initiate thousands of calls quickly) >> >> switch.conf.xml >> I beefed up the max-sessions and sessions-per-second in the >> switch.conf.xml file, but that did not seem to make any difference. >> >> Any thoughts? >> >> - Tina >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From viper at fx-services.com Thu Nov 12 11:14:09 2009 From: viper at fx-services.com (Robin Vleij) Date: Thu, 12 Nov 2009 20:14:09 +0100 Subject: [Freeswitch-users] Large number of destinations Message-ID: <4AFC5E81.9020104@fx-services.com> Hi all, I'm currently building a proof-of-concept box using Freeswitch. Coming from Asterisk/Kamalio/OpenSER it looks very cool so far, very complete. The plan is to make some sort of SIP router, some would call it an SBC I guess. There will be no PBX stuff, just gateways that talk to each other. PSTN Gateways or other operators or systems. If a system is locally connected (say a local voip platform or interconnected partner), traffic to those destinations should be routed directly to that system and not out to PSTN. I'm looking at a potentially large nr of destination nrs or ranges. Not all those destinations are in the local ENUM so I can't use that as a routing system. I'm thinking about mod_lcr, but it seems more suited for eh ... LCR routing, which is not what I want to do here. I just want to define which nrs or nr ranges are "directly" connected, so that when someone calls there from whatever way they come in (I'm running just one instance and thought about defining all gateways/systems as gateways in the SIP profile), they should end up there and not at PSTN. I think I have two ways of doing this: 1. Make a HUGE XML dialplan and use that to fall back to when internal ENUM lookup doesn't give a result back to where a nr is located 2. Use LCR and find out some kind of way to load all of these destinations into a LCR table and use it in the "wrong" way, ie no costs are involved, it should just be a way to know which nrs or ranges are to be sent to which gateway. Nr1 is probably best (anyone experience how many conditions one can have dialplan_xml?), but say that we would exchange traffic with an operator, it would really suck writing an XML dialplan with 5000 number ranges. :) Anyone experience with this or ideas how this can be solved? Since it's a proof-of-concept it's unclear how exactly those customers or systems are looking. They might be 6000 (un)ported individual nrs, or just a few large ranges. /Robin From yehavi.bourvine at gmail.com Thu Nov 12 12:52:40 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 12 Nov 2009 22:52:40 +0200 Subject: [Freeswitch-users] Polycom SoundPoint IP501 In-Reply-To: <50AD10F6-C050-40F7-A235-93E40213273D@freeswitch.org> References: <034a01ca63a4$07c13790$1743a6b0$@com> <012901ca63bf$64152090$2c3f61b0$@net> <50AD10F6-C050-40F7-A235-93E40213273D@freeswitch.org> Message-ID: I am using Polycoms (430 and 501) with FreeSwitch. How do you provision them? Via WEB or config files? If you use config files than I can send you some sample files. Regards, __Yehavi: On Nov 12, 2009, at 11:41 AM, Adam Ford wrote: > > > Has anyone used a Polycom SoundPoint IP501 or similar hard phone with > > FreeSWITCH? I configured one to register with my FreeSWITCH server > > using one > > of the default sip profiles to test and I get "[DEBUG] sofia_reg.c: > > 1688 SIP > > username 1001 does not match auth username" in the log file and the > > phone > > doesn't register. I have confirmed that the auth username and the > > display > > name are both 1001. Is there some additional configuration on the > > FreeSWITCH > > side to get these phones to register? > > > > Thanks for any help you can offer, > > -Adam > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/014e75de/attachment-0002.html From rupa at rupa.com Thu Nov 12 12:59:38 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 12 Nov 2009 14:59:38 -0600 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: <4AFC5E81.9020104@fx-services.com> References: <4AFC5E81.9020104@fx-services.com> Message-ID: Take a look at mod_easyroute. On Thu, Nov 12, 2009 at 1:14 PM, Robin Vleij wrote: > Hi all, > > I'm currently building a proof-of-concept box using Freeswitch. Coming > from Asterisk/Kamalio/OpenSER it looks very cool so far, very complete. > > The plan is to make some sort of SIP router, some would call it an SBC I > guess. There will be no PBX stuff, just gateways that talk to each > other. PSTN Gateways or other operators or systems. > > If a system is locally connected (say a local voip platform or > interconnected partner), traffic to those destinations should be routed > directly to that system and not out to PSTN. I'm looking at a > potentially large nr of destination nrs or ranges. Not all those > destinations are in the local ENUM so I can't use that as a routing system. > > I'm thinking about mod_lcr, but it seems more suited for eh ... LCR > routing, which is not what I want to do here. I just want to define > which nrs or nr ranges are "directly" connected, so that when someone > calls there from whatever way they come in (I'm running just one > instance and thought about defining all gateways/systems as gateways in > the SIP profile), they should end up there and not at PSTN. > > I think I have two ways of doing this: > > 1. Make a HUGE XML dialplan and use that to fall back to when internal > ENUM lookup doesn't give a result back to where a nr is located > 2. Use LCR and find out some kind of way to load all of these > destinations into a LCR table and use it in the "wrong" way, ie no costs > are involved, it should just be a way to know which nrs or ranges are to > be sent to which gateway. > > Nr1 is probably best (anyone experience how many conditions one can have > dialplan_xml?), but say that we would exchange traffic with an operator, > it would really suck writing an XML dialplan with 5000 number ranges. :) > > Anyone experience with this or ideas how this can be solved? Since it's > a proof-of-concept it's unclear how exactly those customers or systems > are looking. They might be 6000 (un)ported individual nrs, or just a few > large ranges. > > /Robin > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From tina at a2unlimited.com Thu Nov 12 13:19:26 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Thu, 12 Nov 2009 16:19:26 -0500 Subject: [Freeswitch-users] CDR for Failed Calls Message-ID: <2d6ab34c4cc1359c15811ed655b8451a.squirrel@emailmg.ipower.com> I am using xml_cdr to generate CDR results from FreeSWITCH servers, and I've noticed that failed call attempts are not showing up in the results. Whereas the failed attempt is showing up in the Master.csv file. For example, I've initiated some outbound calls that show up in the Master.csv as "RECOVERY_ON_TIMER_EXPIRE", but there is not record from the xml_cdr process. Is there a parameter that can be adjusted in xml_cdr.conf.xml that enables the submission of CDR data for failed calls? ------------------ Here is my current xml_cdr.conf.xml configuration: ------------------ From siniypin at gmail.com Thu Nov 12 14:27:02 2009 From: siniypin at gmail.com (RobertT) Date: Fri, 13 Nov 2009 01:27:02 +0300 Subject: [Freeswitch-users] tcp call misses sip message Message-ID: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> Hello everyone! I'v got strange problem with incomplete call via tcp transport. When I perform bridged call from one ua (no matter what transport udp or tcp) through FS this call's leg b message sequence (over tcp) lacks finishing SIP message what in it's turn cause the call to be disconnected by the client by timeout. Everything works fine with local calls, so I guess the problem is somewhere between UA and FS. There is no NAT and calls via udp are being established correctly. The problem is with tcp and tls as well. This is the sender's ua SIP trace: TX 1049 bytes Request msg INVITE/cseq=11615 (tdta0486C000) to UDP : RX 348 bytes Response msg 100/INVITE/cseq=11615 (rdata0482806C) from UDP : RX 813 bytes Response msg 407/INVITE/cseq=11615 (rdata0482806C) from UDP : TX 346 bytes Request msg ACK/cseq=11615 (tdta0486EFD0) to UDP : TX 1324 bytes Request msg INVITE/cseq=11616 (tdta0486C000) to UDP : RX 348 bytes Response msg 100/INVITE/cseq=11616 (rdata0482806C) from UDP : RX 1083 bytes Response msg 200/INVITE/cseq=11616 (rdata0482806C) from UDP : TX 360 bytes Request msg ACK/cseq=11616 (tdta04874E38) to UDP : And this is the reciever's SIP trace: RX 1167 bytes Request msg INVITE/cseq=122911315 (rdata04864E10) from tcp : TX 298 bytes Response msg 100/INVITE/cseq=122911315 (tdta0486D010) to tcp : TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : ------ I guess this is where ACK is supposed to arrive Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=1 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=2 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=3 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : .... Sofia profile config: and super-smart dialplan FS 1.0.5pre5 is running on Windows Server 2007SP1 64bit.This issue first occured with 1.0.4 release. Best regards, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/4a957142/attachment-0002.html From srinivas.ksvreddy at gmail.com Thu Nov 12 14:32:00 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Fri, 13 Nov 2009 04:02:00 +0530 Subject: [Freeswitch-users] mod event socket In-Reply-To: References: Message-ID: HI all, i have connected Freeswtich(mod event socket) through telnet(tcp) 8021 port, when i am trying to connect freeswtich it it taking 20 seconds to get response from FS, can i able to reduce tcp response time? thanks Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/a4d868c3/attachment-0002.html From viper at fx-services.com Thu Nov 12 14:32:33 2009 From: viper at fx-services.com (Robin Vleij) Date: Thu, 12 Nov 2009 23:32:33 +0100 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: References: <4AFC5E81.9020104@fx-services.com> Message-ID: <4AFC8D01.9060401@fx-services.com> On 11/12/09 9:59 PM, Rupa Schomaker wrote: Hi! > Take a look at mod_easyroute. Cool, I remember "quick-reading" about that module and thinking "nah, not needed". Then when the plan changed and I needed the large amount of routes it didn't struck me that easyroute is what I need for what I want to do. Perfect. If I read it right, this is suited for "complete" nrs. So would I have a system connected with lots of DIDs, I would put them in easyroute. Then for systems with lots of number ranges, I would use mod_lcr. My dialplan context where I would handle inbound from anywhere would look like: 1. ENUM lookup to see if it's a ported nr to any directly connected system 2. mod_lcr lookup to see if it's any large nr range directly connected 3. mod_easyroute to see if it's any individual nr directly connected (via some gateway) 4. Give up :) Guess I'll get working on stuff in this order then. Thanks for the tip! /Robin From lists at redbonez.net Thu Nov 12 14:36:15 2009 From: lists at redbonez.net (Adam Ford) Date: Thu, 12 Nov 2009 15:36:15 -0700 Subject: [Freeswitch-users] Polycom SoundPoint IP501 In-Reply-To: References: <034a01ca63a4$07c13790$1743a6b0$@com> <012901ca63bf$64152090$2c3f61b0$@net> <50AD10F6-C050-40F7-A235-93E40213273D@freeswitch.org> Message-ID: <015101ca63e8$8cb47680$a61d6380$@net> I was trying to configure it just on the phone itself, but apparently even though it says Auth. User on the phone setting, it doesn't actually set the auth username according to the web interface. After using the web interface to configure the phone it works now. Thank you for your responses. -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi Bourvine Sent: Thursday, November 12, 2009 1:53 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom SoundPoint IP501 I am using Polycoms (430 and 501) with FreeSwitch. How do you provision them? Via WEB or config files? If you use config files than I can send you some sample files. Regards, __Yehavi: On Nov 12, 2009, at 11:41 AM, Adam Ford wrote: > Has anyone used a Polycom SoundPoint IP501 or similar hard phone with > FreeSWITCH? I configured one to register with my FreeSWITCH server > using one > of the default sip profiles to test and I get "[DEBUG] sofia_reg.c: > 1688 SIP > username 1001 does not match auth username" in the log file and the > phone > doesn't register. I have confirmed that the auth username and the > display > name are both 1001. Is there some additional configuration on the > FreeSWITCH > side to get these phones to register? > > Thanks for any help you can offer, > -Adam > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/2090f293/attachment-0002.html From brian at freeswitch.org Thu Nov 12 14:46:11 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Nov 2009 16:46:11 -0600 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> Message-ID: <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> tack on a ;transport=tcp /b On Nov 12, 2009, at 4:27 PM, RobertT wrote: > From rupa at rupa.com Thu Nov 12 14:53:13 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 12 Nov 2009 16:53:13 -0600 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: <4AFC8D01.9060401@fx-services.com> References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> Message-ID: On Thu, Nov 12, 2009 at 4:32 PM, Robin Vleij wrote: > On 11/12/09 9:59 PM, Rupa Schomaker wrote: > If I read it right, this is suited for "complete" nrs. So would I have a > system connected with lots of DIDs, I would put them in easyroute. Then > for systems with lots of number ranges, I would use mod_lcr. lcr is based on prefix, so the boundaries for which the range is assigned may not match a prefix. You may be better off either: 1) denormalize your ranges and just insert all distinct #s 2) Modify mod_easyroute to support ranges 3) talk to SWK (he is on irc here and there) about his (non free) fancier routing options -- -Rupa From anthony.minessale at gmail.com Thu Nov 12 14:57:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Nov 2009 16:57:35 -0600 Subject: [Freeswitch-users] CDR for Failed Calls In-Reply-To: <2d6ab34c4cc1359c15811ed655b8451a.squirrel@emailmg.ipower.com> References: <2d6ab34c4cc1359c15811ed655b8451a.squirrel@emailmg.ipower.com> Message-ID: <191c3a030911121457k77e83f0alff64bc503b2708fc@mail.gmail.com> enable the b leg logging On Thu, Nov 12, 2009 at 3:19 PM, wrote: > I am using xml_cdr to generate CDR results from FreeSWITCH servers, and > I've noticed that failed call attempts are not showing up in the results. > > Whereas the failed attempt is showing up in the Master.csv file. > For example, I've initiated some outbound calls that show up in the > Master.csv as "RECOVERY_ON_TIMER_EXPIRE", but there is not record from the > xml_cdr process. > > Is there a parameter that can be adjusted in xml_cdr.conf.xml that enables > the submission of CDR data for failed calls? > > ------------------ > Here is my current xml_cdr.conf.xml configuration: > > > > > > > > > > > > > > > ------------------ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/4484890d/attachment-0002.html From msc at freeswitch.org Thu Nov 12 15:29:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Nov 2009 15:29:03 -0800 Subject: [Freeswitch-users] mod event socket In-Reply-To: References: Message-ID: <87f2f3b90911121529ubd6b61dxd9fb27036bba89ca@mail.gmail.com> What exactly are you typing when you connect? Also, which version of FS? -MC On Thu, Nov 12, 2009 at 2:32 PM, srinivasula reddy < srinivas.ksvreddy at gmail.com> wrote: > > > > HI all, > > i have connected Freeswtich(mod event socket) through telnet(tcp) 8021 > port, when i am trying to connect freeswtich it it taking 20 seconds to get > response from FS, > can i able to reduce tcp response time? > > thanks > Srinivasula Reddy K > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/c5ae1fa4/attachment-0002.html From egable+freeswitch at gmail.com Thu Nov 12 17:49:17 2009 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Thu, 12 Nov 2009 20:49:17 -0500 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> Message-ID: Or, of course, there is always mod_xml_curl. Basically, XML dialplan on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a web application server, web application server responds with XML routing response, FreeSWITCH routes the call. On Thu, Nov 12, 2009 at 5:53 PM, Rupa Schomaker wrote: > On Thu, Nov 12, 2009 at 4:32 PM, Robin Vleij wrote: >> On 11/12/09 9:59 PM, Rupa Schomaker wrote: >> If I read it right, this is suited for "complete" nrs. So would I have a >> system connected with lots of DIDs, I would put them in easyroute. Then >> for systems with lots of number ranges, I would use mod_lcr. > > lcr is based on prefix, so the boundaries for which the range is > assigned may not match a prefix. ?You may be better off either: > > > 1) denormalize your ranges and just insert all distinct #s > > 2) Modify mod_easyroute to support ranges > > 3) talk to SWK (he is on irc here and there) about his (non free) > fancier routing options > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From orien at tx.rr.com Thu Nov 12 18:44:57 2009 From: orien at tx.rr.com (Orien Love) Date: Thu, 12 Nov 2009 20:44:57 -0600 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: References: Message-ID: <4AFCC829.2070507@tx.rr.com> Thank you Dana and Michael for your replies, I am getting a spa3000 in the mail soon so I can try it out and see if it will work for my needs, I am going to implement a automatic attendant thanks to the information provided. Since I have not had any replies about the atom board I am guessing that nobody has used one, Could somebody tell me what is a good CPU speed / Memory / FSB be? I really do not have a large budget and cannot afford to buy something that will not work. Thanks again Orien From frank at carmickle.com Thu Nov 12 19:04:30 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 12 Nov 2009 22:04:30 -0500 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: <4AFCC829.2070507@tx.rr.com> References: <4AFCC829.2070507@tx.rr.com> Message-ID: <20091113030429.GS11697@base.carmickle.com> On Thu, Nov 12, Orien Love wrote: > Since I have not had any replies about the atom board I am guessing > that nobody has used one, Could somebody tell me what is a good CPU > speed / Memory / FSB be? > I really do not have a large budget and cannot afford to buy > something that will not work. I have not used an Atom board yet but a few are in the plans. If you do any of them the 330 is the only one to go with as of now. 64 bit and dual core in 8w is pretty nice but then again I don't have one to test with so I can't say for sure. --FC From paul.thirumalai at gmail.com Thu Nov 12 22:27:05 2009 From: paul.thirumalai at gmail.com (Paul Thirumalai) Date: Thu, 12 Nov 2009 22:27:05 -0800 Subject: [Freeswitch-users] Configuring freeswitch with voicepulse In-Reply-To: <900c9adf0911092137vf45ec94ie7473d2c08e5ae12@mail.gmail.com> References: <900c9adf0911092137vf45ec94ie7473d2c08e5ae12@mail.gmail.com> Message-ID: <900c9adf0911122227l74cc638eq5dfab22e6e21caf@mail.gmail.com> Hi Jason Thanks for your response, I setup the configuration with 2 proxies based on the example of the freeswitch wiki. I looked at freeswitch.log and found the following line. Dialplan: sofia/internal/1000 at 74.207.249.79 Action set(effective_caller_id_number=12223334444) Dialplan: sofia/internal/1000 at 74.207.249.79 Action bridge(sofia/gateway/voicepulse/5035440933) 2009-11-13 01:16:23.653519 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/1000 at 74.207.249.79) State Change CS_ROUTING -> CS_EXECUTE 2009-11-13 01:16:23.653519 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1000 at 74.207.249.79 [BREAK] 2009-11-13 01:16:23.653519 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/1000 at 74.207.249.79) State ROUTING going to sleep 2009-11-13 01:16:23.653519 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1000 at 74.207.249.79) Running State Change CS_EXECUTE 2009-11-13 01:16:23.653519 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1000 at 74.207.249.79) State EXECUTE 2009-11-13 01:16:23.653519 [DEBUG] mod_sofia.c:173 sofia/internal/ 1000 at 74.207.249.79 SOFIA EXECUTE 2009-11-13 01:16:23.653519 [DEBUG] switch_core_state_machine.c:151 sofia/internal/1000 at 74.207.249.79 Standard EXECUTE EXECUTE sofia/internal/1000 at 74.207.249.79hash(insert/74.207.249.79-spymap/1000/115be3f6-d01c-11de-8360-976b377ef920) EXECUTE sofia/internal/1000 at 74.207.249.79hash(insert/74.207.249.79-last_dial/1000/5035440933) If this makes sense to someone ,could you please gently guide me in the right direction. Thanks Paul On Mon, Nov 9, 2009 at 9:37 PM, Paul Thirumalai wrote: > Hello All > I am trying to configure freeswitch so that it sends outgoing calls to the > PSTN through voicepulse > My configuration is as follows. > I created a file $PREFIX/conf/sip_profiles/external/voicepulse.xml > > > > > > > > > > > > > > > > > > > > > > > I also have a dial plan defined as follows > > > > data="effective_caller_id_number=12223334444"/> > > > > > > > > When I dial an external number using extension 1000 I get the following > message on the CLI > > ] > freeswitch at ubuntu> 2009-11-10 00:35:44.365614 [NOTICE] > switch_channel.c:602 New Channel sofia/internal/1000 at 74.207.249.79[e4301180-cdba-11de-a864-8927fe94a9f0] > 2009-11-10 00:35:44.366623 [INFO] mod_dialplan_xml.c:315 Processing > Paul->5555555555 in context default > 2009-11-10 00:35:44.368645 [NOTICE] switch_channel.c:602 New Channel > sofia/external/5555555555 [e43092f4-cdba-11de-a864-8927fe94a9f0] > 2009-11-10 00:35:47.59221 [NOTICE] sofia_glue.c:2698 Pre-Answer > sofia/external/5555555555! > 2009-11-10 00:35:47.59221 [INFO] switch_ivr_originate.c:2017 Sending early > media > 2009-11-10 00:35:47.60524 [INFO] mod_sofia.c:1506 Ring SDP: > v=0 > o=FreeSWITCH 1257800805 1257800806 IN IP4 74.207.249.79 > s=FreeSWITCH > c=IN IP4 74.207.249.79 > t=0 0 > m=audio 30542 RTP/AVP 0 101 > a=rtpmap:0 pcmu/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2009-11-10 00:35:47.60524 [NOTICE] mod_sofia.c:1509 Pre-Answer > sofia/internal/1000 at 74.207.249.79! > 2009-11-10 00:35:51.449542 [NOTICE] sofia.c:3849 Hangup > sofia/external/5555555555 [CS_EXCHANGE_MEDIA] [NORMAL_TEMPORARY_FAILURE] > 2009-11-10 00:35:51.452539 [NOTICE] switch_ivr_bridge.c:419 Hangup > sofia/internal/1000 at 74.207.249.79 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 1 > (sofia/internal/1000 at 74.207.249.79) Ended > 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/1000 at 74.207.249.79 [CS_DESTROY] > 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 2 > (sofia/external/5555555555) Ended > 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/external/5555555555 [CS_DESTROY] > > > I am really new to VOIP and having a hard time with this. I am really not > sure how to proceed. Any help would be really appreciated. > > Thanks > Paul > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091112/97055a96/attachment-0002.html From siniypin at gmail.com Thu Nov 12 23:30:05 2009 From: siniypin at gmail.com (RobertT) Date: Fri, 13 Nov 2009 10:30:05 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> Message-ID: <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> but FS does use tcp for that call leg -> RX 1167 bytes ... from *tcp* ...: And after all there can be other SIP transports combinations FS should interconnect... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/c2e9bdb3/attachment-0002.html From viper at fx-services.com Fri Nov 13 02:23:05 2009 From: viper at fx-services.com (Robin Vleij) Date: Fri, 13 Nov 2009 11:23:05 +0100 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> Message-ID: <4AFD3389.6090409@fx-services.com> On 11/13/09 2:49 AM, Eliot Gable wrote: Hi Eliot, > Or, of course, there is always mod_xml_curl. Basically, XML dialplan > on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a > web application server, web application server responds with XML > routing response, FreeSWITCH routes the call. Yeah, been looking at that one, really cool idea. Then I could build my routing database in any way I want. I'm just worried about performance and the extra delay it'll introduce. But technically with my complex routing demands this would be the right solution, instead of a mix of modules (which probably brings the same extra load on the machine). I'll fiddle a bit. :) /Robin From viper at fx-services.com Fri Nov 13 02:25:09 2009 From: viper at fx-services.com (Robin Vleij) Date: Fri, 13 Nov 2009 11:25:09 +0100 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> Message-ID: <4AFD3405.3040902@fx-services.com> On 11/12/09 11:53 PM, Rupa Schomaker wrote: Hi, > lcr is based on prefix, so the boundaries for which the range is > assigned may not match a prefix. You may be better off either: OK, I think I can forget lcr, it's too far off what I want to do to make it work with some fixes. > 1) denormalize your ranges and just insert all distinct #s > 2) Modify mod_easyroute to support ranges Will look at this. When I come up with something interesting I'll put it on the wiki. In the meantime I've begun looking at xml_curl also, that might in the end really be the best one. I can then build a database+php that responds to stuff in whatever way we want. > 3) talk to SWK (he is on irc here and there) about his (non free) > fancier routing options If I can't get it like I want, I'll look for him. :) Thanks for all the pointers and help! /Robin From juanbackson at gmail.com Fri Nov 13 05:13:12 2009 From: juanbackson at gmail.com (Juan Backson) Date: Fri, 13 Nov 2009 21:13:12 +0800 Subject: [Freeswitch-users] need desperate help with zombie channels Message-ID: <27c25bc40911130513u405062c7kb775e14d04761fd4@mail.gmail.com> Hi, I am having difficulty trying to figure out why there are bunch of zombie channels in my system. It seems to me that these zombies come from apr_thread pool. Does anyone have any idea what may be the cause of these problems? freeswitch at internal> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure b789468a-4412-490b-bc66-32f149ba4d1d,outbound,2009-11-13 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 ,,,XML,default,,,,, 7e1ecaaa-b2d8-47a0-9982-25cd44186d4e,outbound,2009-11-13 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 ,,,XML,default,,,,, 01fa2ff6-f807-4ef0-b988-70a9fe8c4536,outbound,2009-11-13 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 ,CS_EXCHANGE_MEDIA,a88999001,a88999001,192.168.1.116, 999100 at 192.168.1.116:9342,incre_call_stat,125 165 182 235 13 3184093 0,XML,default,,,,, 0271541f-f0b5-482c-b05d-b196f85121be,inbound,2009-11-13 20:15:35,1258114535,sofia/external/88999001 at 192.168.1.116:7342 ,CS_EXECUTE,sipp,88999001,192.168.1.116,88999100,hangup,NORMAL_CLEARING,XML,default,,,,, 7e4ccfec-a4ad-4817-9a82-f1166b34576f,outbound,2009-11-13 20:15:35,1258114536,sofia/external/999100 at 192.168.1.116:9342 ,CS_CONSUME_MEDIA,a88999001,a88999001,192.168.1.116, 999100 at 192.168.1.116:9342,,,XML,default,,,,, 5 total. freeswitch at internal> uuid_kill b789468a-4412-490b-bc66-32f149ba4d1d -ERR No Such Channel! These channels actually do not exist in the system! Here is my gcore output with 5 zombies out of 100K test calls : Thread 21 (process 8946): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 128000} #2 0x00002b3cb3bfb8ca in switch_console_loop () at src/switch_console.c:819 arg = 1 thread = (switch_thread_t *) 0x2aaab00320d0 thd_attr = (switch_threadattr_t *) 0x2aaab0032070 pool = (switch_memory_pool_t *) 0x2aaab0031f88 __func__ = "switch_console_loop" __PRETTY_FUNCTION__ = "switch_console_loop" #3 0x0000000000402884 in main (argc=1, argv=) at src/switch.c:753 pid_path = "/usr/local/freeswitch/log/freeswitch.pid", '\0' pid_buffer = "8946", '\0' old_pid_buffer = '\0' pid_len = 4 old_pid_len = 4198811 err = 0x2b3cb3cec77d "Success" ---Type to continue, or q to quit--- nf = 0 runas_user = runas_group = nc = 0 pid = x = opts = opts_str = '\0' local_argv = {0x7ffff6f08c15 "./freeswitch", 0x0 } arg_argv = {0x0 } alt_dirs = 0 known_opt = high_prio = 0 flags = 65 ret = destroy_status = fd = (switch_file_t *) 0xb6293e0 pool = (switch_memory_pool_t *) 0xb629368 rlp = {rlim_cur = 245760, rlim_max = 245760} waste = 0 __PRETTY_FUNCTION__ = "main" Thread 20 (process 20699): ---Type to continue, or q to quit--- #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 0} #2 0x00002aaaab35e926 in read_packet (listener=0x2aaae7523d08, event=0x2aab3b5ab058, timeout=0) at mod_event_socket.c:1255 do_sleep = 1 '\001' mlen = 0 bytes = 0 mbuf = '\0' buf = '\0' len = 123 status = SWITCH_STATUS_BREAK count = start = 1258117263 pop = (void *) 0x2aaad12f6540 ptr = 0x2aab3b5a98a0 "" crcount = 0 '\0' channel = (switch_channel_t *) 0x0 clen = __func__ = "read_packet" __PRETTY_FUNCTION__ = "read_packet" ---Type to continue, or q to quit--- #3 0x00002aaaab36347a in listener_run (thread=, obj=0x2aaae7523d08) at mod_event_socket.c:2093 listener = (listener_t *) 0x0 buf = '\0' len = 1024 status = event = (switch_event_t *) 0x0 reply = "\000OK log level [7]", '\0' session = (switch_core_session_t *) 0x0 channel = revent = (switch_event_t *) 0x0 var = __PRETTY_FUNCTION__ = "listener_run" __func__ = "listener_run" #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 19 (process 14505): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. ---Type to continue, or q to quit--- #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaaf49798, data=0x7afe0080) at misc/apr_queue.c:276 rv = 0 #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( thread=, obj=) at src/switch_event.c:248 pop = (void *) 0x0 event = (switch_event_t *) 0x0 queue = (switch_queue_t *) 0x2aaaaaf49798 my_id = 1 __func__ = "switch_event_dispatch_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 18 (process 9334): #0 0x0000003054e0d2cb in read () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3cd50c8 in read_char (el=0x2aaab0028180, cp=0x4027002f "") at read.c:294 num_read = 1076297860 tried = 0 ---Type to continue, or q to quit--- #2 0x00002b3cb3cd4ceb in el_gets (el=0x2aaab0028180, nread=0x40270084) at read.c:241 cmdnum = 112 'p' num = -1321754256 ch = 0 '\0' #3 0x00002b3cb3bfc4bb in console_thread (thread=, obj=) at src/switch_console.c:464 arg = 1 count = 1 line = 0x2aaab0034e70 "\n" pool = (switch_memory_pool_t *) 0x2aaab0031f88 __func__ = "console_thread" __PRETTY_FUNCTION__ = "console_thread" #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 17 (process 9333): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 ---Type to continue, or q to quit--- tv = {tv_sec = 0, tv_usec = 0} #2 0x00002b3cb3c53895 in softtimer_runtime () at src/switch_time.c:464 current_ms = 692 x = 690 tick = 292 ts = last = 1258117283599783 fwd_errs = 0 rev_errs = 0 __func__ = "softtimer_runtime" #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x0, obj=0x0) at src/switch_loadable_module.c:94 status = ts = (switch_core_thread_session_t *) 0x0 module = (switch_loadable_module_t *) 0xb6c4e00 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 16 (process 9332): ---Type to continue, or q to quit--- #0 0x0000003054e0d4eb in accept () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c707a4 in apr_socket_accept (new=0x416b4020, sock=0xbcfde38, connection_context=0x2aaacda27718) at network_io/unix/sockets.c:187 No locals. #2 0x00002aaaab35f889 in mod_event_socket_runtime () at mod_event_socket.c:2324 pool = (switch_memory_pool_t *) 0xbcfdc88 listener_pool = (switch_memory_pool_t *) 0x2aaacda27718 rv = sa = (switch_sockaddr_t *) 0xbcfdd68 inbound_socket = (switch_socket_t *) 0x2aaacda277f8 listener = x = __func__ = "mod_event_socket_runtime" #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x14f, obj=0x2aaacda27948) at src/switch_loadable_module.c:94 status = ts = (switch_core_thread_session_t *) 0x2aaacda27948 module = (switch_loadable_module_t *) 0x2aaaac0058c0 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 ---Type to continue, or q to quit--- No symbol table info available. #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 15 (process 9330): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 55000} #2 0x00002aaab503cc4c in node_thread_run (thread=, obj=) at mod_fifo.c:580 val = (void *) 0x0 var = (const void *) 0x0 idle_consumers = hi = (switch_hash_index_t *) 0x0 ppl_waiting = 0 consumer_total = 1087699264 node = #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. ---Type to continue, or q to quit--- Thread 14 (process 9329): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 100} #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( thread=, obj=) at sofia.c:763 profile = (sofia_profile_t *) 0xbce2310 ireg_loops = 18 gateway_loops = 0 loops = 72 qsize = 4294966782 pop = (void *) 0x0 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 13 (process 9328): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 ---Type to continue, or q to quit--- No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 0} #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( thread=, obj=) at sofia.c:763 profile = (sofia_profile_t *) 0x2aaab000eb10 ireg_loops = 5 gateway_loops = 0 loops = 93 qsize = 4294966782 pop = (void *) 0x0 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 12 (process 9327): #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 No symbol table info available. #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce71c0, tout=1000) at su_epoll_port.c:495 ---Type to continue, or q to quit--- j = 198076976 n = 0 events = 0 index = 10922 version = 3 M = 4 ev = 0x41204ef0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce71c0) at su_base_port.c:349 tout = 1000 tout2 = 0 __PRETTY_FUNCTION__ = "su_base_port_run" #3 0x00002aaab45c6c51 in su_port_run (self=0xbce71c0) at su_port.h:326 base = (su_virtual_port_t *) 0xbce71c0 #4 0x00002aaab45c6c29 in su_root_run (self=0xbce72a0) at su_root.c:819 __PRETTY_FUNCTION__ = "su_root_run" #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404f7ac0) at su_pthread_port.c:324 arg = (struct clone_args *) 0x0 task = {{sut_port = 0xbce71c0, sut_root = 0xbce72a0}} zap = 1 #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 ---Type to continue, or q to quit--- No symbol table info available. #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 11 (process 9326): #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 No symbol table info available. #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce78b0, tout=1000) at su_epoll_port.c:495 j = -1342070512 n = 10922 events = 0 index = 10922 version = 3 M = 4 ev = 0x411c8ef0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce78b0) at su_base_port.c:349 tout = 1000 tout2 = 0 __PRETTY_FUNCTION__ = "su_base_port_run" #3 0x00002aaab45c6c51 in su_port_run (self=0xbce78b0) at su_port.h:326 ---Type to continue, or q to quit--- base = (su_virtual_port_t *) 0xbce78b0 #4 0x00002aaab45c6c29 in su_root_run (self=0x2aaab001a060) at su_root.c:819 __PRETTY_FUNCTION__ = "su_root_run" #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404bbac0) at su_pthread_port.c:324 arg = (struct clone_args *) 0x0 task = {{sut_port = 0xbce78b0, sut_root = 0x2aaab001a060}} zap = 1 #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 10 (process 9325): #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 No symbol table info available. #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce6c30, tout=1000) at su_epoll_port.c:495 j = -1268971119 n = 10922 events = 0 index = 0 version = 1 ---Type to continue, or q to quit--- M = 4 ev = 0x404f7c40 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbce6c30, tout=1000) at su_base_port.c:467 now = {tv_sec = 3467106082, tv_usec = 971475} __PRETTY_FUNCTION__ = "su_base_port_step" #3 0x00002aaab45c6d6a in su_port_step (self=0xbce6c30, tout=1000) at su_port.h:340 base = (su_virtual_port_t *) 0xbce6c30 #4 0x00002aaab45c6d32 in su_root_step (self=0xbce4650, tout=1000) at su_root.c:858 __PRETTY_FUNCTION__ = "su_root_step" #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( thread=, obj=) at sofia.c:973 profile = (sofia_profile_t *) 0xbce2310 pool = node = (sip_alias_node_t *) 0x0 s_event = (switch_event_t *) 0x0 sanity = worker_thread = (switch_thread_t *) 0xbce36a0 st = SWITCH_STATUS_SUCCESS __func__ = "sofia_profile_thread_run" ---Type to continue, or q to quit--- __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 9 (process 9324): #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 No symbol table info available. #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbcdffb0, tout=1000) at su_epoll_port.c:495 j = -1268971119 n = 10922 events = 0 index = 0 version = 1 M = 4 ev = 0x404bbc40 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbcdffb0, tout=1000) at su_base_port.c:467 now = {tv_sec = 3467106083, tv_usec = 525146} __PRETTY_FUNCTION__ = "su_base_port_step" ---Type to continue, or q to quit--- #3 0x00002aaab45c6d6a in su_port_step (self=0xbcdffb0, tout=1000) at su_port.h:340 base = (su_virtual_port_t *) 0xbcdffb0 #4 0x00002aaab45c6d32 in su_root_step (self=0xbcdfe00, tout=1000) at su_root.c:858 __PRETTY_FUNCTION__ = "su_root_step" #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( thread=, obj=) at sofia.c:973 profile = (sofia_profile_t *) 0x2aaab000eb10 pool = node = (sip_alias_node_t *) 0x0 s_event = (switch_event_t *) 0x0 sanity = worker_thread = (switch_thread_t *) 0x2aaab000fea0 st = SWITCH_STATUS_SUCCESS __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 8 (process 8999): ---Type to continue, or q to quit--- #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 444000} #2 0x00002b3cb3c14e2a in switch_scheduler_task_thread ( thread=, obj=) at src/switch_scheduler.c:171 __func__ = "switch_scheduler_task_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 7 (process 8998): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 100} #2 0x00002b3cb3c054f5 in switch_core_sql_thread ( thread=, obj=) at src/switch_core_sqldb.c:220 ---Type to continue, or q to quit--- pop = (void *) 0x2aaabf3d6220 itterations = 0 trans = 0 '\0' nothing_in_queue = 1 '\001' len = 100 sql_len = 4844546 sqlbuf = 0x2aab135c7010 "" sql = newlen = lc = 0 __PRETTY_FUNCTION__ = "switch_core_sql_thread" __func__ = "switch_core_sql_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 6 (process 8995): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb64c158, data=0x40893088) at misc/apr_queue.c:276 ---Type to continue, or q to quit--- rv = 0 #2 0x00002b3cb3c48ff1 in log_thread (t=, obj=) at src/switch_log.c:288 pop = (void *) 0x0 node = (switch_log_node_t *) 0x0 binding = (switch_log_binding_t *) 0x0 __func__ = "log_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 5 (process 8951): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaac355a8, data=0x40bec070) at misc/apr_queue.c:276 rv = 0 #2 0x00002b3cb3c1fb14 in switch_event_thread (thread=, obj=) at src/switch_event.c:291 pop = (void *) 0x0 event = ---Type to continue, or q to quit--- queue = (switch_queue_t *) 0x2aaaaac355a8 index = 0 my_id = 2 __func__ = "switch_event_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 4 (process 8950): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaab705a8, data=0x4060a070) at misc/apr_queue.c:276 rv = 0 #2 0x00002b3cb3c1fb14 in switch_event_thread (thread=, obj=) at src/switch_event.c:291 pop = (void *) 0x0 event = queue = (switch_queue_t *) 0x2aaaaab705a8 index = 0 my_id = 1 ---Type to continue, or q to quit--- __func__ = "switch_event_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 3 (process 8949): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb638fa8, data=0x405ce070) at misc/apr_queue.c:276 rv = 0 #2 0x00002b3cb3c1fb14 in switch_event_thread (thread=, obj=) at src/switch_event.c:291 pop = (void *) 0x0 event = queue = (switch_queue_t *) 0xb638fa8 index = 0 my_id = 0 __func__ = "switch_event_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. ---Type to continue, or q to quit--- #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. Thread 2 (process 8948): #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaacfa5a8, data=0x40592080) at misc/apr_queue.c:276 rv = 0 #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( thread=, obj=) at src/switch_event.c:248 pop = (void *) 0x0 event = (switch_event_t *) 0x0 queue = (switch_queue_t *) 0x2aaaaacfa5a8 my_id = 0 __func__ = "switch_event_dispatch_thread" #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. ---Type to continue, or q to quit--- Thread 1 (process 8947): #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 No symbol table info available. #1 0x00002b3cb3c72df5 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 451000} #2 0x00002b3cb3c00c95 in pool_thread (thread=, obj=) at src/switch_core_memory.c:490 x = #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 No symbol table info available. (gdb) (gdb) (gdb) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/4426c281/attachment-0002.html From piotr_zurek at biprotech.com Fri Nov 13 05:59:22 2009 From: piotr_zurek at biprotech.com (=?UTF-8?B?UGlvdHIgxbt1cmVr?=) Date: Fri, 13 Nov 2009 14:59:22 +0100 Subject: [Freeswitch-users] How to pick up someone's phone remotely. In-Reply-To: <4AFC005A.4090200@biprotech.com> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> <4AFC005A.4090200@biprotech.com> Message-ID: <4AFD663A.1030707@biprotech.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/514ab68c/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: piotr_zurek.vcf Type: text/x-vcard Size: 414 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/514ab68c/attachment-0002.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3678 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/514ab68c/attachment-0002.bin From tayeb.meftah at gmail.com Fri Nov 13 08:31:38 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Fri, 13 Nov 2009 16:31:38 +0000 Subject: [Freeswitch-users] SIP trunk without authentication In-Reply-To: <1352396721.20091110232720@mail.ru> References: <1352396721.20091110232720@mail.ru> Message-ID: <4AFD89EA.2000905@gmail.com> hi use sofia/internal/$1 at your provider ip where $1 is the number thanks Sergey Kobzar a ?crit : > Hello. > > I'm FS newbie and want connect it to SIP provider which does not > require authentication - it make authentication using my IP. > > I've searched through FS documentation and didn't find clear answer. > > Could you help me or maybe give a link to a doc which can help? > > Thanks. > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From anthony.minessale at gmail.com Fri Nov 13 07:40:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Nov 2009 09:40:52 -0600 Subject: [Freeswitch-users] need desperate help with zombie channels In-Reply-To: <27c25bc40911130513u405062c7kb775e14d04761fd4@mail.gmail.com> References: <27c25bc40911130513u405062c7kb775e14d04761fd4@mail.gmail.com> Message-ID: <191c3a030911130740n200b54d1o3785cc703d8d1438@mail.gmail.com> >From the looks of that you probably have an equal number of zombie event socket processes. We do not get involved in load testing. Consider consulting at freeswitch.orgfor professional help. On Fri, Nov 13, 2009 at 7:13 AM, Juan Backson wrote: > Hi, > > I am having difficulty trying to figure out why there are bunch of zombie > channels in my system. It seems to me that these zombies come from > apr_thread pool. > > Does anyone have any idea what may be the cause of these problems? > > > freeswitch at internal> show channels > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure > b789468a-4412-490b-bc66-32f149ba4d1d,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 > ,,,XML,default,,,,, > 7e1ecaaa-b2d8-47a0-9982-25cd44186d4e,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 > ,,,XML,default,,,,, > 01fa2ff6-f807-4ef0-b988-70a9fe8c4536,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_EXCHANGE_MEDIA,a88999001,a88999001,192.168.1.116, > 999100 at 192.168.1.116:9342,incre_call_stat,125 165 182 235 13 3184093 > 0,XML,default,,,,, > 0271541f-f0b5-482c-b05d-b196f85121be,inbound,2009-11-13 > 20:15:35,1258114535,sofia/external/88999001 at 192.168.1.116:7342 > ,CS_EXECUTE,sipp,88999001,192.168.1.116,88999100,hangup,NORMAL_CLEARING,XML,default,,,,, > 7e4ccfec-a4ad-4817-9a82-f1166b34576f,outbound,2009-11-13 > 20:15:35,1258114536,sofia/external/999100 at 192.168.1.116:9342 > ,CS_CONSUME_MEDIA,a88999001,a88999001,192.168.1.116, > 999100 at 192.168.1.116:9342,,,XML,default,,,,, > > 5 total. > > freeswitch at internal> uuid_kill b789468a-4412-490b-bc66-32f149ba4d1d > -ERR No Such Channel! > > These channels actually do not exist in the system! > > > Here is my gcore output with 5 zombies out of 100K test calls : > > > Thread 21 (process 8946): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 128000} > #2 0x00002b3cb3bfb8ca in switch_console_loop () at > src/switch_console.c:819 > arg = 1 > thread = (switch_thread_t *) 0x2aaab00320d0 > thd_attr = (switch_threadattr_t *) 0x2aaab0032070 > pool = (switch_memory_pool_t *) 0x2aaab0031f88 > __func__ = "switch_console_loop" > __PRETTY_FUNCTION__ = "switch_console_loop" > #3 0x0000000000402884 in main (argc=1, argv=) > at src/switch.c:753 > pid_path = "/usr/local/freeswitch/log/freeswitch.pid", '\0' > > pid_buffer = "8946", '\0' > old_pid_buffer = '\0' > pid_len = 4 > old_pid_len = 4198811 > err = 0x2b3cb3cec77d "Success" > ---Type to continue, or q to quit--- > nf = 0 > runas_user = > runas_group = > nc = 0 > pid = > x = > opts = > opts_str = '\0' > local_argv = {0x7ffff6f08c15 "./freeswitch", 0x0 times>} > arg_argv = {0x0 } > alt_dirs = 0 > known_opt = > high_prio = 0 > flags = 65 > ret = > destroy_status = > fd = (switch_file_t *) 0xb6293e0 > pool = (switch_memory_pool_t *) 0xb629368 > rlp = {rlim_cur = 245760, rlim_max = 245760} > waste = 0 > __PRETTY_FUNCTION__ = "main" > > Thread 20 (process 20699): > ---Type to continue, or q to quit--- > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002aaaab35e926 in read_packet (listener=0x2aaae7523d08, > event=0x2aab3b5ab058, timeout=0) at mod_event_socket.c:1255 > do_sleep = 1 '\001' > mlen = 0 > bytes = 0 > mbuf = '\0' > buf = '\0' > len = 123 > status = SWITCH_STATUS_BREAK > count = > start = 1258117263 > pop = (void *) 0x2aaad12f6540 > ptr = 0x2aab3b5a98a0 "" > crcount = 0 '\0' > channel = (switch_channel_t *) 0x0 > clen = > __func__ = "read_packet" > __PRETTY_FUNCTION__ = "read_packet" > ---Type to continue, or q to quit--- > #3 0x00002aaaab36347a in listener_run (thread=, > obj=0x2aaae7523d08) at mod_event_socket.c:2093 > listener = (listener_t *) 0x0 > buf = '\0' > len = 1024 > status = > event = (switch_event_t *) 0x0 > reply = "\000OK log level [7]", '\0' > session = (switch_core_session_t *) 0x0 > channel = > revent = (switch_event_t *) 0x0 > var = > __PRETTY_FUNCTION__ = "listener_run" > __func__ = "listener_run" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 19 (process 14505): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaaf49798, > data=0x7afe0080) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( > thread=, obj=) > at src/switch_event.c:248 > pop = (void *) 0x0 > event = (switch_event_t *) 0x0 > queue = (switch_queue_t *) 0x2aaaaaf49798 > my_id = 1 > __func__ = "switch_event_dispatch_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 18 (process 9334): > #0 0x0000003054e0d2cb in read () from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3cd50c8 in read_char (el=0x2aaab0028180, cp=0x4027002f "") > at read.c:294 > num_read = 1076297860 > tried = 0 > ---Type to continue, or q to quit--- > #2 0x00002b3cb3cd4ceb in el_gets (el=0x2aaab0028180, nread=0x40270084) > at read.c:241 > cmdnum = 112 'p' > num = -1321754256 > ch = 0 '\0' > #3 0x00002b3cb3bfc4bb in console_thread (thread=, > obj=) at src/switch_console.c:464 > arg = 1 > count = 1 > line = 0x2aaab0034e70 "\n" > pool = (switch_memory_pool_t *) 0x2aaab0031f88 > __func__ = "console_thread" > __PRETTY_FUNCTION__ = "console_thread" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 17 (process 9333): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > ---Type to continue, or q to quit--- > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002b3cb3c53895 in softtimer_runtime () at src/switch_time.c:464 > current_ms = 692 > x = 690 > tick = 292 > ts = > last = 1258117283599783 > fwd_errs = 0 > rev_errs = 0 > __func__ = "softtimer_runtime" > #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x0, obj=0x0) > at src/switch_loadable_module.c:94 > status = > ts = (switch_core_thread_session_t *) 0x0 > module = (switch_loadable_module_t *) 0xb6c4e00 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 16 (process 9332): > ---Type to continue, or q to quit--- > #0 0x0000003054e0d4eb in accept () from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c707a4 in apr_socket_accept (new=0x416b4020, > sock=0xbcfde38, > connection_context=0x2aaacda27718) at network_io/unix/sockets.c:187 > No locals. > #2 0x00002aaaab35f889 in mod_event_socket_runtime () > at mod_event_socket.c:2324 > pool = (switch_memory_pool_t *) 0xbcfdc88 > listener_pool = (switch_memory_pool_t *) 0x2aaacda27718 > rv = > sa = (switch_sockaddr_t *) 0xbcfdd68 > inbound_socket = (switch_socket_t *) 0x2aaacda277f8 > listener = > x = > __func__ = "mod_event_socket_runtime" > #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x14f, > obj=0x2aaacda27948) at src/switch_loadable_module.c:94 > status = > ts = (switch_core_thread_session_t *) 0x2aaacda27948 > module = (switch_loadable_module_t *) 0x2aaaac0058c0 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > ---Type to continue, or q to quit--- > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 15 (process 9330): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 55000} > #2 0x00002aaab503cc4c in node_thread_run (thread=, > obj=) at mod_fifo.c:580 > val = (void *) 0x0 > var = (const void *) 0x0 > idle_consumers = > hi = (switch_hash_index_t *) 0x0 > ppl_waiting = 0 > consumer_total = 1087699264 > node = > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > ---Type to continue, or q to quit--- > > Thread 14 (process 9329): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100} > #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( > thread=, obj=) at sofia.c:763 > profile = (sofia_profile_t *) 0xbce2310 > ireg_loops = 18 > gateway_loops = 0 > loops = 72 > qsize = 4294966782 > pop = (void *) 0x0 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 13 (process 9328): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > ---Type to continue, or q to quit--- > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( > thread=, obj=) at sofia.c:763 > profile = (sofia_profile_t *) 0x2aaab000eb10 > ireg_loops = 5 > gateway_loops = 0 > loops = 93 > qsize = 4294966782 > pop = (void *) 0x0 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 12 (process 9327): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce71c0, > tout=1000) > at su_epoll_port.c:495 > ---Type to continue, or q to quit--- > j = 198076976 > n = 0 > events = 0 > index = 10922 > version = 3 > M = 4 > ev = 0x41204ef0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce71c0) > at su_base_port.c:349 > tout = 1000 > tout2 = 0 > __PRETTY_FUNCTION__ = "su_base_port_run" > #3 0x00002aaab45c6c51 in su_port_run (self=0xbce71c0) at su_port.h:326 > base = (su_virtual_port_t *) 0xbce71c0 > #4 0x00002aaab45c6c29 in su_root_run (self=0xbce72a0) at su_root.c:819 > __PRETTY_FUNCTION__ = "su_root_run" > #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404f7ac0) > at su_pthread_port.c:324 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0xbce71c0, sut_root = 0xbce72a0}} > zap = 1 > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > ---Type to continue, or q to quit--- > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 11 (process 9326): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce78b0, > tout=1000) > at su_epoll_port.c:495 > j = -1342070512 > n = 10922 > events = 0 > index = 10922 > version = 3 > M = 4 > ev = 0x411c8ef0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce78b0) > at su_base_port.c:349 > tout = 1000 > tout2 = 0 > __PRETTY_FUNCTION__ = "su_base_port_run" > #3 0x00002aaab45c6c51 in su_port_run (self=0xbce78b0) at su_port.h:326 > ---Type to continue, or q to quit--- > base = (su_virtual_port_t *) 0xbce78b0 > #4 0x00002aaab45c6c29 in su_root_run (self=0x2aaab001a060) at > su_root.c:819 > __PRETTY_FUNCTION__ = "su_root_run" > #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404bbac0) > at su_pthread_port.c:324 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0xbce78b0, sut_root = 0x2aaab001a060}} > zap = 1 > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 10 (process 9325): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce6c30, > tout=1000) > at su_epoll_port.c:495 > j = -1268971119 > n = 10922 > events = 0 > index = 0 > version = 1 > ---Type to continue, or q to quit--- > M = 4 > ev = 0x404f7c40 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbce6c30, tout=1000) > at su_base_port.c:467 > now = {tv_sec = 3467106082, tv_usec = 971475} > __PRETTY_FUNCTION__ = "su_base_port_step" > #3 0x00002aaab45c6d6a in su_port_step (self=0xbce6c30, tout=1000) > at su_port.h:340 > base = (su_virtual_port_t *) 0xbce6c30 > #4 0x00002aaab45c6d32 in su_root_step (self=0xbce4650, tout=1000) > at su_root.c:858 > __PRETTY_FUNCTION__ = "su_root_step" > #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( > thread=, obj=) at sofia.c:973 > profile = (sofia_profile_t *) 0xbce2310 > pool = > node = (sip_alias_node_t *) 0x0 > s_event = (switch_event_t *) 0x0 > sanity = > worker_thread = (switch_thread_t *) 0xbce36a0 > st = SWITCH_STATUS_SUCCESS > __func__ = "sofia_profile_thread_run" > ---Type to continue, or q to quit--- > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 9 (process 9324): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbcdffb0, > tout=1000) > at su_epoll_port.c:495 > j = -1268971119 > n = 10922 > events = 0 > index = 0 > version = 1 > M = 4 > ev = 0x404bbc40 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbcdffb0, tout=1000) > at su_base_port.c:467 > now = {tv_sec = 3467106083, tv_usec = 525146} > __PRETTY_FUNCTION__ = "su_base_port_step" > ---Type to continue, or q to quit--- > #3 0x00002aaab45c6d6a in su_port_step (self=0xbcdffb0, tout=1000) > at su_port.h:340 > base = (su_virtual_port_t *) 0xbcdffb0 > #4 0x00002aaab45c6d32 in su_root_step (self=0xbcdfe00, tout=1000) > at su_root.c:858 > __PRETTY_FUNCTION__ = "su_root_step" > #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( > thread=, obj=) at sofia.c:973 > profile = (sofia_profile_t *) 0x2aaab000eb10 > pool = > node = (sip_alias_node_t *) 0x0 > s_event = (switch_event_t *) 0x0 > sanity = > worker_thread = (switch_thread_t *) 0x2aaab000fea0 > st = SWITCH_STATUS_SUCCESS > __func__ = "sofia_profile_thread_run" > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 8 (process 8999): > ---Type to continue, or q to quit--- > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 444000} > #2 0x00002b3cb3c14e2a in switch_scheduler_task_thread ( > thread=, obj=) > at src/switch_scheduler.c:171 > __func__ = "switch_scheduler_task_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 7 (process 8998): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100} > #2 0x00002b3cb3c054f5 in switch_core_sql_thread ( > thread=, obj=) > at src/switch_core_sqldb.c:220 > ---Type to continue, or q to quit--- > pop = (void *) 0x2aaabf3d6220 > itterations = 0 > trans = 0 '\0' > nothing_in_queue = 1 '\001' > len = 100 > sql_len = 4844546 > sqlbuf = 0x2aab135c7010 "" > sql = > newlen = > lc = 0 > __PRETTY_FUNCTION__ = "switch_core_sql_thread" > __func__ = "switch_core_sql_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 6 (process 8995): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb64c158, data=0x40893088) > at misc/apr_queue.c:276 > ---Type to continue, or q to quit--- > rv = 0 > #2 0x00002b3cb3c48ff1 in log_thread (t=, > obj=) at src/switch_log.c:288 > pop = (void *) 0x0 > node = (switch_log_node_t *) 0x0 > binding = (switch_log_binding_t *) 0x0 > __func__ = "log_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 5 (process 8951): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaac355a8, > data=0x40bec070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > ---Type to continue, or q to quit--- > queue = (switch_queue_t *) 0x2aaaaac355a8 > index = 0 > my_id = 2 > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 4 (process 8950): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaab705a8, > data=0x4060a070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > queue = (switch_queue_t *) 0x2aaaaab705a8 > index = 0 > my_id = 1 > ---Type to continue, or q to quit--- > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 3 (process 8949): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb638fa8, data=0x405ce070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > queue = (switch_queue_t *) 0xb638fa8 > index = 0 > my_id = 0 > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 2 (process 8948): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaacfa5a8, > data=0x40592080) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( > thread=, obj=) > at src/switch_event.c:248 > pop = (void *) 0x0 > event = (switch_event_t *) 0x0 > queue = (switch_queue_t *) 0x2aaaaacfa5a8 > my_id = 0 > __func__ = "switch_event_dispatch_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > ---Type to continue, or q to quit--- > Thread 1 (process 8947): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 451000} > #2 0x00002b3cb3c00c95 in pool_thread (thread=, > obj=) at src/switch_core_memory.c:490 > x = > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > (gdb) > (gdb) > (gdb) > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/c1957db0/attachment-0002.html From mgende at gendesign.com Fri Nov 13 08:03:32 2009 From: mgende at gendesign.com (Michael Gende) Date: Fri, 13 Nov 2009 10:03:32 -0600 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: <20091113030429.GS11697@base.carmickle.com> References: <4AFCC829.2070507@tx.rr.com> <20091113030429.GS11697@base.carmickle.com> Message-ID: Hey Orien, I've put FS on a couple of different commodity hardware platforms, from a 1U (dual CPU, dual core, Gig of memory) server to an old Dell PC (less than a gig of memory, single CPU, a few years old so its a dog) and found I had plenty of juice for a small office, say. On the FS website there is a suggested hardware list if I'm not mistaken. However, depending upon how hard you plan to hammer the system usage-wise, any above average PC platform would probably serve well, in my humble - if not entirely educated - opinion. Regards, Mike G. On Thu, Nov 12, 2009 at 9:04 PM, Frank Carmickle wrote: > On Thu, Nov 12, Orien Love wrote: > > Since I have not had any replies about the atom board I am guessing > > that nobody has used one, Could somebody tell me what is a good CPU > > speed / Memory / FSB be? > > I really do not have a large budget and cannot afford to buy > > something that will not work. > > I have not used an Atom board yet but a few are in the plans. If you do > any of them the 330 is the only one to go with as of now. 64 bit and dual > core in 8w is pretty nice but then again I don't have one to test with so I > can't say for sure. > > --FC > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/37700ebf/attachment-0002.html From jerry.richards at teotech.com Fri Nov 13 09:18:55 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 13 Nov 2009 09:18:55 -0800 Subject: [Freeswitch-users] How To Disable MD5 Authentication? Message-ID: <094BEE1BBB684AD692DD928A6E6E4EAD@greyhawk.tonecommander.com> How can I disable MD5 Authentication upon registration? Best Regards, Jerry From brian at freeswitch.org Fri Nov 13 09:26:12 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Nov 2009 11:26:12 -0600 Subject: [Freeswitch-users] How To Disable MD5 Authentication? In-Reply-To: <094BEE1BBB684AD692DD928A6E6E4EAD@greyhawk.tonecommander.com> References: <094BEE1BBB684AD692DD928A6E6E4EAD@greyhawk.tonecommander.com> Message-ID: <5185D4ED-FCAE-42B0-BE24-52BF35463DAA@freeswitch.org> Are you still wanting to authenticate users? auth-calls=false, blind- registration=true on the profile. /b On Nov 13, 2009, at 11:18 AM, Jerry Richards wrote: > > How can I disable MD5 Authentication upon registration? > > Best Regards, > Jerry > From egable+freeswitch at gmail.com Fri Nov 13 09:58:16 2009 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Fri, 13 Nov 2009 12:58:16 -0500 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: <4AFD3389.6090409@fx-services.com> References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> <4AFD3389.6090409@fx-services.com> Message-ID: Performance is not an issue. I clocked 300 calls per second on such a setup using a Dell R710 with two XEON X5570s and 32 GB RAM as the FreeSWITCH server and a Dell 2950 4-core system with 8 GB RAM as the app server. The app server was at 15% - 20% idle at that rate and the Dell R710 was 65% - 70% idle. The main bottleneck I ran into was using the limit application with ODBC. A mutex lock around the ODBC calls meant that I could only pull 160 calls per second, even though the app server was 55% - 60% idle at that rate, because the ODBC call took 1/160th of a second to complete and all the requests were serialized. In theory, you should get better performance using mod_xml_curl because FreeSWITCH will not have to parse a large XML dial plan. One of the drawbacks of the XML dial plan is that any time it tries to locate a route element, it must perform an XML linear search until it finds the correct child (as can be seen in the source code). Thus, searching the XML dialplan is O(n) operation while mod_xml_curl is typically constant time, or at worst, O(log n), depending on how you are storing / querying your data from your database system. Actually, I suppose you could just be a bad programmer and end up making it exponential, but I'm assuming you know how to write code and design your database in a way that avoids that. I have been considering writing a hash cache for the XML dialplan so that lookups can become constant time, but I have no idea when or if I will find the time to do that. :) On Fri, Nov 13, 2009 at 5:23 AM, Robin Vleij wrote: > On 11/13/09 2:49 AM, Eliot Gable wrote: > > Hi Eliot, > >> Or, of course, there is always mod_xml_curl. Basically, XML dialplan >> on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a >> web application server, web application server responds with XML >> routing response, FreeSWITCH routes the call. > > Yeah, been looking at that one, really cool idea. Then I could build my > routing database in any way I want. I'm just worried about performance > and the extra delay it'll introduce. But technically with my complex > routing demands this would be the right solution, instead of a mix of > modules (which probably brings the same extra load on the machine). > > I'll fiddle a bit. :) > > /Robin > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From egable+freeswitch at gmail.com Fri Nov 13 10:16:47 2009 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Fri, 13 Nov 2009 13:16:47 -0500 Subject: [Freeswitch-users] suggestions for hardware. In-Reply-To: <4AF4AF73.8070804@tx.rr.com> References: <4AF4AF73.8070804@tx.rr.com> Message-ID: I have built some low-wattage systems before, and from what I have seen, you want to stay away from the low-wattage processors. On a price per performance per watt scale, the lowest wattage Core 2 Duo processors are the best bet. I have a logic-supply ITX system here at home that has a VIA processor in it and it is dirt slow. It cost about $350 - $400. It sucks about 5W max, according to my Watt meter but it takes forever to do anything. Compiling FreeSWITCH on it is an absolute nightmare (it takes hours). I have an alternate MicroATX system with a Core 2 Duo and it pulls about 15W all the time. If I compile FreeSWITCH on that system, it takes a few minutes. The Core 2 Duo system was bought about 6 months after the VIA system. The Core 2 Duo cost about $270 for everything, including 2 GB of RAM. The amount of money saved by running the VIA system at 5W is nowhere close to the inconvenience of waiting for it to do anything. Also, I had to go through about 5-6 different Linux distributions before I found one that would actually install on the VIA processor. I think Suse Linux was the one that finally worked on it. Tried CentOS, Slackware, Ubuntu, Debian, and a couple of others. Now, I have not tried the Atom processor, so it could be very different. However, I have read the Tom's Hardware review that also showed that the Core 2 Duo was several times better on the price per performance per watt scale than the Atom processor, and again, it was mainly because the Atom processor took so much longer to do anything. On Fri, Nov 6, 2009 at 6:21 PM, Orien Love wrote: > First of all, Thanks to the help I received on my pfSense installation, > especially to Michael. ?I have a basic test system up and running. I am > still waiting on some hardware but the base system is working!!!! > > I am looking on advice on how to set up a simple office PBX, 20 phones > and 4 outside lines.with 2 or 3 "operator" phones and the rest will be > extensions. > > Here is my plan, please let me know if it does not make sense, or if I > am going about it > > System Hardware > ?4 spa3000's to handle the outside lines. > ?2-3 polycom 601 phones with expansion modules (Operator phones) > ?18 polycom 330 or other phones for desks. > ?2-24 port cisco POE switches > ?1 pfSense server. > > System Design. > > ?Extension Numbers 2xx > ?Outside line access 1xxxxxxxxxx > ?groups 3xx > ?auto-attendent ??? > > here are my questions > ? ?#1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of > memory handle this proposed system? (Here is the MB I am thing of using > MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) > ? ?#2 how do I pool my spa 3000 FXO lines so that the outgoing calls > use the first available line? also how do insure that metro (non long > distance) calls go to a specific line if available? > > I have learned a lot on how to set up Polycom 601 phones, I am planning > on writing a how to document, is there any specific format? > > Thanks Orien > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From jerry.richards at teotech.com Fri Nov 13 13:59:12 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 13 Nov 2009 13:59:12 -0800 Subject: [Freeswitch-users] Accessing Config Info From Database Message-ID: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> Is there a way to access configuration information from a database (e.g. SQL) rather than from the XML files? Best Regards, Jerry From pjintheusa at gmail.com Fri Nov 13 14:26:59 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 13 Nov 2009 17:26:59 -0500 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> Message-ID: <367751820911131426j46bdf6f4t78b535ea989dfccb@mail.gmail.com> Take a look at http://wiki.freeswitch.org/wiki/Mod_xml_curl to get started. On Fri, Nov 13, 2009 at 4:59 PM, Jerry Richards wrote: > Is there a way to access configuration information from a database (e.g. > SQL) rather than from the XML files? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/1edf5d84/attachment-0002.html From leon at scarlet-internet.nl Fri Nov 13 14:28:55 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Fri, 13 Nov 2009 23:28:55 +0100 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> Message-ID: <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> Hi, You can use mod_xml_curl (generate xml on a webserver): http://wiki.freeswitch.org/wiki/Mod_xml_curl or mod_xml_odbc (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Mod_xml_odbc or LUA together with luasql (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration regards, Leon On Fri, 2009-11-13 at 13:59 -0800, Jerry Richards wrote: > Is there a way to access configuration information from a database (e.g. > SQL) rather than from the XML files? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From robert.hadley at teotech.com Fri Nov 13 09:26:40 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Fri, 13 Nov 2009 09:26:40 -0800 Subject: [Freeswitch-users] Freeswitch configure error using --srcdir option Message-ID: <3894885261AF440BB1025F7488B92E97@greyhawk.tonecommander.com> Hello All On CentOS 5.3, I am trying to build Freeswitch in a different directory and use the -srcdir= option. One reason I want to do this to have Debug and Release build targets from the same source. It doesn't work, the configure errors when it gets to the first library subdirectory lib/srtp and tries to configure in there. The steps I am doing are: 1. Building as root 2. Unzip freeswitch-1.0.4-tar.gz in /opt 3. cd into /opt/freeswitch-1.0.4 4. mkdir Debug 5. cd Debug 6. ../configure -srcdir=".." CFLAGS="-g -ggdb -O2" 7. After several seconds of configuring I get: === configuring in libs/srtp (/opt/freeswitch-1.0.4/Debug/libs/srtp) configure: running /bin/sh ../../../libs/srtp/configure.gnu --disable-option-checking '--prefix=/usr/local/freeswitch' 'CFLAGS=-g -ggdb -O2' --cache-file=/dev/null --srcdir=../../../libs/srtp ../../../libs/srtp/configure.gnu: line 2: ./configure: No such file or directory configure: error: ../../../libs/srtp/configure.gnu failed for libs/srtp [root at roberth-c53 Debug]# The file that's executing is this: [root at roberth-c53 srtp]# cd libs/srtp; cat ../../../libs/srtp/configure.gnu #! /bin/sh ./configure "$@" --disable-shared --with-pic Please tell me if I understood the -srcdir option correctly and if there is a way to do build in a different directory. Thanks, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/354433a3/attachment-0002.html From lists at tigertech.com Fri Nov 13 16:23:19 2009 From: lists at tigertech.com (Robert L Mathews) Date: Fri, 13 Nov 2009 16:23:19 -0800 Subject: [Freeswitch-users] Using play_and_get_digits (or IVR) without a final delay Message-ID: <4AFDF877.5040007@tigertech.com> Hi, I'm a new FreeSWITCH convert from asterisk. It's great software; thanks for making it. I'm trying to play a sound file while listening for possible digits dialed (although in most cases callers will not be dialing anything). If callers do start dialing an extension while the sound file is playing, I want them to be able to dial it slowly without any problems. So the inter-digit timeout should be, say, 2 seconds. However, if people don't start dialing anything while the sound file is playing, I don't want any delay at the end of it. I've tried play_and_get_digits with a 2000 timeout -- but that causes a 2 second "dead air" pause at the end of the sound file if callers don't dial anything. I've also tried using a trivial IVR menu to simulate this, but it has the same problem. Interestingly, it doesn't *look* like IVRs should have the problem, because they allow both "inter-digit-timeout" and "timeout" to be specified separately -- but at least in FreeSWITCH 1.0.4, a short "timeout" value always interrupts the dialing of digits, even if "inter-digit-timeout" is much longer. Is there any way to play a sound file from the dial plan with a long inter-digit delay, but without any final delay if no digits are dialed? Thanks for your time! -- Robert L Mathews, Tiger Technologies From brian at freeswitch.org Fri Nov 13 16:56:38 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Nov 2009 18:56:38 -0600 Subject: [Freeswitch-users] Freeswitch configure error using --srcdir option In-Reply-To: <3894885261AF440BB1025F7488B92E97@greyhawk.tonecommander.com> References: <3894885261AF440BB1025F7488B92E97@greyhawk.tonecommander.com> Message-ID: Don't use --srcdir we don't fully support that and the howto guides do not mention it AT ALL. So doing things that are not in the howto aren't really tested nor supported. /b On Nov 13, 2009, at 11:26 AM, Robert Hadley wrote: > Hello All > > On CentOS 5.3, I am trying to build Freeswitch in a different > directory and use the ?srcdir= option. One reason I want to do this > to have Debug and Release build targets from the same source. > > It doesn?t work, the configure errors when it gets to the first > library subdirectory lib/srtp and tries to configure in there. > > The steps I am doing are: > Building as root > Unzip freeswitch-1.0.4-tar.gz in /opt > cd into /opt/freeswitch-1.0.4 > mkdir Debug > cd Debug > ../configure ?srcdir=?..? CFLAGS=?-g ?ggdb ?O2? > After several seconds of configuring I get: > === configuring in libs/srtp (/opt/freeswitch-1.0.4/Debug/libs/srtp) > configure: running /bin/sh ../../../libs/srtp/configure.gnu -- > disable-option-checking '--prefix=/usr/local/freeswitch' 'CFLAGS=-g > -ggdb -O2' --cache-file=/dev/null --srcdir=../../../libs/srtp > ../../../libs/srtp/configure.gnu: line 2: ./configure: No such file > or directory > configure: error: ../../../libs/srtp/configure.gnu failed for libs/ > srtp > [root at roberth-c53 Debug]# > > The file that?s executing is this: > [root at roberth-c53 srtp]# cd libs/srtp; cat ../../../libs/srtp/ > configure.gnu > #! /bin/sh > ./configure "$@" --disable-shared --with-pic > > > Please tell me if I understood the ?srcdir option correctly and if > there is a way to do build in a different directory. > > Thanks, > Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/dd2c9a64/attachment-0002.html From mike at jerris.com Fri Nov 13 18:12:36 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 13 Nov 2009 21:12:36 -0500 Subject: [Freeswitch-users] Freeswitch configure error using --srcdir option In-Reply-To: References: <3894885261AF440BB1025F7488B92E97@greyhawk.tonecommander.com> Message-ID: <375DC77F-138C-49C7-8CD0-A8A05C485588@jerris.com> Patches to make this work would be gladly accepted. Mike On Nov 13, 2009, at 7:56 PM, Brian West wrote: > Don't use --srcdir we don't fully support that and the howto guides do not mention it AT ALL. So doing things that are not in the howto aren't really tested nor supported. > > /b > > On Nov 13, 2009, at 11:26 AM, Robert Hadley wrote: > >> Hello All >> >> On CentOS 5.3, I am trying to build Freeswitch in a different directory and use the ?srcdir= option. One reason I want to do this to have Debug and Release build targets from the same source. >> >> It doesn?t work, the configure errors when it gets to the first library subdirectory lib/srtp and tries to configure in there. >> >> The steps I am doing are: >> Building as root >> Unzip freeswitch-1.0.4-tar.gz in /opt >> cd into /opt/freeswitch-1.0.4 >> mkdir Debug >> cd Debug >> ../configure ?srcdir=?..? CFLAGS=?-g ?ggdb ?O2? >> After several seconds of configuring I get: >> === configuring in libs/srtp (/opt/freeswitch-1.0.4/Debug/libs/srtp) >> configure: running /bin/sh ../../../libs/srtp/configure.gnu --disable-option-checking '--prefix=/usr/local/freeswitch' 'CFLAGS=-g -ggdb -O2' --cache-file=/dev/null --srcdir=../../../libs/srtp >> ../../../libs/srtp/configure.gnu: line 2: ./configure: No such file or directory >> configure: error: ../../../libs/srtp/configure.gnu failed for libs/srtp >> [root at roberth-c53 Debug]# >> >> The file that?s executing is this: >> [root at roberth-c53 srtp]# cd libs/srtp; cat ../../../libs/srtp/configure.gnu >> #! /bin/sh >> ./configure "$@" --disable-shared --with-pic >> >> >> Please tell me if I understood the ?srcdir option correctly and if there is a way to do build in a different directory. >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091113/a3e11532/attachment-0002.html From andrew at hijacked.us Fri Nov 13 22:48:40 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Sat, 14 Nov 2009 01:48:40 -0500 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> Message-ID: <20091114064840.GC21765@hijacked.us> On Fri, Nov 13, 2009 at 11:28:55PM +0100, Leon de Rooij wrote: > Hi, > > You can use mod_xml_curl (generate xml on a webserver): > > http://wiki.freeswitch.org/wiki/Mod_xml_curl > > or mod_xml_odbc (generate xml in freeswitch): > > http://wiki.freeswitch.org/wiki/Mod_xml_odbc > > or LUA together with luasql (generate xml in freeswitch): > > http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration > Or, if you're really crazy, the erlang module can do it too (even dynamically): http://wiki.freeswitch.org/wiki/Mod_erlang_event#XML_search_bindings :P Andrew From woodydickson at gmail.com Sat Nov 14 00:31:12 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sat, 14 Nov 2009 16:31:12 +0800 Subject: [Freeswitch-users] problem with mod_xml_odbc Message-ID: Hi, I am having problem trying to use mod_xml_odbc using freeswitch-1.0.5pre. Here is the error I am getting: 2009-11-15 00:17:23.571293 [INFO] mod_xml_odbc.c:647 XML ODBC module loading... 2009-11-15 00:17:23.571354 [NOTICE] mod_xml_odbc.c:563 Binding XML Search Function [directory] 2009-11-15 00:17:23.572299 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-11-15 00:17:23.572361 [CRIT] mod_xml_odbc.c:617 Cannot Open ODBC Database! 2009-11-15 00:17:23.572397 [ERR] mod_xml_odbc.c:650 Unable to load xml_odbc config file 2009-11-15 00:17:23.572424 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_xml_odbc.so **Module load routine returned an error** In my config, I have: [root at localhost autoload_configs]# isql myodbc -v +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> How can I fix this problem? thanks, woody -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/c4533556/attachment-0002.html From dome at tel.co.th Sat Nov 14 01:42:32 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 14 Nov 2009 16:42:32 +0700 Subject: [Freeswitch-users] FreeSWITCH Now Supports Broadvoice BV16, BV32 Voice Codecs Message-ID: <8ccbff060911140142j5fa12113l5a63084200719a0@mail.gmail.com> http://freeswitch.org/node/217 Very fast develop :) one reason why i love FS. another is good performance Dome C. From brian at freeswitch.org Sat Nov 14 03:14:29 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Nov 2009 05:14:29 -0600 Subject: [Freeswitch-users] FreeSWITCH Now Supports Broadvoice BV16, BV32 Voice Codecs In-Reply-To: <8ccbff060911140142j5fa12113l5a63084200719a0@mail.gmail.com> References: <8ccbff060911140142j5fa12113l5a63084200719a0@mail.gmail.com> Message-ID: I'm working out one final detail with Broadcom about on wire bitpacking so you might not be compatible with any device that uses these codecs just yet due to confusion in the RFC and the API. Seems the G.192 bitpacking might have been used on the wire for the devices adding un-needed overhead and causing it to not officially be compatible with the RFC. FreeSWITCH can talk to FreeSWITCH without a problem. /b On Nov 14, 2009, at 3:42 AM, Dome Charoenyost wrote: > http://freeswitch.org/node/217 > > Very fast develop :) > one reason why i love FS. another is good performance > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Sat Nov 14 03:15:50 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Nov 2009 05:15:50 -0600 Subject: [Freeswitch-users] problem with mod_xml_odbc In-Reply-To: References: Message-ID: <2CF76C78-1550-421E-96D2-9B3383E674E5@freeswitch.org> Try using isql to connect to the db... /b On Nov 14, 2009, at 2:31 AM, Woody Dickson wrote: > Hi, > > I am having problem trying to use mod_xml_odbc using > freeswitch-1.0.5pre. From viper at fx-services.com Sat Nov 14 05:31:20 2009 From: viper at fx-services.com (Robin Vleij) Date: Sat, 14 Nov 2009 14:31:20 +0100 Subject: [Freeswitch-users] Large number of destinations In-Reply-To: References: <4AFC5E81.9020104@fx-services.com> <4AFC8D01.9060401@fx-services.com> <4AFD3389.6090409@fx-services.com> Message-ID: <4AFEB128.8090606@fx-services.com> On 11/13/09 6:58 PM, Eliot Gable wrote: Hi Eliot! > Performance is not an issue. I clocked 300 calls per second on such a > setup using a Dell R710 with two XEON X5570s and 32 GB RAM as the > FreeSWITCH server and a Dell 2950 4-core system with 8 GB RAM as the > app server. The app server was at 15% - 20% idle at that rate and the Haha, that should do the trick for me. Sounds realy good. I still prefer to use the internal ENUM system for all lookups, but if that's not possible then the mod_xml_curl method is the one for me I think. That takes all routing complexity away from the FS system. > meant that I could only pull 160 calls per second, even though the app > server was 55% - 60% idle at that rate, because the ODBC call took > 1/160th of a second to complete and all the requests were serialized. OK, that's a problem. On the other hand 160 cps is still very good. If I keep an eye on where I use the limit app, it'll be OK. > are storing / querying your data from your database system. Actually, > I suppose you could just be a bad programmer and end up making it > exponential, but I'm assuming you know how to write code and design > your database in a way that avoids that. There's people in my department that know a few things about that, so I can always let my design get approved by them. :) > I have been considering writing a hash cache for the XML dialplan so > that lookups can become constant time, but I have no idea when or if I > will find the time to do that. :) I know the feeling, working on three things at the same time. But it sounds really good all of this. Have to start thinking about the app server design, see how we can do that. It's always a balance, since in the beginning it will be a very small nr of routes. Having the xml_curl setup is a lot of overkill there. But I also know that when connected systems increases there's no time anymore to do it properly, so it's probably best to build the rocketship right from the start. :) /robin From paul.thirumalai at gmail.com Sat Nov 14 17:11:31 2009 From: paul.thirumalai at gmail.com (Paul Thirumalai) Date: Sat, 14 Nov 2009 17:11:31 -0800 Subject: [Freeswitch-users] Question about channel creation messages in logfile Message-ID: <900c9adf0911141711h11e0e251r748034dfa92959f9@mail.gmail.com> Hello All I am a freeswitch newbie. I have managed to get internal phones setup on different computers at home, using X-lite My issue is with making outbound calls. I'm trying to use voicepulse for that. My Freeswitch server IP is 11.111.123.23 My issue is as follows. I used extension 1000 (X-Lite softphone) to make an out going call my cell phone (555-123-1234) >From the freeswitch.log logfile I see that the channel for 1000 goes from CS_NEW-> CS_INIT->CS_ROUTING . At this point it starts going through the dial plans and finds the correct dialplan. According to the dialplan the outgoing call needs to go to sofia/gateway/voicepulse/5551231234 and I see the following lines in the logfile EXECUTE sofia/internal/1000 at 11.111.123.23hash(insert/11.111.123.23-last_dial/global/115be3f6-d01c-11de-8360-976b377ef920) EXECUTE sofia/internal/1000 at 11.111.123.23bridge(sofia/gateway/voicepulse/5551231234) * *This message makes sense to me and appears to be right. Freeswitch is trying to bridge a call from extension # 1000 to outbound number 5551231234 Now the very next line I see freeswitch attempting to create a channel to the outbound number. 2009-11-13 01:16:23.654519 [NOTICE] switch_channel.c:602 New Channel sofia/external/5551231234 [115c7c76-d01c-11de-8360-976b377ef920] Please correct me if I'm mistaken, but isnt freeswitch supposed to create a channel for sofia/gateway/voicepulse/5551231234 and not sjofia/external/5551231235 In any case the channel to sofia/external/5551231234 changes state from CS_NEW->CS_INIT->CS_ROUTING->CS_CONSUME_MEDIA . At this point the freeswitch gets an Remote SDP, I am assuming from voicepulse. In the end I see message which reads 2009-11-13 01:16:30.880417 [DEBUG] mod_sofia.c:306 sofia/external/5035440933 Overriding SIP cause 503 with 500 from the other leg Can someone please tell me what this error means. SIP error code 500 is what I get in X-Lite also. Also could someone please explain what Pre-Answer is. Thanks Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/3b403cfb/attachment-0002.html From abeka at greatiam.com Sat Nov 14 17:11:52 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Sun, 15 Nov 2009 01:11:52 +0000 Subject: [Freeswitch-users] Registration Error 408 Message-ID: <4AFF5558.3080408@greatiam.com> Hello Please pardon me if the solution to this is somewhere already that I have been unable to locate. I have just got a straight out-of-the-box build of FS. According to the wiki, I should be able to test using user IDs 1001 and 1002. However, I am get the above error. If I, however, un-tick register with domain I do net get the error but does not communicate either. Is there a conf that I should have done ? Thanks in advance. Abeka From abeka at greatiam.com Sat Nov 14 17:18:57 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Sun, 15 Nov 2009 01:18:57 +0000 Subject: [Freeswitch-users] Registration Error - 408 timeout Message-ID: <4AFF5701.8010508@greatiam.com> Hello Please pardon me if the solution to this is somewhere already that I have been unable to locate. I have just got a straight out-of-the-box build of FS. According to the wiki, I should be able to test using user IDs 1001 and 1002. However, I am get the above error. If I, however, un-tick register with domain I do net get the error but does not communicate either. Is there a conf that I should have done ? I am using X-lite3 Thanks in advance. Abeka From brian at freeswitch.org Sat Nov 14 17:41:48 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Nov 2009 19:41:48 -0600 Subject: [Freeswitch-users] Question about channel creation messages in logfile In-Reply-To: <900c9adf0911141711h11e0e251r748034dfa92959f9@mail.gmail.com> References: <900c9adf0911141711h11e0e251r748034dfa92959f9@mail.gmail.com> Message-ID: <635CA8D5-7AB8-470E-AA9D-D98A21A049BB@freeswitch.org> Paul, The channel name is useless it means absolutely NOTHING in FreeSWITCH... what matters in FreeSWITCH is the uuid. You can even set the channel names to what ever you like with the include app in dptools. set_name,Name the channel,,mod_dptools So to answer your question. No it shouldn't have the gateway name in the channel name... As for the 500 error your provider responded 500 Internal Server Error. Pre-Answer is usually early media. /b On Nov 14, 2009, at 7:11 PM, Paul Thirumalai wrote: > Hello All > I am a freeswitch newbie. I have managed to get internal phones > setup on different computers at home, using X-lite > My issue is with making outbound calls. I'm trying to use voicepulse > for that. My Freeswitch server IP is 11.111.123.23 > > My issue is as follows. I used extension 1000 (X-Lite softphone) to > make an out going call my cell phone (555-123-1234) > From the freeswitch.log logfile I see that the channel for 1000 goes > from CS_NEW-> CS_INIT->CS_ROUTING . At this point it starts going > through the dial plans and finds the correct dialplan. According to > the dialplan the outgoing call needs to go to sofia/gateway/ > voicepulse/5551231234 and I see the following lines in the logfile > EXECUTE sofia/internal/1000 at 11.111.123.23 hash(insert/11.111.123.23- > last_dial/global/115be3f6-d01c-11de-8360-976b377ef920) > EXECUTE sofia/internal/1000 at 11.111.123.23 bridge(sofia/gateway/ > voicepulse/5551231234) > > > This message makes sense to me and appears to be right. Freeswitch > is trying to bridge a call from extension # 1000 to outbound number > 5551231234 > > Now the very next line I see freeswitch attempting to create a > channel to the outbound number. > > 2009-11-13 01:16:23.654519 [NOTICE] switch_channel.c:602 New Channel > sofia/external/5551231234 [115c7c76-d01c-11de-8360-976b377ef920] > > Please correct me if I'm mistaken, but isnt freeswitch supposed to > create a channel for sofia/gateway/voicepulse/5551231234 and not > sjofia/external/5551231235 > > > In any case the channel to sofia/external/5551231234 changes state > from CS_NEW->CS_INIT->CS_ROUTING->CS_CONSUME_MEDIA . At this point > the freeswitch gets an Remote SDP, I am assuming from voicepulse. > > In the end I see message which reads > 2009-11-13 01:16:30.880417 [DEBUG] mod_sofia.c:306 sofia/external/ > 5035440933 Overriding SIP cause 503 with 500 from the other leg > > Can someone please tell me what this error means. SIP error code 500 > is what I get in X-Lite also. > > Also could someone please explain what Pre-Answer is. > > Thanks > Paul > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/527b3e02/attachment-0002.html From brian at freeswitch.org Sat Nov 14 17:54:33 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Nov 2009 19:54:33 -0600 Subject: [Freeswitch-users] Registration Error 408 In-Reply-To: <4AFF5558.3080408@greatiam.com> References: <4AFF5558.3080408@greatiam.com> Message-ID: <34223AC5-699B-499B-A3B9-CED0F9CF1C59@freeswitch.org> I'm going to venture to guess you're doing this all on the same machine? /b On Nov 14, 2009, at 7:11 PM, Samuel Abekah-Mensah wrote: > Hello > > Please pardon me if the solution to this is somewhere already that I > have been unable to locate. I have just got a straight out-of-the-box > build of FS. According to the wiki, I should be able to test using > user > IDs 1001 and 1002. However, I am get the above error. If I, however, > un-tick register with domain I do net get the error but does not > communicate either. Is there a conf that I should have done ? > > Thanks in advance. > > Abeka > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From paul.thirumalai at gmail.com Sat Nov 14 18:38:06 2009 From: paul.thirumalai at gmail.com (Paul Thirumalai) Date: Sat, 14 Nov 2009 18:38:06 -0800 Subject: [Freeswitch-users] (no subject) In-Reply-To: <900c9adf0911141836r690909w84a0c281f9c0610c@mail.gmail.com> References: <900c9adf0911141836r690909w84a0c281f9c0610c@mail.gmail.com> Message-ID: <900c9adf0911141838o3472a8edv2da66c399c36a6e4@mail.gmail.com> Hi Brian, Thanks for your clarification. By uuid do you mean uuid of the channel? Can you tell me how I could determine the uuid of the channel Thanks Paul On Sat, Nov 14, 2009 at 6:36 PM, Paul Thirumalai wrote: > Paul, > The channel name is useless it means absolutely NOTHING in > FreeSWITCH... what matters in FreeSWITCH is the uuid. You can even > set the channel names to what ever you like with the include app in > > dptools. > > set_name,Name the channel,,mod_dptools > > So to answer your question. No it shouldn't have the gateway name in > the channel name... As for the 500 error your provider responded 500 > > Internal Server Error. > > Pre-Answer is usually early media. > > /b > > On Nov 14, 2009, at 7:11 PM, Paul Thirumalai wrote: > > >* Hello All > *>* I am a freeswitch newbie. I have managed to get internal phones > *>* setup on different computers at home, using X-lite > *>* My issue is with making outbound calls. I'm trying to use voicepulse > *>* for that. My Freeswitch server IP is 11.111.123.23 > *>* > *>* My issue is as follows. I used extension 1000 (X-Lite softphone) to > *>* make an out going call my cell phone (555-123-1234) > *>* From the freeswitch.log logfile I see that the channel for 1000 goes > *>* from CS_NEW-> CS_INIT->CS_ROUTING . At this point it starts going > *>* through the dial plans and finds the correct dialplan. According to > *>* the dialplan the outgoing call needs to go to sofia/gateway/ > *>* voicepulse/5551231234 and I see the following lines in the logfile > *>* EXECUTE sofia/internal/1000 at 11.111.123.23 hash(insert/11.111.123.23- > *>* last_dial/global/115be3f6-d01c-11de-8360-976b377ef920) > *>* EXECUTE sofia/internal/1000 at 11.111.123.23 bridge(sofia/gateway/ > *>* voicepulse/5551231234) > *>* > *>* > *>* This message makes sense to me and appears to be right. Freeswitch > *>* is trying to bridge a call from extension # 1000 to outbound number > *>* 5551231234 > *>* > *>* Now the very next line I see freeswitch attempting to create a > *>* channel to the outbound number. > *>* > *>* 2009-11-13 01:16:23.654519 [NOTICE] switch_channel.c:602 New Channel > *>* sofia/external/5551231234 [115c7c76-d01c-11de-8360-976b377ef920] > *>* > *>* Please correct me if I'm mistaken, but isnt freeswitch supposed to > *>* create a channel for sofia/gateway/voicepulse/5551231234 and not > *>* sjofia/external/5551231235 > *>* > *>* > *>* In any case the channel to sofia/external/5551231234 changes state > *>* from CS_NEW->CS_INIT->CS_ROUTING->CS_CONSUME_MEDIA . At this point > *>* the freeswitch gets an Remote SDP, I am assuming from voicepulse. > *>* > *>* In the end I see message which reads > *>* 2009-11-13 01:16:30.880417 [DEBUG] mod_sofia.c:306 sofia/external/ > *>* 5035440933 Overriding SIP cause 503 with 500 from the other leg > *>* > *>* Can someone please tell me what this error means. SIP error code 500 > *>* is what I get in X-Lite also. > *>* > *>* Also could someone please explain what Pre-Answer is. > *>* > *>* Thanks > *>* Paul > *>* > *>* _______________________________________________ > *>* FreeSWITCH-users mailing list > *>* FreeSWITCH-users at lists.freeswitch.org > *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > *>* users > *>* http://www.freeswitch.org > * > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/14b6f8c5/attachment-0002.html From tina at a2unlimited.com Sat Nov 14 18:41:13 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Sat, 14 Nov 2009 21:41:13 -0500 Subject: [Freeswitch-users] fs_cli Error Message-ID: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> I'm trying to setup fs_cli on a server that is not running the FreeSWITCH server, and I keep getting the following error: "error while loading shared libraries: libedit.so.0: cannot open shared object file: No such file or directory" When I go to one of my FreeSWITCH servers, where fs_cli is working fine, I cannot find the existence of libedit.so.0 anywhere on the server, so I'm not sure what I'm missing... Any thoughts? - Tina From jason at jasonjgw.net Sat Nov 14 18:53:18 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 15 Nov 2009 13:53:18 +1100 Subject: [Freeswitch-users] fs_cli Error In-Reply-To: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> References: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> Message-ID: <20091115025318.GA2052@jdc.jasonjgw.net> tina at a2unlimited.com wrote: > I'm trying to setup fs_cli on a server that is not running the FreeSWITCH > server, and I keep getting the following error: > > "error while loading shared libraries: libedit.so.0: cannot open shared > object file: No such file or directory" For me, under Debian, it's in the libedit2 package. However, fs_cli isn't looking for it, at least not directly. From william.suffill at gmail.com Sat Nov 14 18:57:14 2009 From: william.suffill at gmail.com (William Suffill) Date: Sat, 14 Nov 2009 21:57:14 -0500 Subject: [Freeswitch-users] fs_cli Error In-Reply-To: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> References: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> Message-ID: <6b65470d0911141857h1ab0696fie35a32bf83e265a5@mail.gmail.com> Libedit shared library isn't on the box and fs_cli needs it. It's included as part of the FreeSWITCH build process so any boxes you have FreeSWITCH installed on would have it. You should be able to install libedit on the box you want to use fs_cli on to fix this. -- W From brian at freeswitch.org Sat Nov 14 18:58:16 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Nov 2009 20:58:16 -0600 Subject: [Freeswitch-users] (no subject) In-Reply-To: <900c9adf0911141838o3472a8edv2da66c399c36a6e4@mail.gmail.com> References: <900c9adf0911141836r690909w84a0c281f9c0610c@mail.gmail.com> <900c9adf0911141838o3472a8edv2da66c399c36a6e4@mail.gmail.com> Message-ID: <005E2F47-F1F8-4931-AB5D-1CBBC30590A3@freeswitch.org> show channels its the first item listed. /b On Nov 14, 2009, at 8:38 PM, Paul Thirumalai wrote: > Hi Brian, > Thanks for your clarification. By uuid do you mean uuid of the > channel? > > Can you tell me how I could determine the uuid of the channel > > Thanks > Paul > > On Sat, Nov 14, 2009 at 6:36 PM, Paul Thirumalai > wrote: > Paul, > The channel name is useless it means absolutely NOTHING in > FreeSWITCH... what matters in FreeSWITCH is the uuid. You can even > > set the channel names to what ever you like with the include app in > > dptools. > > set_name,Name the channel,,mod_dptools > > So to answer your question. No it shouldn't have the gateway name in > the channel name... As for the 500 error your provider responded 500 > > > Internal Server Error. > > Pre-Answer is usually early media. > > /b > > On Nov 14, 2009, at 7:11 PM, Paul Thirumalai wrote: > > > Hello All > > I am a freeswitch newbie. I have managed to get internal phones > > > > setup on different computers at home, using X-lite > > My issue is with making outbound calls. I'm trying to use voicepulse > > for that. My Freeswitch server IP is 11.111.123.23 > > > > > > My issue is as follows. I used extension 1000 (X-Lite softphone) to > > make an out going call my cell phone (555-123-1234) > > From the freeswitch.log logfile I see that the channel for 1000 goes > > > > from CS_NEW-> CS_INIT->CS_ROUTING . At this point it starts going > > through the dial plans and finds the correct dialplan. According to > > the dialplan the outgoing call needs to go to sofia/gateway/ > > > > voicepulse/5551231234 and I see the following lines in the logfile > > EXECUTE sofia/internal/1000 at 11.111.123.23 hash(insert/ > 11.111.123.23- > > > > last_dial/global/115be3f6-d01c-11de-8360-976b377ef920) > > EXECUTE sofia/internal/1000 at 11.111.123.23 bridge(sofia/gateway/ > > > > voicepulse/5551231234) > > > > > > This message makes sense to me and appears to be right. Freeswitch > > is trying to bridge a call from extension # 1000 to outbound number > > > > 5551231234 > > > > Now the very next line I see freeswitch attempting to create a > > channel to the outbound number. > > > > 2009-11-13 01:16:23.654519 [NOTICE] switch_channel.c:602 New Channel > > > > sofia/external/5551231234 [115c7c76-d01c-11de-8360-976b377ef920] > > > > Please correct me if I'm mistaken, but isnt freeswitch supposed to > > create a channel for sofia/gateway/voicepulse/5551231234 and not > > > > sjofia/external/5551231235 > > > > > > In any case the channel to sofia/external/5551231234 changes state > > from CS_NEW->CS_INIT->CS_ROUTING->CS_CONSUME_MEDIA . At this point > > > > the freeswitch gets an Remote SDP, I am assuming from voicepulse. > > > > In the end I see message which reads > > 2009-11-13 01:16:30.880417 [DEBUG] mod_sofia.c:306 sofia/external/ > > > > 5035440933 Overriding SIP cause 503 with 500 from the other leg > > > > Can someone please tell me what this error means. SIP error code 500 > > is what I get in X-Lite also. > > > > > > Also could someone please explain what Pre-Answer is. > > > > Thanks > > Paul > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/af361aae/attachment-0002.html From anthony.minessale at gmail.com Sat Nov 14 20:16:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Nov 2009 22:16:53 -0600 Subject: [Freeswitch-users] fs_cli Error In-Reply-To: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> References: <3ee7d77cee7a65157cfcf78bd25ade8f.squirrel@emailmg.ipower.com> Message-ID: <191c3a030911142016o6d2d01e9vcf0f3512a7c0b49@mail.gmail.com> from build root: cd libs/esl make the resulting fs_cli that is in that dir one you type make should be more portable than the one created by the top level make in FS On Sat, Nov 14, 2009 at 8:41 PM, wrote: > I'm trying to setup fs_cli on a server that is not running the FreeSWITCH > server, and I keep getting the following error: > > "error while loading shared libraries: libedit.so.0: cannot open shared > object file: No such file or directory" > > When I go to one of my FreeSWITCH servers, where fs_cli is working fine, I > cannot find the existence of libedit.so.0 anywhere on the server, so I'm > not sure what I'm missing... > > Any thoughts? > > - Tina > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091114/155749cc/attachment-0002.html From samuelmukoti at gmail.com Sun Nov 15 08:39:18 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Sun, 15 Nov 2009 18:39:18 +0200 Subject: [Freeswitch-users] FS mod_SQL Message-ID: <2584B7AF-4F61-483A-86C6-A9A1961E8EA8@gmail.com> Hi, I'm a newbie to FS, and I wanted to implement a setup where I provision the sip endpoints though a SQL database like mysql and also manage call routing too? Is this possible since I understand FS uses XML config files. Best regards Sam From jmesquita at freeswitch.org Sun Nov 15 10:10:05 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 15 Nov 2009 15:10:05 -0300 Subject: [Freeswitch-users] FS mod_SQL In-Reply-To: <2584B7AF-4F61-483A-86C6-A9A1961E8EA8@gmail.com> References: <2584B7AF-4F61-483A-86C6-A9A1961E8EA8@gmail.com> Message-ID: This is the final answer: http://wiki.freeswitch.org/wiki/Mod_xml_curl JM On Sun, Nov 15, 2009 at 1:39 PM, Samuel Mukoti wrote: > Hi, > > I'm a newbie to FS, and I wanted to implement a setup where I > provision the sip endpoints though a SQL database like mysql and also > manage call routing too? Is this possible since I understand FS uses > XML config files. > > Best regards > > Sam > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091115/9777fe2e/attachment-0002.html From mike at jerris.com Sun Nov 15 10:20:51 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Nov 2009 13:20:51 -0500 Subject: [Freeswitch-users] FS mod_SQL In-Reply-To: <2584B7AF-4F61-483A-86C6-A9A1961E8EA8@gmail.com> References: <2584B7AF-4F61-483A-86C6-A9A1961E8EA8@gmail.com> Message-ID: <632798A4-9812-42D5-AA73-F4DDE912954F@jerris.com> http://wiki.freeswitch.org/wiki/Mod_xml_curl On Nov 15, 2009, at 11:39 AM, Samuel Mukoti wrote: > Hi, > > I'm a newbie to FS, and I wanted to implement a setup where I > provision the sip endpoints though a SQL database like mysql and also > manage call routing too? Is this possible since I understand FS uses > XML config files. > From vedamaker at netscape.net Sun Nov 15 09:42:16 2009 From: vedamaker at netscape.net (vedamaker at netscape.net) Date: Sun, 15 Nov 2009 12:42:16 -0500 Subject: [Freeswitch-users] Problem with Siemens A580 IP Phones Message-ID: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> I am FS beginner and I have a basic PBX setup using FS with the Siemens A580 IP Phones. I thought everything was working fine since I could make and receive basic calls without any obvious issues. However, recently I wanted to use more advanced functions in FS and discovered that I could not use any of DTMF based functions (e.g. call transfer/record) during calls with the Siemens IP phones. The same functions work fine when I use a softphone. So, I started looking at the log file and I think there is some problem between the Siemens IP phones and FS (log file attached below). It seems that when a call comes in, FS calls the extensions and then the extensions send back confirmation and SIP status codes. With softphone extensions, I see 180 (Ringing) and 200 (OK) as normal status. However, with Siemens IP phone extensions, I see 480 (Temporarily Unavailable) which seems to cause FS to terminate the session. So, FS log shows there is actually no active session which explains why it does not performs DTMF detection for the call session. However, the call to Siemens IP phones actually continues with ringing when an extension handset answers the call is established with the caller with full voice communication. I don't know how FS works but this seems very strange. I would like to know how to get FS to work properly with Siemens IP phones including the DTMF functions during calls. Any help would be appreciated. ---------------------------------------------- 2009-11-14 09:35:43.942450 [NOTICE] switch_channel.c:602 New Channel sofia/internal/4155559999 at 192.168.1.254 [22f8ee00-d144-11de-a41f-e5a6b5425f55] 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_NEW 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:404 (sofia/internal/4155559999 at 192.168.1.254) State NEW 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3289 Channel sofia/internal/4155559999 at 192.168.1.254 entering state [received][100] 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3296 Remote SDP: v=0 o=- 119640485 119640485 IN IP4 192.168.1.97 s=- c=IN IP4 192.168.1.97 t=0 0 m=audio 16430 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20] 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20] 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20] 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:2029 Set Codec sofia/internal/4155559999 at 192.168.1.254 PCMU/8000 20 ms 160 samples 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3455 (sofia/internal/4155559999 at 192.168.1.254) State Change CS_NEW -> CS_INIT 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_INIT 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/4155559999 at 192.168.1.254) State INIT 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:83 sofia/internal/4155559999 at 192.168.1.254 SOFIA INIT 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:111 (sofia/internal/4155559999 at 192.168.1.254) State Change CS_INIT -> CS_ROUTING 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/4155559999 at 192.168.1.254) State INIT going to sleep 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_ROUTING 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/4155559999 at 192.168.1.254) State ROUTING 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:130 sofia/internal/4155559999 at 192.168.1.254 SOFIA ROUTING 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:78 sofia/internal/4155559999 at 192.168.1.254 Standard ROUTING 2009-11-14 09:35:43.951943 [INFO] mod_dialplan_xml.c:315 Processing WIRELESS CALLER->4155553333 in context default Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->unloop] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->tod_example] continue=true Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [global-intercept] destination_number(4155553333) =~ /^886$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [group-intercept] destination_number(4155553333) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [intercept-ext] destination_number(4155553333) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->redial] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [redial] destination_number(4155553333) =~ /^870$|^\*66$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->global] continue=true Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/4155559999 at 192.168.1.254 Absolute Condition [global] Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [snom-demo-2] destination_number(4155553333) =~ /^9001$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [snom-demo-1] destination_number(4155553333) =~ /^9000$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [eavesdrop] destination_number(4155553333) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [eavesdrop] destination_number(4155553333) =~ /^779$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->call_return] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [call_return] destination_number(4155553333) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->del-group] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [del-group] destination_number(4155553333) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->add-group] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [add-group] destination_number(4155553333) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [call-group-simo] destination_number(4155553333) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [call-group-order] destination_number(4155553333) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [extension-intercom] destination_number(4155553333) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [Local_Extension] destination_number(4155553333) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->group_dial_ringables] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [group_dial_ringables] destination_number(4155553333) =~ /^1999$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->mobile_extensions] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [mobile_extensions] destination_number(4155553333) =~ /^(20[01][0-9])$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->vmain] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [vmain] destination_number(4155553333) =~ /^vmain$|^4000$$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->vm1000] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [vm1000] destination_number(4155553333) =~ /^vm1000$|^4100$|^\*98$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->sip_uri] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [sip_uri] destination_number(4155553333) =~ /^sip:(.*)$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->nb_conferences] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [nb_conferences] destination_number(4155553333) =~ /^(30\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->wb_conferences] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [wb_conferences] destination_number(4155553333) =~ /^(31\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->uwb_conferences] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [uwb_conferences] destination_number(4155553333) =~ /^(32\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->cdquality_conferences] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [cdquality_conferences] destination_number(4155553333) =~ /^(33\d{2})$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->freeswitch_public_conf_via_sip] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(4155553333) =~ /^9(888|1616|3232)$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [mad_boss_intercom] destination_number(4155553333) =~ /^0911$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [mad_boss_intercom] destination_number(4155553333) =~ /^0912$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->mad_boss] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [mad_boss] destination_number(4155553333) =~ /^0913$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ivr_demo] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ivr_demo] destination_number(4155553333) =~ /^5000$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->dynamic_conference] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [dynamic_conference] destination_number(4155553333) =~ /^5001$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->rtp_multicast_page] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [rtp_multicast_page] destination_number(4155553333) =~ /^pagegroup$|^7243$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] destination_number(4155553333) =~ /^5900$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->unpark] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] destination_number(4155553333) =~ /^5901$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] destination_number(4155553333) =~ /park\+(\d+)/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->unpark] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] destination_number(4155553333) =~ /^parking$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] destination_number(4155553333) =~ /callpark/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->unpark] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] destination_number(4155553333) =~ /pickup/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->wait] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [wait] destination_number(4155553333) =~ /^wait$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->fax_receive] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [fax_receive] destination_number(4155553333) =~ /^9978$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->fax_transmit] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [fax_transmit] destination_number(4155553333) =~ /^9979$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ringback_180] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ringback_180] destination_number(4155553333) =~ /^9980$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ringback_183_uk_ring] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ringback_183_uk_ring] destination_number(4155553333) =~ /^9981$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ringback_183_music_ring] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ringback_183_music_ring] destination_number(4155553333) =~ /^9982$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ringback_post_answer_uk_ring] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(4155553333) =~ /^9983$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ringback_post_answer_music] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ringback_post_answer_music] destination_number(4155553333) =~ /^9984$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->ClueCon] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ClueCon] destination_number(4155553333) =~ /^9991$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->show_info] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [show_info] destination_number(4155553333) =~ /^9992$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->video_record] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [video_record] destination_number(4155553333) =~ /^9993$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->video_playback] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [video_playback] destination_number(4155553333) =~ /^9994$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->delay_echo] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [delay_echo] destination_number(4155553333) =~ /^9995$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->echo] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [echo] destination_number(4155553333) =~ /^9996$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->milliwatt] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [milliwatt] destination_number(4155553333) =~ /^9997$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->tone_stream] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [tone_stream] destination_number(4155553333) =~ /^9998$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->zrtp_enrollement] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [zrtp_enrollement] destination_number(4155553333) =~ /^9787$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->hold_music] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [hold_music] destination_number(4155553333) =~ /^9999$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->fax] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [fax] destination_number(4155553333) =~ /^fax|9777$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->test-9555] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [test-9555] destination_number(4155553333) =~ /^9555$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->test-9666] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [test-9666] destination_number(4155553333) =~ /^9666$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->pizza_demo] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [pizza_demo] destination_number(4155553333) =~ /^(pizza|74992)$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->Inbound-4155553333] continue=false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [Inbound-4155553333] destination_number(4155553333) =~ /^4155553333$/ break=on-false Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action ring_ready() Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action set(ringback=${us-ring}) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action set(call_timeout=28) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action set(continue_on_fail=true) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bridge(${group_call(ringables@${domain_name})}) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action answer() Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action sleep(1000) Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action voicemail(default ${domain_name} 1000) 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/4155559999 at 192.168.1.254) State Change CS_ROUTING -> CS_EXECUTE 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/4155559999 at 192.168.1.254) State ROUTING going to sleep 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_EXECUTE 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/4155559999 at 192.168.1.254) State EXECUTE 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:173 sofia/internal/4155559999 at 192.168.1.254 SOFIA EXECUTE 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:151 sofia/internal/4155559999 at 192.168.1.254 Standard EXECUTE EXECUTE sofia/internal/4155559999 at 192.168.1.254 hash(insert/192.168.1.254-spymap/4155559999/22f8ee00-d144-11de-a41f-e5a6b5425f55) EXECUTE sofia/internal/4155559999 at 192.168.1.254 hash(insert/192.168.1.254-last_dial/4155559999/4155553333) EXECUTE sofia/internal/4155559999 at 192.168.1.254 hash(insert/192.168.1.254-last_dial/global/22f8ee00-d144-11de-a41f-e5a6b5425f55) EXECUTE sofia/internal/4155559999 at 192.168.1.254 ring_ready() 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:415 sofia/internal/4155559999 at 192.168.1.254 receive message [RINGING] 2009-11-14 09:35:43.951943 [NOTICE] mod_sofia.c:1449 Ring-Ready sofia/internal/4155559999 at 192.168.1.254! 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:43.951943 [NOTICE] mod_dptools.c:415 Ring Ready sofia/internal/4155559999 at 192.168.1.254! EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(1 b s execute_extension::dx XML features) 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 execute_extension::dx XML features EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/4155559999.2009-11-14-09-35-43.wav) 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/4155559999.2009-11-14-09-35-43.wav EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(3 b s execute_extension::cf XML features) 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 execute_extension::cf XML features EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(ringback=%(2000,4000,440.0,480.0)) 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/4155559999 at 192.168.1.254 SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(transfer_ringback=local_stream://moh) 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/4155559999 at 192.168.1.254 SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(call_timeout=28) 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/4155559999 at 192.168.1.254 SET [call_timeout]=[28] EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(hangup_after_bridge=true) 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/4155559999 at 192.168.1.254 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(continue_on_fail=true) 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/4155559999 at 192.168.1.254 SET [continue_on_fail]=[true] 2009-11-14 09:35:43.966601 [DEBUG] sofia.c:3289 Channel sofia/internal/4155559999 at 192.168.1.254 entering state [early][180] EXECUTE sofia/internal/4155559999 at 192.168.1.254 bridge([presence_id=1011 at 192.168.1.254]sofia/internal/sip:1011 at 192.168.1.98:5872,[presence_id=1012 at 192.168.1.254]sofia/internal/sip:1012 at 192.168.1.98:5872,[presence_id=1014 at 192.168.1.254]sofia/internal/sip:1014 at 192.168.1.97:5060) 2009-11-14 09:35:43.986485 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:1011 at 192.168.1.98:5872 [22ff385a-d144-11de-a41f-e5a6b5425f55] 2009-11-14 09:35:43.986485 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_NEW -> CS_INIT 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.990495 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:1012 at 192.168.1.98:5872 [22ff6230-d144-11de-a41f-e5a6b5425f55] 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_INIT 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1011 at 192.168.1.98:5872) State INIT 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1011 at 192.168.1.98:5872 SOFIA INIT 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_INIT -> CS_ROUTING 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1011 at 192.168.1.98:5872) State INIT going to sleep 2009-11-14 09:35:43.990495 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1011 at 192.168.1.98:5872 entering state [calling][0] 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_ROUTING 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1011 at 192.168.1.98:5872) State ROUTING 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:130 sofia/internal/sip:1011 at 192.168.1.98:5872 SOFIA ROUTING 2009-11-14 09:35:43.990495 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1011 at 192.168.1.98:5872) State ROUTING going to sleep 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_CONSUME_MEDIA 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1011 at 192.168.1.98:5872) State CONSUME_MEDIA 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_NEW -> CS_INIT 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.994449 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:1014 at 192.168.1.97:5060 [22fffdb2-d144-11de-a41f-e5a6b5425f55] 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_INIT 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1012 at 192.168.1.98:5872) State INIT 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1012 at 192.168.1.98:5872 SOFIA INIT 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_INIT -> CS_ROUTING 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.994449 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1012 at 192.168.1.98:5872 entering state [calling][0] 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1012 at 192.168.1.98:5872) State INIT going to sleep 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_ROUTING 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1012 at 192.168.1.98:5872) State ROUTING 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:130 sofia/internal/sip:1012 at 192.168.1.98:5872 SOFIA ROUTING 2009-11-14 09:35:43.994449 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1012 at 192.168.1.98:5872) State ROUTING going to sleep 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_CONSUME_MEDIA 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1012 at 192.168.1.98:5872) State CONSUME_MEDIA 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_NEW -> CS_INIT 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] 2009-11-14 09:35:43.998457 [DEBUG] switch_ivr_originate.c:1701 sofia/internal/4155559999 at 192.168.1.254 receive message [PROGRESS] 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_INIT 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1014 at 192.168.1.97:5060) State INIT 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1014 at 192.168.1.97:5060 SOFIA INIT 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_INIT -> CS_ROUTING 2009-11-14 09:35:43.998457 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] 2009-11-14 09:35:43.998457 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1014 at 192.168.1.97:5060 entering state [calling][0] 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:1014 at 192.168.1.97:5060) State INIT going to sleep 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_ROUTING 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1014 at 192.168.1.97:5060) State ROUTING 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:130 sofia/internal/sip:1014 at 192.168.1.97:5060 SOFIA ROUTING 2009-11-14 09:35:43.998457 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-14 09:35:43.998457 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:1014 at 192.168.1.97:5060) State ROUTING going to sleep 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_CONSUME_MEDIA 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1014 at 192.168.1.97:5060) State CONSUME_MEDIA 2009-11-14 09:35:43.998457 [INFO] switch_ivr_originate.c:1701 Sending early media 2009-11-14 09:35:44.2435 [DEBUG] sofia_glue.c:2263 AUDIO RTP [sofia/internal/4155559999 at 192.168.1.254] 192.168.1.254 port 31052 -> 192.168.1.97 port 16430 codec: 0 ms: 20 2009-11-14 09:35:44.2435 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms 2009-11-14 09:35:44.6432 [INFO] mod_sofia.c:1506 Ring SDP: v=0 o=FreeSWITCH 1258189091 1258189092 IN IP4 192.168.1.254 s=FreeSWITCH c=IN IP4 192.168.1.254 t=0 0 m=audio 31052 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-11-14 09:35:44.6432 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/4155559999 at 192.168.1.254! 2009-11-14 09:35:44.6432 [DEBUG] sofia.c:3289 Channel sofia/internal/4155559999 at 192.168.1.254 entering state [early][183] 2009-11-14 09:35:44.6432 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:44.6432 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2009-11-14 09:35:44.6432 [DEBUG] switch_ivr_originate.c:1777 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-14 09:35:44.18430 [DEBUG] switch_core_io.c:649 sofia/internal/4155559999 at 192.168.1.254 receive message [TRANSCODING_NECESSARY] 2009-11-14 09:35:44.22473 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1014 at 192.168.1.97:5060 entering state [proceeding][180] 2009-11-14 09:35:44.22473 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:1014 at 192.168.1.97:5060! 2009-11-14 09:35:52.326423 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1011 at 192.168.1.98:5872 entering state [terminated][480] 2009-11-14 09:35:52.326423 [NOTICE] sofia.c:3849 Hangup sofia/internal/sip:1011 at 192.168.1.98:5872 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-11-14 09:35:52.326423 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [KILL] 2009-11-14 09:35:52.326423 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1011 at 192.168.1.98:5872) State CONSUME_MEDIA going to sleep 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_HANGUP 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1011 at 192.168.1.98:5872) State HANGUP 2009-11-14 09:35:52.330490 [DEBUG] mod_sofia.c:306 sofia/internal/sip:1011 at 192.168.1.98:5872 Overriding SIP cause 408 with 480 from the other leg 2009-11-14 09:35:52.330490 [DEBUG] mod_sofia.c:338 Channel sofia/internal/sip:1011 at 192.168.1.98:5872 hanging up, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1011 at 192.168.1.98:5872 Standard HANGUP, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1011 at 192.168.1.98:5872) State HANGUP going to sleep 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_HANGUP -> CS_REPORTING 2009-11-14 09:35:52.330490 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_REPORTING 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1011 at 192.168.1.98:5872) State REPORTING 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:1011 at 192.168.1.98:5872 Standard REPORTING, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1011 at 192.168.1.98:5872) State REPORTING going to sleep 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_REPORTING -> CS_DESTROY 2009-11-14 09:35:52.330490 [DEBUG] switch_core_session.c:1068 Session 542 (sofia/internal/sip:1011 at 192.168.1.98:5872) Locked, Waiting on external entities 2009-11-14 09:35:52.618414 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1012 at 192.168.1.98:5872 entering state [terminated][480] 2009-11-14 09:35:52.618414 [NOTICE] sofia.c:3849 Hangup sofia/internal/sip:1012 at 192.168.1.98:5872 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-11-14 09:35:52.618414 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [KILL] 2009-11-14 09:35:52.618414 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1012 at 192.168.1.98:5872) State CONSUME_MEDIA going to sleep 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_HANGUP 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1012 at 192.168.1.98:5872) State HANGUP 2009-11-14 09:35:52.626426 [DEBUG] mod_sofia.c:306 sofia/internal/sip:1012 at 192.168.1.98:5872 Overriding SIP cause 408 with 480 from the other leg 2009-11-14 09:35:52.626426 [DEBUG] mod_sofia.c:338 Channel sofia/internal/sip:1012 at 192.168.1.98:5872 hanging up, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1012 at 192.168.1.98:5872 Standard HANGUP, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1012 at 192.168.1.98:5872) State HANGUP going to sleep 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_HANGUP -> CS_REPORTING 2009-11-14 09:35:52.626426 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_REPORTING 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1012 at 192.168.1.98:5872) State REPORTING 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:1012 at 192.168.1.98:5872 Standard REPORTING, cause: NO_USER_RESPONSE 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1012 at 192.168.1.98:5872) State REPORTING going to sleep 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_REPORTING -> CS_DESTROY 2009-11-14 09:35:52.626426 [DEBUG] switch_core_session.c:1068 Session 543 (sofia/internal/sip:1012 at 192.168.1.98:5872) Locked, Waiting on external entities 2009-11-14 09:35:59.778421 [DEBUG] sofia.c:3289 Channel sofia/internal/4155559999 at 192.168.1.254 entering state [terminated][487] 2009-11-14 09:35:59.778421 [NOTICE] sofia.c:3849 Hangup sofia/internal/4155559999 at 192.168.1.254 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2009-11-14 09:35:59.778421 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/4155559999 at 192.168.1.254 [KILL] 2009-11-14 09:35:59.778421 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:59.798426 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/internal/sip:1014 at 192.168.1.97:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2009-11-14 09:35:59.798426 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [KILL] 2009-11-14 09:35:59.798426 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] 2009-11-14 09:35:59.798426 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1086 Session 542 (sofia/internal/sip:1011 at 192.168.1.98:5872) Ended 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/sip:1011 at 192.168.1.98:5872 [CS_DESTROY] 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1011 at 192.168.1.98:5872) State DESTROY 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:255 sofia/internal/sip:1011 at 192.168.1.98:5872 SOFIA DESTROY 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1011 at 192.168.1.98:5872 Standard DESTROY 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1011 at 192.168.1.98:5872) State DESTROY going to sleep 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1086 Session 543 (sofia/internal/sip:1012 at 192.168.1.98:5872) Ended 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/sip:1012 at 192.168.1.98:5872 [CS_DESTROY] 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1012 at 192.168.1.98:5872) State DESTROY 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:255 sofia/internal/sip:1012 at 192.168.1.98:5872 SOFIA DESTROY 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1012 at 192.168.1.98:5872 Standard DESTROY 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1012 at 192.168.1.98:5872) State DESTROY going to sleep 2009-11-14 09:35:59.798426 [INFO] mod_dptools.c:2093 Originate Failed. Cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/4155559999 at 192.168.1.254) State EXECUTE going to sleep 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_HANGUP 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/4155559999 at 192.168.1.254) State HANGUP 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:306 sofia/internal/4155559999 at 192.168.1.254 Overriding SIP cause 487 with 487 from the other leg 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:338 Channel sofia/internal/4155559999 at 192.168.1.254 hanging up, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:46 sofia/internal/4155559999 at 192.168.1.254 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/4155559999 at 192.168.1.254) State HANGUP going to sleep 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:1014 at 192.168.1.97:5060) State CONSUME_MEDIA going to sleep 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_HANGUP 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1014 at 192.168.1.97:5060) State HANGUP 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:306 sofia/internal/sip:1014 at 192.168.1.97:5060 Overriding SIP cause 487 with 487 from the other leg 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:338 Channel sofia/internal/sip:1014 at 192.168.1.97:5060 hanging up, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:406 Sending CANCEL to sofia/internal/sip:1014 at 192.168.1.97:5060 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1014 at 192.168.1.97:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:1014 at 192.168.1.97:5060) State HANGUP going to sleep 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_HANGUP -> CS_REPORTING 2009-11-14 09:35:59.810601 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_REPORTING 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1014 at 192.168.1.97:5060) State REPORTING 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:1014 at 192.168.1.97:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/sip:1014 at 192.168.1.97:5060) State REPORTING going to sleep 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_REPORTING -> CS_DESTROY 2009-11-14 09:35:59.810601 [DEBUG] switch_core_session.c:1068 Session 544 (sofia/internal/sip:1014 at 192.168.1.97:5060) Locked, Waiting on external entities 2009-11-14 09:35:59.810601 [NOTICE] switch_core_session.c:1086 Session 544 (sofia/internal/sip:1014 at 192.168.1.97:5060) Ended 2009-11-14 09:35:59.810601 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/sip:1014 at 192.168.1.97:5060 [CS_DESTROY] 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1014 at 192.168.1.97:5060) State DESTROY 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:255 sofia/internal/sip:1014 at 192.168.1.97:5060 SOFIA DESTROY 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1014 at 192.168.1.97:5060 Standard DESTROY 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/sip:1014 at 192.168.1.97:5060) State DESTROY going to sleep 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/4155559999 at 192.168.1.254) State Change CS_HANGUP -> CS_REPORTING 2009-11-14 09:35:59.814986 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/4155559999 at 192.168.1.254 [BREAK] 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_REPORTING 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/4155559999 at 192.168.1.254) State REPORTING 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:53 sofia/internal/4155559999 at 192.168.1.254 Standard REPORTING, cause: ORIGINATOR_CANCEL 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/4155559999 at 192.168.1.254) State REPORTING going to sleep 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/4155559999 at 192.168.1.254) State Change CS_REPORTING -> CS_DESTROY 2009-11-14 09:35:59.814986 [DEBUG] switch_core_session.c:1068 Session 541 (sofia/internal/4155559999 at 192.168.1.254) Locked, Waiting on external entities 2009-11-14 09:35:59.814986 [NOTICE] switch_core_session.c:1086 Session 541 (sofia/internal/4155559999 at 192.168.1.254) Ended 2009-11-14 09:35:59.814986 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/4155559999 at 192.168.1.254 [CS_DESTROY] 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/4155559999 at 192.168.1.254) State DESTROY 2009-11-14 09:35:59.814986 [DEBUG] mod_sofia.c:255 sofia/internal/4155559999 at 192.168.1.254 SOFIA DESTROY 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:60 sofia/internal/4155559999 at 192.168.1.254 Standard DESTROY 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/4155559999 at 192.168.1.254) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091115/3f873e0d/attachment-0002.html From abeka at greatiam.com Sun Nov 15 10:53:27 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Sun, 15 Nov 2009 18:53:27 +0000 Subject: [Freeswitch-users] Registration Error 408 In-Reply-To: <34223AC5-699B-499B-A3B9-CED0F9CF1C59@freeswitch.org> References: <4AFF5558.3080408@greatiam.com> <34223AC5-699B-499B-A3B9-CED0F9CF1C59@freeswitch.org> Message-ID: <4B004E27.3040600@greatiam.com> Hi Brian I have FS running on a FC11 box and the clients IDs 1001 and 1002 running off 2 Windows boxes using X-lite3 Thanks Brian West wrote: >
I'm going > to venture to guess you're doing this all on the same machine? > > /b > > On Nov 14, 2009, at 7:11 PM, Samuel Abekah-Mensah wrote: > >> Hello >> >> Please pardon me if the solution to this is somewhere already that I >> have been unable to locate. I have just got a straight out-of-the-box >> build of FS. According to the wiki, I should be able to test using user >> IDs 1001 and 1002. However, I am get the above error. If I, however, >> un-tick register with domain I do net get the error but does not >> communicate either. Is there a conf that I should have done ? >> >> Thanks in advance. >> >> Abeka >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > >
> From mike at jerris.com Sun Nov 15 11:02:37 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Nov 2009 14:02:37 -0500 Subject: [Freeswitch-users] Problem with Siemens A580 IP Phones In-Reply-To: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> References: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> Message-ID: It doesn't look like your call ever gets setup in this trace, if you enable the sip trace you might see a bit more, but it looks like we are receiving a 480 response from the called phone. Mike On Nov 15, 2009, at 12:42 PM, vedamaker at netscape.net wrote: > > I am FS beginner and I have a basic PBX setup using FS with the Siemens A580 IP Phones. I thought everything was working fine since I could make and receive basic calls without any obvious issues. However, recently I wanted to use more advanced functions in FS and discovered that I could not use any of DTMF based functions (e.g. call transfer/record) during calls with the Siemens IP phones. The same functions work fine when I use a softphone. So, I started looking at the log file and I think there is some problem between the Siemens IP phones and FS (log file attached below). It seems that when a call comes in, FS calls the extensions and then the extensions send back confirmation and SIP status codes. With softphone extensions, I see 180 (Ringing) and 200 (OK) as normal status. However, with Siemens IP phone extensions, I see 480 (Temporarily Unavailable) which seems to cause FS to terminate the session. So, FS log shows there is actually no active session which explains why it does not performs DTMF detection for the call session. However, the call to Siemens IP phones actually continues with ringing when an extension handset answers the call is established with the caller with full voice communication. I don't know how FS works but this seems very strange. I would like to know how to get FS to work properly with Siemens IP phones including the DTMF functions during calls. Any help would be appreciated. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091115/66e5c971/attachment-0002.html From jmesquita at freeswitch.org Sun Nov 15 11:07:43 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 15 Nov 2009 17:07:43 -0200 Subject: [Freeswitch-users] Problem with Siemens A580 IP Phones In-Reply-To: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> References: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> Message-ID: I have the same phone and all works fine. Make sure you are setting DTMF to RFC2833 on the phone config page. This log is supposed to go on pastebin (http://pastebin.freeswitch.org), not here. Also, you are making a group call which makes you ring 2 endpoints at the same time. Verify your dialplan and make sure that's what you need/want. Regards, JM On Sun, Nov 15, 2009 at 3:42 PM, wrote: > > I am FS beginner and I have a basic PBX setup using FS with the Siemens > A580 IP Phones. I thought everything was working fine since I could make > and receive basic calls without any obvious issues. However, recently I > wanted to use more advanced functions in FS and discovered that I could not > use any of DTMF based functions (e.g. call transfer/record) during calls > with the Siemens IP phones. The same functions work fine when I use a > softphone. So, I started looking at the log file and I think there is some > problem between the Siemens IP phones and FS (log file attached below). It > seems that when a call comes in, FS calls the extensions and then the > extensions send back confirmation and SIP status codes. With softphone > extensions, I see 180 (Ringing) and 200 (OK) as normal status. However, > with Siemens IP phone extensions, I see 480 (Temporarily Unavailable) which > seems to cause FS to terminate the session. So, FS log shows there is > actually no active session which explains why it does not performs DTMF > detection for the call session. However, the call to Siemens IP phones > actually continues with ringing when an extension handset answers the call > is established with the caller with full voice communication. I don't know > how FS works but this seems very strange. I would like to know how to get > FS to work properly with Siemens IP phones including the DTMF functions > during calls. Any help would be appreciated. > > > ---------------------------------------------- > 2009-11-14 09:35:43.942450 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/4155559999 at 192.168.1.254[22f8ee00-d144-11de-a41f-e5a6b5425f55] > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_NEW > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:404 > (sofia/internal/4155559999 at 192.168.1.254) State NEW > 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3289 Channel sofia/internal/ > 4155559999 at 192.168.1.254 entering state [received][100] > 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3296 Remote SDP: > v=0 > o=- 119640485 119640485 IN IP4 192.168.1.97 > s=- > c=IN IP4 192.168.1.97 > t=0 0 > m=audio 16430 RTP/AVP 0 100 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:115:32000:20] > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:107:16000:20] > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G722:9:8000:20] > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[PCMU:0:8000:20] > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:2029 Set Codec > sofia/internal/4155559999 at 192.168.1.254 PCMU/8000 20 ms 160 samples > 2009-11-14 09:35:43.951943 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload > to 101 > 2009-11-14 09:35:43.951943 [DEBUG] sofia.c:3455 (sofia/internal/ > 4155559999 at 192.168.1.254) State Change CS_NEW -> CS_INIT > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_INIT > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/4155559999 at 192.168.1.254) State INIT > 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:83 sofia/internal/ > 4155559999 at 192.168.1.254 SOFIA INIT > 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 4155559999 at 192.168.1.254) State Change CS_INIT -> CS_ROUTING > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/4155559999 at 192.168.1.254) State INIT going to sleep > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_ROUTING > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/4155559999 at 192.168.1.254) State ROUTING > 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:130 sofia/internal/ > 4155559999 at 192.168.1.254 SOFIA ROUTING > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/4155559999 at 192.168.1.254 Standard ROUTING > 2009-11-14 09:35:43.951943 [INFO] mod_dialplan_xml.c:315 Processing > WIRELESS CALLER->4155553333 in context default > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->unloop] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->tod_example] continue=true > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->global-intercept] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [global-intercept] destination_number(4155553333) =~ /^886$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->group-intercept] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [group-intercept] destination_number(4155553333) =~ /^\*8$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->intercept-ext] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [intercept-ext] destination_number(4155553333) =~ /^\*\*(\d+)$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->redial] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [redial] > destination_number(4155553333) =~ /^870$|^\*66$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->global] continue=true > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [global] > ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ > break=never > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Absolute Condition > [global] > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->snom-demo-2] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [snom-demo-2] destination_number(4155553333) =~ /^9001$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->snom-demo-1] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [snom-demo-1] destination_number(4155553333) =~ /^9000$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->eavesdrop] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [eavesdrop] > destination_number(4155553333) =~ /^88(.*)$|^\*0(.*)$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->eavesdrop] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [eavesdrop] > destination_number(4155553333) =~ /^779$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->call_return] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [call_return] destination_number(4155553333) =~ /^\*69$|^869$|^lcr$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->del-group] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [del-group] > destination_number(4155553333) =~ /^80(\d{2})$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->add-group] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [add-group] > destination_number(4155553333) =~ /^81(\d{2})$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->call-group-simo] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [call-group-simo] destination_number(4155553333) =~ /^82(\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->call-group-order] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [call-group-order] destination_number(4155553333) =~ /^83(\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->extension-intercom] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [extension-intercom] destination_number(4155553333) =~ /^8(10[01][0-9])$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->Local_Extension] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [Local_Extension] destination_number(4155553333) =~ /^(10[01][0-9])$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->group_dial_ringables] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [group_dial_ringables] destination_number(4155553333) =~ /^1999$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->mobile_extensions] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [mobile_extensions] destination_number(4155553333) =~ /^(20[01][0-9])$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->vmain] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [vmain] > destination_number(4155553333) =~ /^vmain$|^4000$$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->vm1000] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [vm1000] > destination_number(4155553333) =~ /^vm1000$|^4100$|^\*98$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->sip_uri] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [sip_uri] > destination_number(4155553333) =~ /^sip:(.*)$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->nb_conferences] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [nb_conferences] destination_number(4155553333) =~ /^(30\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->wb_conferences] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [wb_conferences] destination_number(4155553333) =~ /^(31\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->uwb_conferences] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [uwb_conferences] destination_number(4155553333) =~ /^(32\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->cdquality_conferences] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [cdquality_conferences] destination_number(4155553333) =~ /^(33\d{2})$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->freeswitch_public_conf_via_sip] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [freeswitch_public_conf_via_sip] destination_number(4155553333) =~ > /^9(888|1616|3232)$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->mad_boss_intercom] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [mad_boss_intercom] destination_number(4155553333) =~ /^0911$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->mad_boss_intercom] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [mad_boss_intercom] destination_number(4155553333) =~ /^0912$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->mad_boss] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [mad_boss] > destination_number(4155553333) =~ /^0913$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ivr_demo] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ivr_demo] > destination_number(4155553333) =~ /^5000$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->dynamic_conference] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [dynamic_conference] destination_number(4155553333) =~ /^5001$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->rtp_multicast_page] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [rtp_multicast_page] destination_number(4155553333) =~ /^pagegroup$|^7243$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] > destination_number(4155553333) =~ /^5900$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->unpark] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] > destination_number(4155553333) =~ /^5901$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] > destination_number(4155553333) =~ /park\+(\d+)/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->unpark] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] > destination_number(4155553333) =~ /^parking$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->park] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [park] > destination_number(4155553333) =~ /callpark/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->unpark] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [unpark] > destination_number(4155553333) =~ /pickup/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->wait] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [wait] > destination_number(4155553333) =~ /^wait$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->fax_receive] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [fax_receive] destination_number(4155553333) =~ /^9978$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->fax_transmit] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [fax_transmit] destination_number(4155553333) =~ /^9979$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ringback_180] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [ringback_180] destination_number(4155553333) =~ /^9980$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ringback_183_uk_ring] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [ringback_183_uk_ring] destination_number(4155553333) =~ /^9981$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ringback_183_music_ring] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [ringback_183_music_ring] destination_number(4155553333) =~ /^9982$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ringback_post_answer_uk_ring] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [ringback_post_answer_uk_ring] destination_number(4155553333) =~ /^9983$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ringback_post_answer_music] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [ringback_post_answer_music] destination_number(4155553333) =~ /^9984$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->ClueCon] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [ClueCon] > destination_number(4155553333) =~ /^9991$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->show_info] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [show_info] > destination_number(4155553333) =~ /^9992$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->video_record] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [video_record] destination_number(4155553333) =~ /^9993$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->video_playback] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [video_playback] destination_number(4155553333) =~ /^9994$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->delay_echo] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [delay_echo] destination_number(4155553333) =~ /^9995$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->echo] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [echo] > destination_number(4155553333) =~ /^9996$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->milliwatt] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [milliwatt] > destination_number(4155553333) =~ /^9997$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->tone_stream] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [tone_stream] destination_number(4155553333) =~ /^9998$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->zrtp_enrollement] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [zrtp_enrollement] destination_number(4155553333) =~ /^9787$/ break=on-false > > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->hold_music] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [hold_music] destination_number(4155553333) =~ /^9999$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing [default->fax] > continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [fax] > destination_number(4155553333) =~ /^fax|9777$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->test-9555] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [test-9555] > destination_number(4155553333) =~ /^9555$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->test-9666] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) [test-9666] > destination_number(4155553333) =~ /^9666$/ break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->pizza_demo] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (FAIL) > [pizza_demo] destination_number(4155553333) =~ /^(pizza|74992)$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 parsing > [default->Inbound-4155553333] continue=false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Regex (PASS) > [Inbound-4155553333] destination_number(4155553333) =~ /^4155553333$/ > break=on-false > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action ring_ready() > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(1 b > s execute_extension::dx XML features) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(2 b > s > record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) > > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action bind_meta_app(3 b > s execute_extension::cf XML features) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > set(ringback=${us-ring}) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > set(transfer_ringback=local_stream://moh) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > set(call_timeout=28) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > set(hangup_after_bridge=true) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > set(continue_on_fail=true) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action > bridge(${group_call(ringables@${domain_name})}) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action answer() > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action sleep(1000) > Dialplan: sofia/internal/4155559999 at 192.168.1.254 Action voicemail(default > ${domain_name} 1000) > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:114 > (sofia/internal/4155559999 at 192.168.1.254) State Change CS_ROUTING -> > CS_EXECUTE > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/4155559999 at 192.168.1.254) State ROUTING going to sleep > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_EXECUTE > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/4155559999 at 192.168.1.254) State EXECUTE > 2009-11-14 09:35:43.951943 [DEBUG] mod_sofia.c:173 sofia/internal/ > 4155559999 at 192.168.1.254 SOFIA EXECUTE > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_state_machine.c:151 > sofia/internal/4155559999 at 192.168.1.254 Standard EXECUTE > EXECUTE sofia/internal/4155559999 at 192.168.1.254hash(insert/192.168.1.254-spymap/4155559999/22f8ee00-d144-11de-a41f-e5a6b5425f55) > > EXECUTE sofia/internal/4155559999 at 192.168.1.254hash(insert/192.168.1.254-last_dial/4155559999/4155553333) > EXECUTE sofia/internal/4155559999 at 192.168.1.254hash(insert/192.168.1.254-last_dial/global/22f8ee00-d144-11de-a41f-e5a6b5425f55) > > EXECUTE sofia/internal/4155559999 at 192.168.1.254 ring_ready() > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:415 sofia/internal/ > 4155559999 at 192.168.1.254 receive message [RINGING] > 2009-11-14 09:35:43.951943 [NOTICE] mod_sofia.c:1449 Ring-Ready > sofia/internal/4155559999 at 192.168.1.254! > 2009-11-14 09:35:43.951943 [DEBUG] switch_core_session.c:630 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:43.951943 [NOTICE] mod_dptools.c:415 Ring Ready > sofia/internal/4155559999 at 192.168.1.254! > EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(1 b s > execute_extension::dx XML features) > 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 > execute_extension::dx XML features > EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(2 b s > record_session::/usr/local/freeswitch/recordings/4155559999.2009-11-14-09-35-43.wav) > > 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 > record_session::/usr/local/freeswitch/recordings/4155559999.2009-11-14-09-35-43.wav > > EXECUTE sofia/internal/4155559999 at 192.168.1.254 bind_meta_app(3 b s > execute_extension::cf XML features) > 2009-11-14 09:35:43.951943 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 > execute_extension::cf XML features > EXECUTE sofia/internal/4155559999 at 192.168.1.254set(ringback=%(2000,4000,440.0,480.0)) > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/ > 4155559999 at 192.168.1.254 SET [ringback]=[%(2000,4000,440.0,480.0)] > EXECUTE sofia/internal/4155559999 at 192.168.1.254set(transfer_ringback=local_stream://moh) > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/ > 4155559999 at 192.168.1.254 SET [transfer_ringback]=[local_stream://moh] > EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(call_timeout=28) > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/ > 4155559999 at 192.168.1.254 SET [call_timeout]=[28] > EXECUTE sofia/internal/4155559999 at 192.168.1.254set(hangup_after_bridge=true) > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/ > 4155559999 at 192.168.1.254 SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/4155559999 at 192.168.1.254 set(continue_on_fail=true) > > 2009-11-14 09:35:43.951943 [DEBUG] mod_dptools.c:748 sofia/internal/ > 4155559999 at 192.168.1.254 SET [continue_on_fail]=[true] > 2009-11-14 09:35:43.966601 [DEBUG] sofia.c:3289 Channel sofia/internal/ > 4155559999 at 192.168.1.254 entering state [early][180] > EXECUTE sofia/internal/4155559999 at 192.168.1.254 bridge([presence_id= > 1011 at 192.168.1.254]sofia/internal/sip:1011 at 192.168.1.98:5872,[presence_id= > 1012 at 192.168.1.254]sofia/internal/sip:1012 at 192.168.1.98:5872,[presence_id= > 1014 at 192.168.1.254]sofia/internal/sip:1014 at 192.168.1.97:5060) > 2009-11-14 09:35:43.986485 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/sip:1011 at 192.168.1.98:5872[22ff385a-d144-11de-a41f-e5a6b5425f55] > 2009-11-14 09:35:43.986485 [DEBUG] mod_sofia.c:2811 (sofia/internal/ > sip:1011 at 192.168.1.98:5872) State Change CS_NEW -> CS_INIT > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.990495 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/sip:1012 at 192.168.1.98:5872[22ff6230-d144-11de-a41f-e5a6b5425f55] > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_INIT > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State INIT > 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:83 sofia/internal/ > sip:1011 at 192.168.1.98:5872 SOFIA INIT > 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:111 (sofia/internal/ > sip:1011 at 192.168.1.98:5872) State Change CS_INIT -> CS_ROUTING > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State INIT going to sleep > 2009-11-14 09:35:43.990495 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1011 at 192.168.1.98:5872 entering state [calling][0] > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change > CS_ROUTING > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State ROUTING > 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:130 sofia/internal/ > sip:1011 at 192.168.1.98:5872 SOFIA ROUTING > 2009-11-14 09:35:43.990495 [DEBUG] switch_ivr_originate.c:63 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State ROUTING going to sleep > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State CONSUME_MEDIA > 2009-11-14 09:35:43.990495 [DEBUG] mod_sofia.c:2811 (sofia/internal/ > sip:1012 at 192.168.1.98:5872) State Change CS_NEW -> CS_INIT > 2009-11-14 09:35:43.990495 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.994449 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/sip:1014 at 192.168.1.97:5060[22fffdb2-d144-11de-a41f-e5a6b5425f55] > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_INIT > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State INIT > 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:83 sofia/internal/ > sip:1012 at 192.168.1.98:5872 SOFIA INIT > 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:111 (sofia/internal/ > sip:1012 at 192.168.1.98:5872) State Change CS_INIT -> CS_ROUTING > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.994449 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1012 at 192.168.1.98:5872 entering state [calling][0] > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State INIT going to sleep > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change > CS_ROUTING > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State ROUTING > 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:130 sofia/internal/ > sip:1012 at 192.168.1.98:5872 SOFIA ROUTING > 2009-11-14 09:35:43.994449 [DEBUG] switch_ivr_originate.c:63 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State ROUTING going to sleep > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State CONSUME_MEDIA > 2009-11-14 09:35:43.994449 [DEBUG] mod_sofia.c:2811 (sofia/internal/ > sip:1014 at 192.168.1.97:5060) State Change CS_NEW -> CS_INIT > 2009-11-14 09:35:43.994449 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] > 2009-11-14 09:35:43.998457 [DEBUG] switch_ivr_originate.c:1701 > sofia/internal/4155559999 at 192.168.1.254 receive message [PROGRESS] > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_INIT > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State INIT > 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:83 sofia/internal/ > sip:1014 at 192.168.1.97:5060 SOFIA INIT > 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:111 (sofia/internal/ > sip:1014 at 192.168.1.97:5060) State Change CS_INIT -> CS_ROUTING > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] > 2009-11-14 09:35:43.998457 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1014 at 192.168.1.97:5060 entering state [calling][0] > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State INIT going to sleep > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change > CS_ROUTING > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State ROUTING > 2009-11-14 09:35:43.998457 [DEBUG] mod_sofia.c:130 sofia/internal/ > sip:1014 at 192.168.1.97:5060 SOFIA ROUTING > 2009-11-14 09:35:43.998457 [DEBUG] switch_ivr_originate.c:63 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State ROUTING going to sleep > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change > CS_CONSUME_MEDIA > 2009-11-14 09:35:43.998457 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State CONSUME_MEDIA > 2009-11-14 09:35:43.998457 [INFO] switch_ivr_originate.c:1701 Sending early > media > 2009-11-14 09:35:44.2435 [DEBUG] sofia_glue.c:2263 AUDIO RTP > [sofia/internal/4155559999 at 192.168.1.254] 192.168.1.254 port 31052 -> > 192.168.1.97 port 16430 codec: 0 ms: 20 > 2009-11-14 09:35:44.2435 [DEBUG] switch_rtp.c:1138 Starting timer [soft] > 160 bytes per 20ms > 2009-11-14 09:35:44.6432 [INFO] mod_sofia.c:1506 Ring SDP: > v=0 > o=FreeSWITCH 1258189091 1258189092 IN IP4 192.168.1.254 > s=FreeSWITCH > c=IN IP4 192.168.1.254 > t=0 0 > m=audio 31052 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2009-11-14 09:35:44.6432 [NOTICE] mod_sofia.c:1509 Pre-Answer > sofia/internal/4155559999 at 192.168.1.254! > 2009-11-14 09:35:44.6432 [DEBUG] sofia.c:3289 Channel sofia/internal/ > 4155559999 at 192.168.1.254 entering state [early][183] > 2009-11-14 09:35:44.6432 [DEBUG] switch_core_session.c:630 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:44.6432 [DEBUG] switch_ivr_originate.c:1718 Raw Codec > Activation Success L16 at 8000hz 1 channel 20ms > 2009-11-14 09:35:44.6432 [DEBUG] switch_ivr_originate.c:1777 Play Ringback > Tone [%(2000,4000,440.0,480.0)] > 2009-11-14 09:35:44.18430 [DEBUG] switch_core_io.c:649 sofia/internal/ > 4155559999 at 192.168.1.254 receive message [TRANSCODING_NECESSARY] > 2009-11-14 09:35:44.22473 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1014 at 192.168.1.97:5060 entering state [proceeding][180] > 2009-11-14 09:35:44.22473 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/ > sip:1014 at 192.168.1.97:5060! > 2009-11-14 09:35:52.326423 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1011 at 192.168.1.98:5872 entering state [terminated][480] > 2009-11-14 09:35:52.326423 [NOTICE] sofia.c:3849 Hangup sofia/internal/ > sip:1011 at 192.168.1.98:5872 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > 2009-11-14 09:35:52.326423 [DEBUG] switch_channel.c:1683 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [KILL] > 2009-11-14 09:35:52.326423 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State CONSUME_MEDIA going to > sleep > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change CS_HANGUP > > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State HANGUP > 2009-11-14 09:35:52.330490 [DEBUG] mod_sofia.c:306 sofia/internal/ > sip:1011 at 192.168.1.98:5872 Overriding SIP cause 408 with 480 from the > other leg > 2009-11-14 09:35:52.330490 [DEBUG] mod_sofia.c:338 Channel sofia/internal/ > sip:1011 at 192.168.1.98:5872 hanging up, cause: NO_USER_RESPONSE > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/sip:1011 at 192.168.1.98:5872 Standard HANGUP, cause: > NO_USER_RESPONSE > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State HANGUP going to sleep > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:476 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_HANGUP -> > CS_REPORTING > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1011 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Running State Change > CS_REPORTING > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State REPORTING > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/sip:1011 at 192.168.1.98:5872 Standard REPORTING, cause: > NO_USER_RESPONSE > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State REPORTING going to sleep > > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_state_machine.c:411 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State Change CS_REPORTING -> > CS_DESTROY > 2009-11-14 09:35:52.330490 [DEBUG] switch_core_session.c:1068 Session 542 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Locked, Waiting on external > entities > 2009-11-14 09:35:52.618414 [DEBUG] sofia.c:3289 Channel sofia/internal/ > sip:1012 at 192.168.1.98:5872 entering state [terminated][480] > 2009-11-14 09:35:52.618414 [NOTICE] sofia.c:3849 Hangup sofia/internal/ > sip:1012 at 192.168.1.98:5872 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > 2009-11-14 09:35:52.618414 [DEBUG] switch_channel.c:1683 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [KILL] > 2009-11-14 09:35:52.618414 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State CONSUME_MEDIA going to > sleep > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change CS_HANGUP > > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State HANGUP > 2009-11-14 09:35:52.626426 [DEBUG] mod_sofia.c:306 sofia/internal/ > sip:1012 at 192.168.1.98:5872 Overriding SIP cause 408 with 480 from the > other leg > 2009-11-14 09:35:52.626426 [DEBUG] mod_sofia.c:338 Channel sofia/internal/ > sip:1012 at 192.168.1.98:5872 hanging up, cause: NO_USER_RESPONSE > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/sip:1012 at 192.168.1.98:5872 Standard HANGUP, cause: > NO_USER_RESPONSE > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State HANGUP going to sleep > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:476 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_HANGUP -> > CS_REPORTING > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1012 at 192.168.1.98:5872 [BREAK] > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Running State Change > CS_REPORTING > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State REPORTING > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/sip:1012 at 192.168.1.98:5872 Standard REPORTING, cause: > NO_USER_RESPONSE > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State REPORTING going to sleep > > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_state_machine.c:411 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State Change CS_REPORTING -> > CS_DESTROY > 2009-11-14 09:35:52.626426 [DEBUG] switch_core_session.c:1068 Session 543 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Locked, Waiting on external > entities > 2009-11-14 09:35:59.778421 [DEBUG] sofia.c:3289 Channel sofia/internal/ > 4155559999 at 192.168.1.254 entering state [terminated][487] > 2009-11-14 09:35:59.778421 [NOTICE] sofia.c:3849 Hangup sofia/internal/ > 4155559999 at 192.168.1.254 [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2009-11-14 09:35:59.778421 [DEBUG] switch_channel.c:1683 Send signal > sofia/internal/4155559999 at 192.168.1.254 [KILL] > 2009-11-14 09:35:59.778421 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:59.798426 [NOTICE] switch_ivr_originate.c:1994 Hangup > sofia/internal/sip:1014 at 192.168.1.97:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2009-11-14 09:35:59.798426 [DEBUG] switch_channel.c:1683 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [KILL] > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] > 2009-11-14 09:35:59.798426 [DEBUG] switch_ivr_originate.c:2134 Originate > Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1086 Session 542 > (sofia/internal/sip:1011 at 192.168.1.98:5872) Ended > 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/sip:1011 at 192.168.1.98:5872 [CS_DESTROY] > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:255 sofia/internal/ > sip:1011 at 192.168.1.98:5872 SOFIA DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/sip:1011 at 192.168.1.98:5872 Standard DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1011 at 192.168.1.98:5872) State DESTROY going to sleep > 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1086 Session 543 > (sofia/internal/sip:1012 at 192.168.1.98:5872) Ended > 2009-11-14 09:35:59.798426 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/sip:1012 at 192.168.1.98:5872 [CS_DESTROY] > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:255 sofia/internal/ > sip:1012 at 192.168.1.98:5872 SOFIA DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/sip:1012 at 192.168.1.98:5872 Standard DESTROY > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1012 at 192.168.1.98:5872) State DESTROY going to sleep > 2009-11-14 09:35:59.798426 [INFO] mod_dptools.c:2093 Originate Failed. > Cause: ORIGINATOR_CANCEL > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/4155559999 at 192.168.1.254) State EXECUTE going to sleep > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change CS_HANGUP > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/4155559999 at 192.168.1.254) State HANGUP > 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:306 sofia/internal/ > 4155559999 at 192.168.1.254 Overriding SIP cause 487 with 487 from the other > leg > 2009-11-14 09:35:59.798426 [DEBUG] mod_sofia.c:338 Channel sofia/internal/ > 4155559999 at 192.168.1.254 hanging up, cause: ORIGINATOR_CANCEL > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/4155559999 at 192.168.1.254 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2009-11-14 09:35:59.798426 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/4155559999 at 192.168.1.254) State HANGUP going to sleep > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:503 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State CONSUME_MEDIA going to > sleep > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change CS_HANGUP > > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State HANGUP > 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:306 sofia/internal/ > sip:1014 at 192.168.1.97:5060 Overriding SIP cause 487 with 487 from the > other leg > 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:338 Channel sofia/internal/ > sip:1014 at 192.168.1.97:5060 hanging up, cause: ORIGINATOR_CANCEL > 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:406 Sending CANCEL to > sofia/internal/sip:1014 at 192.168.1.97:5060 > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/sip:1014 at 192.168.1.97:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State HANGUP going to sleep > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:476 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_HANGUP -> > CS_REPORTING > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/sip:1014 at 192.168.1.97:5060 [BREAK] > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Running State Change > CS_REPORTING > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State REPORTING > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/sip:1014 at 192.168.1.97:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State REPORTING going to sleep > > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:411 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State Change CS_REPORTING -> > CS_DESTROY > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_session.c:1068 Session 544 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Locked, Waiting on external > entities > 2009-11-14 09:35:59.810601 [NOTICE] switch_core_session.c:1086 Session 544 > (sofia/internal/sip:1014 at 192.168.1.97:5060) Ended > 2009-11-14 09:35:59.810601 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/sip:1014 at 192.168.1.97:5060 [CS_DESTROY] > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State DESTROY > 2009-11-14 09:35:59.810601 [DEBUG] mod_sofia.c:255 sofia/internal/ > sip:1014 at 192.168.1.97:5060 SOFIA DESTROY > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/sip:1014 at 192.168.1.97:5060 Standard DESTROY > 2009-11-14 09:35:59.810601 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/sip:1014 at 192.168.1.97:5060) State DESTROY going to sleep > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:476 > (sofia/internal/4155559999 at 192.168.1.254) State Change CS_HANGUP -> > CS_REPORTING > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/4155559999 at 192.168.1.254 [BREAK] > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/4155559999 at 192.168.1.254) Running State Change > CS_REPORTING > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/4155559999 at 192.168.1.254) State REPORTING > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/4155559999 at 192.168.1.254 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:612 > (sofia/internal/4155559999 at 192.168.1.254) State REPORTING going to sleep > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:411 > (sofia/internal/4155559999 at 192.168.1.254) State Change CS_REPORTING -> > CS_DESTROY > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_session.c:1068 Session 541 > (sofia/internal/4155559999 at 192.168.1.254) Locked, Waiting on external > entities > 2009-11-14 09:35:59.814986 [NOTICE] switch_core_session.c:1086 Session 541 > (sofia/internal/4155559999 at 192.168.1.254) Ended > 2009-11-14 09:35:59.814986 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/4155559999 at 192.168.1.254 [CS_DESTROY] > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/4155559999 at 192.168.1.254) State DESTROY > 2009-11-14 09:35:59.814986 [DEBUG] mod_sofia.c:255 sofia/internal/ > 4155559999 at 192.168.1.254 SOFIA DESTROY > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/4155559999 at 192.168.1.254 Standard DESTROY > 2009-11-14 09:35:59.814986 [DEBUG] switch_core_state_machine.c:564 > (sofia/internal/4155559999 at 192.168.1.254) State DESTROY going to sleep > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091115/7bad9a71/attachment-0002.html From brian at freeswitch.org Sun Nov 15 14:14:08 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Nov 2009 16:14:08 -0600 Subject: [Freeswitch-users] Registration Error 408 In-Reply-To: <4B004E27.3040600@greatiam.com> References: <4AFF5558.3080408@greatiam.com> <34223AC5-699B-499B-A3B9-CED0F9CF1C59@freeswitch.org> <4B004E27.3040600@greatiam.com> Message-ID: <5E9EF7D5-2154-484E-9565-48A276A7D8A3@freeswitch.org> service iptables stop /b On Nov 15, 2009, at 12:53 PM, Samuel Abekah-Mensah wrote: > Hi Brian > > I have FS running on a FC11 box and the clients IDs 1001 and 1002 > running off 2 Windows boxes using X-lite3 > > Thanks > From ahmedmunir007 at gmail.com Sun Nov 15 23:29:50 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Mon, 16 Nov 2009 12:29:50 +0500 Subject: [Freeswitch-users] How to implement mod_lcr + mod_limit Message-ID: Hi, I've worked on setup for carriers routing using mod_lcr + mod_nibble + mod_xml_curl and mod_xml_cdr. The setup is working fine as I desired. Now I want to include mod_limit in to my setup. As I read the wiki pages of mod_limit I want to know how can I limit the calls per destination basis while running mod_lcr? Because LCR is routing to different carriers, how can I call mod_limit in mod_lcr? Kindly advise this issue soon. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/4dd76107/attachment-0002.html From mustafa.pk at gmail.com Sun Nov 15 23:45:49 2009 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Mon, 16 Nov 2009 12:45:49 +0500 Subject: [Freeswitch-users] How to implement mod_lcr + mod_limit In-Reply-To: References: Message-ID: <8213d6070911152345k3f67fb93y730c46f085e386fc@mail.gmail.com> Ahmed, i hope you find the answer here. http://wiki.freeswitch.org/wiki/Mod_limit#limit_hash_execute -m On Mon, Nov 16, 2009 at 12:29 PM, Ahmed Munir wrote: > Hi, > > I've worked on setup for carriers routing using mod_lcr + mod_nibble + > mod_xml_curl and mod_xml_cdr. The setup is working fine as I desired. Now I > want to include mod_limit in to my setup. > > As I read the wiki pages of mod_limit I want to know how can I limit the > calls per destination basis while running mod_lcr? Because LCR is routing to > different carriers, how can I call mod_limit in mod_lcr? > > Kindly advise this issue soon. > > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com From vedamaker at netscape.net Mon Nov 16 00:29:36 2009 From: vedamaker at netscape.net (VedaM) Date: Mon, 16 Nov 2009 00:29:36 -0800 (PST) Subject: [Freeswitch-users] Problem with Siemens A580 IP Phones In-Reply-To: References: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> Message-ID: <26368294.post@talk.nabble.com> I discovered that the problem was due to having the incoming PSTN line connected to FS and the Siemens A580 base station. The Siemens A580 handsets were configured not to accept incoming calls from the PSTN but that does not seem work. In any case, I have disconnected the PSTN line from the Siemens A580 base station but I am still not able to use FS to transfer incoming calls between the Siemens A580 handsets. When I answer an incoming call with one handset (ext 1011) and use the "*1" DTMF command to transfer to another handset (ext 1012) the call is transferred to voicemail because the second handset (ext 1012) returns a 486 (Busy Here) status. Since you are using the same phones, can you tell me if you are able to use FS to transfer calls between handsets? If yes, how are you doing it? Your help is appreciated. Jo?o Mesquita-4 wrote: > > I have the same phone and all works fine. Make sure you are setting DTMF > to > RFC2833 on the phone config page. > > This log is supposed to go on pastebin (http://pastebin.freeswitch.org), > not > here. > > Also, you are making a group call which makes you ring 2 endpoints at the > same time. Verify your dialplan and make sure that's what you need/want. > > Regards, > > JM > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Problem-with-Siemens-A580-IP-Phones-tp26361779p26368294.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Mon Nov 16 00:57:53 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 16 Nov 2009 09:57:53 +0100 Subject: [Freeswitch-users] siptrace/debug log timestamp difference Message-ID: <65d96fc80911160057u1171bc41wc196d4719c10edc4@mail.gmail.com> Hi, just a thing i noticed... the debug log and sip trace have different time ... one hour difference ... looks like UTC/GMT issue. where do i set the time for siptrace correctly ? 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:411 (sofia/external/00010038516659280 at 10.4.5.107:5060) State Change CS_REPORTING -> CS_DESTROY 2009-11-16 09:47:13.779070 [DEBUG] switch_core_session.c:1068 Session 31 (sofia/external/00010038516659280 at 10.4.5.107:5060) Locked, Waiting on external entities 2009-11-16 09:47:13.779070 [NOTICE] switch_core_session.c:1086 Session 31 (sofia/external/00010038516659280 at 10.4.5.107:5060) Ended 2009-11-16 09:47:13.779070 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/00010038516659280 at 10.4.5.107:5060 [CS_DESTROY] 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:564 (sofia/external/00010038516659280 at 10.4.5.107:5060) State DESTROY 2009-11-16 09:47:13.779070 [DEBUG] mod_sofia.c:255 sofia/external/ 00010038516659280 at 10.4.5.107:5060 SOFIA DESTROY 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:60 sofia/external/00010038516659280 at 10.4.5.107:5060 Standard DESTROY 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:564 (sofia/external/00010038516659280 at 10.4.5.107:5060) State DESTROY going to sleep recv 462 bytes from udp/[10.4.5.107]:5060 at 08:47:13.799578: ------------------------------------------------------------------------ ACK sip:30003038515000403 at l01sipindir2.ot.hr:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.4.5.107:5060 ;branch=z9hG4bKterm-13e-30003038515000403-00010038516659280-59521 From: 00010038516659280 ;tag=261638185 To: 30003038515000403 ;tag=9v58macH5mNNH Call-ID: 3189ce3b-3da37db2-3ac943f-140 at 10.4.5.107 CSeq: 1 ACK Max-Forwards: 10 Content-Length: 0 ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/4cd45639/attachment-0002.html From rupa at rupa.com Mon Nov 16 05:39:20 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 16 Nov 2009 07:39:20 -0600 Subject: [Freeswitch-users] Problem with Siemens A580 IP Phones In-Reply-To: <26368294.post@talk.nabble.com> References: <8CC34321B5B218F-D84-1D705@webmail-d079.sysops.aol.com> <26368294.post@talk.nabble.com> Message-ID: Just use the siemens to do the transfer. Hit the menu key, select internal (if you want to do an siemens-siemens transfer), select the extension, hit talk, then menu and choose conference or transfer. If youw ant another extension, then choose external, dial the extension, and the rest is the same. On Mon, Nov 16, 2009 at 2:29 AM, VedaM wrote: > > > I discovered that the problem was due to having the incoming PSTN line > connected to FS and the Siemens A580 base station. ?The Siemens A580 > handsets were configured not to accept incoming calls from the PSTN but that > does not seem work. ?In any case, I have disconnected the PSTN line from the > Siemens A580 base station but I am still not able to use FS to transfer > incoming calls between the Siemens A580 handsets. ?When I answer an incoming > call with one handset (ext 1011) and use the "*1" DTMF command to transfer > to another handset (ext 1012) the call is transferred to voicemail because > the second handset (ext 1012) returns a 486 (Busy Here) status. ?Since you > are using the same phones, can you tell me if you are able to use FS to > transfer calls between handsets? ?If yes, how are you doing it? ?Your help > is appreciated. > > > > Jo?o Mesquita-4 wrote: >> >> I have the same phone and all works fine. Make sure you are setting DTMF >> to >> RFC2833 on the phone config page. >> >> This log is supposed to go on pastebin (http://pastebin.freeswitch.org), >> not >> here. >> >> Also, you are making a group call which makes you ring 2 endpoints at the >> same time. Verify your dialplan and make sure that's what you need/want. >> >> Regards, >> >> JM >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://old.nabble.com/Problem-with-Siemens-A580-IP-Phones-tp26361779p26368294.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From michaelt at voxcore.voxtelecom.co.za Sun Nov 15 23:07:01 2009 From: michaelt at voxcore.voxtelecom.co.za (Michael Toop) Date: Mon, 16 Nov 2009 09:07:01 +0200 Subject: [Freeswitch-users] DTMF Digits Lost when Under Load Message-ID: <330316f60911152307w2800f2e1r87c77d6dcd70be65@mail.gmail.com> Hi All, I have an issue that when my call volumes on my FS IVR box > 30 calls DTMF digits are lost (using RFC2833). It is definitely load related as it all works perfectly under 30 calls. Any pointers or a solution to the problem? Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/be8e93e6/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 16 07:25:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Nov 2009 09:25:51 -0600 Subject: [Freeswitch-users] DTMF Digits Lost when Under Load In-Reply-To: <330316f60911152307w2800f2e1r87c77d6dcd70be65@mail.gmail.com> References: <330316f60911152307w2800f2e1r87c77d6dcd70be65@mail.gmail.com> Message-ID: <191c3a030911160725k38ebcda8ta8c38c36eb80e627@mail.gmail.com> That's a pretty small problem description to be so sure about something. It would probably be better to capture some evidence of the exact problem you are having since we are using computers and we need to see the computers in action doing something specifically incorrect to diagnose any sort of problem. Take the time to describe the origin and destination of your calls, the call flow, the hardware in use on both ends of the call, detailed console logs on debug level, (maybe even uncomment the 2833 debug ifded in switch_rtp.c) and gather something to go on besides "I seem to be losing dtmf) maybe a packect capture of the networking interface on both ends of these calls. Also problems should be reported to http://jira.freeswitch.org not this mailing list. Save us a step if you report a jira and provide all the info above or we will just have to ask for it again. On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop < michaelt at voxcore.voxtelecom.co.za> wrote: > Hi All, > > I have an issue that when my call volumes on my FS IVR box > 30 calls DTMF > digits are lost (using RFC2833). It is definitely load related as it all > works perfectly under 30 calls. > > Any pointers or a solution to the problem? > > Thanks, > > Michael > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/a4927b96/attachment-0002.html From dschwartz at xconnect.net Mon Nov 16 07:45:49 2009 From: dschwartz at xconnect.net (David Schwartz) Date: Mon, 16 Nov 2009 17:45:49 +0200 Subject: [Freeswitch-users] gtalk Message-ID: <6EA53FAD386F9D46B97D49BFE148D5140603A069@ISR-JLM-MAIL1.xconnect.co.il> Thanks to everyone for all the help. I finally got gtalk to work in both directions - well almost. I have the gtalk client indicating that user 1000 1000 is trying to call him only there is no button on the gtalk client to answer the call. (when I call from a gtalk client to gtalk client there is an "answer" button). Has anyone encountered this problem? Only other weird part is that in the debug log I see: 2009-11-16 15:59:01.517850 [DEBUG] libdingaling.c:1228 sasl authentication failed 2009-11-16 15:59:01.517850 [DEBUG] libdingaling.c:1546 io error 2 7 retry in 20 second(s) even though I have connected to the XMPP server (the user I have in the client profile is "online" in the gtalk client) Don't know if these are related. Thanks, David From brian at freeswitch.org Mon Nov 16 08:20:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Nov 2009 10:20:20 -0600 Subject: [Freeswitch-users] gtalk In-Reply-To: <6EA53FAD386F9D46B97D49BFE148D5140603A069@ISR-JLM-MAIL1.xconnect.co.il> References: <6EA53FAD386F9D46B97D49BFE148D5140603A069@ISR-JLM-MAIL1.xconnect.co.il> Message-ID: I think you missed this step http://wiki.freeswitch.org/wiki/Dingaling#TLS So your dingaling is failing to work properly... :P /b On Nov 16, 2009, at 9:45 AM, David Schwartz wrote: > Thanks to everyone for all the help. > > I finally got gtalk to work in both directions - well almost. > > I have the gtalk client indicating that user 1000 1000 is trying to > call him only there is no button on the gtalk client to answer the > call. (when I call from a gtalk client to gtalk client there is an > "answer" button). > > Has anyone encountered this problem? > > Only other weird part is that in the debug log I see: > > 2009-11-16 15:59:01.517850 [DEBUG] libdingaling.c:1228 sasl > authentication failed > > 2009-11-16 15:59:01.517850 [DEBUG] libdingaling.c:1546 io error 2 7 > retry in 20 second(s) > > even though I have connected to the XMPP server (the user I have in > the client profile is "online" in the gtalk client) > > Don't know if these are related. > > Thanks, > > David From michaelt at voxcore.voxtelecom.co.za Mon Nov 16 09:20:11 2009 From: michaelt at voxcore.voxtelecom.co.za (Michael Toop) Date: Mon, 16 Nov 2009 19:20:11 +0200 Subject: [Freeswitch-users] DTMF Digits Lost when Under Load In-Reply-To: <191c3a030911160725k38ebcda8ta8c38c36eb80e627@mail.gmail.com> References: <330316f60911152307w2800f2e1r87c77d6dcd70be65@mail.gmail.com> <191c3a030911160725k38ebcda8ta8c38c36eb80e627@mail.gmail.com> Message-ID: <330316f60911160920n78edd236h82c6c1e7f71de1b6@mail.gmail.com> Hi Anthony, Thanks for the input. I will try & reproduce the problem & give you something more concrete to work with & log it in Jira. Thanks again, Michael On Mon, Nov 16, 2009 at 5:25 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > That's a pretty small problem description to be so sure about something. > It would probably be better to capture some evidence of the exact problem > you are having since we are using computers and we need to see the computers > in action doing something specifically incorrect to diagnose any sort of > problem. Take the time to describe the origin and destination of your > calls, the call flow, the hardware in use on both ends of the call, detailed > console logs on debug level, (maybe even uncomment the 2833 debug ifded in > switch_rtp.c) and gather something to go on besides "I seem to be losing > dtmf) maybe a packect capture of the networking interface on both ends of > these calls. > > Also problems should be reported to http://jira.freeswitch.org not this > mailing list. > Save us a step if you report a jira and provide all the info above or we > will just have to ask for it again. > > > On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop < > michaelt at voxcore.voxtelecom.co.za> wrote: > >> Hi All, >> >> I have an issue that when my call volumes on my FS IVR box > 30 calls >> DTMF digits are lost (using RFC2833). It is definitely load related as it >> all works perfectly under 30 calls. >> >> Any pointers or a solution to the problem? >> >> Thanks, >> >> Michael >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/dca6d724/attachment-0002.html From msc at freeswitch.org Mon Nov 16 09:21:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 09:21:15 -0800 Subject: [Freeswitch-users] Registration Error - 408 timeout In-Reply-To: <4AFF5701.8010508@greatiam.com> References: <4AFF5701.8010508@greatiam.com> Message-ID: <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> On Sat, Nov 14, 2009 at 5:18 PM, Samuel Abekah-Mensah wrote: > Hello > > Please pardon me if the solution to this is somewhere already that I > have been unable to locate. I have just got a straight out-of-the-box > build of FS. According to the wiki, I should be able to test using user > IDs 1001 and 1002. However, I am get the above error. If I, however, > un-tick register with domain I do net get the error but does not > communicate either. Is there a conf that I should have done ? > > I am using X-lite3 > > Is NAT involved or are the x-lite clients on the same LAN? Also, you might want to turn on a SIP trace at the console to see if there are any clues. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/d27593ba/attachment-0002.html From jerry.richards at teotech.com Mon Nov 16 09:36:01 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 16 Nov 2009 09:36:01 -0800 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> Message-ID: <1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> I have a bit of confusion about Lua scripting. When a script is invoked, should it always return an XML string that is used by FS? Or as in the case of dialplan examples, does it actually execute the dialplan (e.g. "session:answer();")? Best Regards, Jerry -----Original Message----- From: Leon de Rooij [mailto:leon at scarlet-internet.nl] Sent: Friday, November 13, 2009 2:29 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Accessing Config Info From Database Hi, You can use mod_xml_curl (generate xml on a webserver): http://wiki.freeswitch.org/wiki/Mod_xml_curl or mod_xml_odbc (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Mod_xml_odbc or LUA together with luasql (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration regards, Leon On Fri, 2009-11-13 at 13:59 -0800, Jerry Richards wrote: > Is there a way to access configuration information from a database (e.g. > SQL) rather than from the XML files? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From Prometheus001 at gmx.net Mon Nov 16 10:26:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 16 Nov 2009 19:26:25 +0100 Subject: [Freeswitch-users] att_xfer and Loopback In-Reply-To: <191c3a030911120750p34d27d44u2fec4015caf2f367@mail.gmail.com> References: <4AFB3A3D.1050602@gmx.net> <191c3a030911112011i7f98f440s953dc1cc5f9db05@mail.gmail.com> <191c3a030911112012i63000f3j9867308057c5f318@mail.gmail.com> <4AFBC98C.4070602@gmx.net> <191c3a030911120750p34d27d44u2fec4015caf2f367@mail.gmail.com> Message-ID: <4B019951.7070207@gmx.net> Hello Anthony, I made a console trace today: http://pastebin.freeswitch.org/11125 Different from the mail below, in this case A and C have voice. Best regards Peter Anthony Minessale schrieb: > if you provide a console trace of both situations with console > loglevel debug and put them on pastebin i can tell you what's happening. > > > On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX > wrote: > > Thanks Anthony, > > however this rather deteriorated the situation. > Now it works the following > - A calls B > - B enters *4 gets an announcement and enters digits for C (A get MOH) > - C is called > - As soon as C picks up the call, A and C both have no voice (and B is > dropped) > - When A hangs up, C hangs up > > Before it did: > - A calls B > - B enters *4 gets an announcement and enters digits for C (A get MOH) > - C is called > - As soon as C picks up the call, A and C are connected and B is > dropped > - When A hangs up, C hangs up > > Best regards > Peter > > Anthony Minessale schrieb: > > hit send too soon > > you want to set loopback_bowout=false > > > > This keeps loopback from trying to destroy itself when it sees a > > chance to cut out of the call path. > > > > > > On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale > > > >> wrote: > > > > > > set/export the channel variable loopback_bowout=true so it's on > > the loopback leg > > > > > > > > > > On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX > > > >> wrote: > > > > Hello, > > > > I have some problems with attended transfer and loopback > > > > Scenario how id work > > - A calls B > > - B enters *4 gets an announcement and enter digits for C (A > > get MOH) > > - C is called > > - As soon as C picks up the call, A and C are connected > and B > > is dropped > > > > How it work until here: > > - A calls B > > - B enters *4 gets an announcement and enter digits for C (A > > get MOH) > > - C is called > > - As soon as C picks up the call, B and C are connected (A > > still MOH) > > > > The dial string for C is dynamic and dependent on certain > > parameters, > > therefore C must be called via Loopback in our scenario. > > > > > > Here are the configs: > > In dialplan for calling B: > > > > > > Dialplan for executing the att_xfer: > > > > > expression="^attended_xfer$"> > > data="continue_on_fail=true"/> > > > > data="origination_cancel_key=#"/> > > > data="loopback/${attxfer_callthis}"/> > > > > > > > > So this is pretty standard, except the loopback. SVN is > 15322. > > > > Anybody has a solution for this? > > > > > > Best regards > > Peter > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Nov 16 11:17:04 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 11:17:04 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conf Call - Nov 20 Message-ID: <87f2f3b90911161117k1a0ef20cs80dda5d89beaefe5@mail.gmail.com> FYI, I've added the skeleton of the agenda for this week's call: http://wiki.freeswitch.org/wiki/FS_weekly_2009_11_20 The agendas have been pretty light lately. I would like everyone to think about questions that could be brought up for discussion. Also, I'd like to take this time to say thank you to the many folks who have signed up for the wiki lately and have been adding content. We've had quite a few people join over the past few weeks and they have been doing tweaks and adding content to the wiki. We definitely appreciate your help. If you would like to help out with anything else on the wiki (or other janitorial projects) please contact me off list. We have lots of wiki and JIRA things that folks can help with. Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/00d190fd/attachment-0002.html From abeka at greatiam.com Mon Nov 16 11:31:58 2009 From: abeka at greatiam.com (Samuel 'Otis' Abekah-Mensah) Date: Mon, 16 Nov 2009 19:31:58 +0000 Subject: [Freeswitch-users] Registration Error - 408 timeout In-Reply-To: <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> Message-ID: <4B01A8AE.7070708@greatiam.com> Hello thanks so much. The machines are on the same lan , 2 have static IP with one on DHCP just for variation . I do get there errors on stating FS 1. Error stacksize too large 4194303 offers advise to run ./freeswitch -wate 2. Error checking for PMP [GENERAL ERROR] and 3. [WARNING] sofia_reg.c:1788: Can't register a pointer I do not know if any of this is could help The 2 boxes I run X-lite from are windows 2k service pack 4 Oh I ahve had a go and I am now getting Error 403 - Forbidden on the Xlite clients side. I have also tried using Zoiper but it seems to register but then comes up with an error "bearercapability " Thanks for your time, Michael and may thanks Brian. I am not sure if the iptables bit has caused the change from error 408 to error 403. Thanks; I apperecitae your help . Michael Collins wrote: > > > On Sat, Nov 14, 2009 at 5:18 PM, Samuel Abekah-Mensah > > wrote: > > Hello > > Please pardon me if the solution to this is somewhere already that I > have been unable to locate. I have just got a straight out-of-the-box > build of FS. According to the wiki, I should be able to test using > user > IDs 1001 and 1002. However, I am get the above error. If I, however, > un-tick register with domain I do net get the error but does not > communicate either. Is there a conf that I should have done ? > > I am using X-lite3 > > > Is NAT involved or are the x-lite clients on the same LAN? Also, you > might want to turn on a SIP trace at the console to see if there are > any clues. > -MC > From msc at freeswitch.org Mon Nov 16 11:33:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 11:33:09 -0800 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> <1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> Message-ID: <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards wrote: > > I have a bit of confusion about Lua scripting. When a script is invoked, > should it always return an XML string that is used by FS? Or as in the > case > of dialplan examples, does it actually execute the dialplan (e.g. > "session:answer();")? > > Best Regards, > Jerry > > Jerry, A Lua script that is explicitly called from the dialplan will indeed execute dialplan-ish stuff. For example, let's say you had this in conf/dialplan/default.xml: Then myluascript.lua has something like: --Sample Lua script session:answer() session:sleep(1000) session:streamFile("/path/to/file.wav") session:hangup() Assuming an otherwise default install, the above Lua script would execute when a caller dialed 9876, or if a call was x-ferred to 9876. However, if you're wanting to use Lua to serve up a dialplan then it's totally different. Lua is not called from the dialplan; Lua provides the dialplan to FreeSWITCH. This latter case is the scenario discussed in the wiki section you referenced. ( http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration) Are you trying to use Lua scripting for serving up a dynamic configuration of some sort? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/c73ec21e/attachment-0002.html From leon at scarlet-internet.nl Mon Nov 16 14:02:53 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 16 Nov 2009 23:02:53 +0100 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> <1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> Message-ID: <1258408973.9730.91.camel@desk.bofh.scarlet-internet.nl> Hi, Since recently it's also possible to use lua *as* a dialplan: http://wiki.freeswitch.org/wiki/Mod_lua#For_dialplan regards, Leon On Mon, 2009-11-16 at 11:33 -0800, Michael Collins wrote: > > > On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards > wrote: > > I have a bit of confusion about Lua scripting. When a script > is invoked, > should it always return an XML string that is used by FS? Or > as in the case > of dialplan examples, does it actually execute the dialplan > (e.g. > "session:answer();")? > > Best Regards, > Jerry > > > Jerry, > > A Lua script that is explicitly called from the dialplan will indeed > execute dialplan-ish stuff. For example, let's say you had this in > conf/dialplan/default.xml: > > > > > > > > Then myluascript.lua has something like: > > --Sample Lua script > session:answer() > session:sleep(1000) > session:streamFile("/path/to/file.wav") > session:hangup() > > Assuming an otherwise default install, the above Lua script would > execute when a caller dialed 9876, or if a call was x-ferred to 9876. > > However, if you're wanting to use Lua to serve up a dialplan then it's > totally different. Lua is not called from the dialplan; Lua provides > the dialplan to FreeSWITCH. This latter case is the scenario discussed > in the wiki section you referenced. > (http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration) > > Are you trying to use Lua scripting for serving up a dynamic > configuration of some sort? > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Mon Nov 16 15:05:29 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 17 Nov 2009 10:05:29 +1100 Subject: [Freeswitch-users] TLS support on debian lenny Message-ID: <33c87fa30911161505y6b59312cm2d631dae65cb531d@mail.gmail.com> Hi! I am trying to enable SSL support in FS. I have followed the wiki at http://wiki.freeswitch.org/wiki/SIP_TLS I already had libssl-dev installed, so I thought support should already have been compiled into FS, however enabling Internal_ssl_enable=true in vars.xml results in FS internal profile to not start: 2009-11-17 09:31:48.593240 [NOTICE] sofia.c:3016 Started Profile internal [sofia_reg_internal] 2009-11-17 09:31:48.907740 [ERR] sofia.c:1006 Error Creating SIP UA for profile: internal Checking freeswitch/libs/sofia-sip/config.log I see the following, which I assume means TLS has not been compiled with support: configure:27892: checking openssl/tls1.h usability configure:27909: gcc -c -DSU_DEBUG=0 -g -ggdb conftest.c >&5 conftest.c:156:26: error: openssl/tls1.h: No such file or directory What package should I have installed prior to compiling FS on debian? There is no OpenSSL-Dev. Is it libcurl4-openssl-dev? Thanks From timuckun at gmail.com Mon Nov 16 15:07:18 2009 From: timuckun at gmail.com (Tim Uckun) Date: Tue, 17 Nov 2009 12:07:18 +1300 Subject: [Freeswitch-users] 1.05 Message-ID: <855e4dcf0911161507j4d03ed3fof9bed52926c6bcbf@mail.gmail.com> Where is 1.05? The trunk? Is trunk stable? Thanks. From msc at freeswitch.org Mon Nov 16 15:16:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 15:16:17 -0800 Subject: [Freeswitch-users] 1.05 In-Reply-To: <855e4dcf0911161507j4d03ed3fof9bed52926c6bcbf@mail.gmail.com> References: <855e4dcf0911161507j4d03ed3fof9bed52926c6bcbf@mail.gmail.com> Message-ID: <87f2f3b90911161516s485c055cyd85935295b2c8d32@mail.gmail.com> We are still working on 1.0.5. Right now the best place to be is that latest trunk. More information is forthcoming... -MC On Mon, Nov 16, 2009 at 3:07 PM, Tim Uckun wrote: > Where is 1.05? The trunk? Is trunk stable? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/e2b2bc30/attachment-0002.html From brian at freeswitch.org Mon Nov 16 15:22:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Nov 2009 17:22:15 -0600 Subject: [Freeswitch-users] 1.05 In-Reply-To: <855e4dcf0911161507j4d03ed3fof9bed52926c6bcbf@mail.gmail.com> References: <855e4dcf0911161507j4d03ed3fof9bed52926c6bcbf@mail.gmail.com> Message-ID: <38AA65CA-FCB4-443F-A249-E8CC187356A0@freeswitch.org> Tim, 1.0.5 is coming soon... We were ready to release on the tuesday morning we said but we woke up and Jira was flooded with tons of new issues half of which we asked for more info on and the reporters aren't responding. So the key is if you open a jira be ready to respond because I'm not going to chase people down anymore. On that note we need more people to help out on Jira... All it takes is asking questions and trying to reproduce things that are reported thats the hard part... Once something can be reproduced reliably we can fix it faster. Thanks, /b On Nov 16, 2009, at 5:07 PM, Tim Uckun wrote: > Where is 1.05? The trunk? Is trunk stable? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From tina at a2unlimited.com Mon Nov 16 15:22:01 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Mon, 16 Nov 2009 18:22:01 -0500 Subject: [Freeswitch-users] ESL: No matching function... Message-ID: <673dfcbcbab316d312ea4ae87d13418c.squirrel@emailmg.ipower.com> I have three FreeSWITCH servers currently setup with perl modules using ESL to send call instructions and monitor events. On two of the servers, my modules execute without error, but on a third, I keep getting the following error: No matching function for overloaded 'new_ESLconnection' at /usr/lib64/perl5/site_perl/5.8.8/x86_64-linux-thread-multi/ESL.pm line 116. Is this something in ESL that I'm doing wrong, or is it an issue related to the perl configuration on the server? As far as I can tell, the three servers have been setup the same. - Tina From msc at freeswitch.org Mon Nov 16 15:42:06 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 15:42:06 -0800 Subject: [Freeswitch-users] ESL: No matching function... In-Reply-To: <673dfcbcbab316d312ea4ae87d13418c.squirrel@emailmg.ipower.com> References: <673dfcbcbab316d312ea4ae87d13418c.squirrel@emailmg.ipower.com> Message-ID: <87f2f3b90911161542k4490dc0bx6218122c31728cb@mail.gmail.com> On Mon, Nov 16, 2009 at 3:22 PM, wrote: > I have three FreeSWITCH servers currently setup with perl modules using > ESL to send call instructions and monitor events. On two of the servers, > my modules execute without error, but on a third, I keep getting the > following error: > > No matching function for overloaded 'new_ESLconnection' at > /usr/lib64/perl5/site_perl/5.8.8/x86_64-linux-thread-multi/ESL.pm line > 116. > > Is this something in ESL that I'm doing wrong, or is it an issue related > to the perl configuration on the server? > > As far as I can tell, the three servers have been setup the same. > > - Tina > > Can you cd into libs/esl and do "make && make perlmod" and see if any errors show up? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/61ce07b3/attachment-0002.html From msc at freeswitch.org Mon Nov 16 16:25:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Nov 2009 16:25:09 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.0.5 Status Update Message-ID: <87f2f3b90911161625yb69f604i4b19f7f6a6a31800@mail.gmail.com> Hello folks! I just wanted to let everyone know that there is a status update on the main FreeSWITCH page. Here's a quick link for your convenience: http://bit.ly/3RSY9F The abridged version is this: we're working on it, there are some outstanding JIRA reports that people need to review & test, and we would appreciate more people updating to the latest trunk and making sure all is well. Thanks for helping! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091116/506f467d/attachment-0002.html From mcampbellsmith at gmail.com Mon Nov 16 19:15:35 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 17 Nov 2009 14:15:35 +1100 Subject: [Freeswitch-users] TLS support on debian lenny In-Reply-To: <33c87fa30911161505y6b59312cm2d631dae65cb531d@mail.gmail.com> References: <33c87fa30911161505y6b59312cm2d631dae65cb531d@mail.gmail.com> Message-ID: <33c87fa30911161915p3a2a082ame20d5427ea0bd4c4@mail.gmail.com> I installed libcurl4-openssl-dev, but this automatically removed libcurl4-gnutls-dev, which is required by mod_dingaling. Now mod_dingaling fails to build with: Compiling mod_dingaling.c ... mod_dingaling.c:309:78: error: macro "switch_odbc_handle_callback_exec" requires 5 arguments, but only 4 given mod_dingaling.c: In function ???mdl_execute_sql_callback???: mod_dingaling.c:309: error: ???switch_odbc_handle_callback_exec??? undeclared (first use in this function) mod_dingaling.c:309: error: (Each undeclared identifier is reported only once mod_dingaling.c:309: error: for each function it appears in.) make[6]: *** [mod_dingaling.lo] Error 1 Anyone know which package should be installed so that TLS works on Debian? On Tue, Nov 17, 2009 at 10:05 AM, Mark Campbell-Smith wrote: > Hi! > > I am trying to enable SSL support in FS. ?I have followed the wiki at > http://wiki.freeswitch.org/wiki/SIP_TLS > > I already had libssl-dev installed, so I thought support should > already have been compiled into FS, however enabling > Internal_ssl_enable=true in vars.xml results in FS internal profile to > not start: > > 2009-11-17 09:31:48.593240 [NOTICE] sofia.c:3016 Started Profile > internal [sofia_reg_internal] > 2009-11-17 09:31:48.907740 [ERR] sofia.c:1006 Error Creating SIP UA > for profile: internal > > Checking freeswitch/libs/sofia-sip/config.log I see the following, > which I assume means TLS has not been compiled with support: > configure:27892: checking openssl/tls1.h usability > configure:27909: gcc -c ?-DSU_DEBUG=0 -g -ggdb ?conftest.c >&5 > conftest.c:156:26: error: openssl/tls1.h: No such file or directory > > What package should I have installed prior to compiling FS on debian? > There is no OpenSSL-Dev. ?Is it libcurl4-openssl-dev? > > Thanks > From brian at freeswitch.org Mon Nov 16 19:24:43 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Nov 2009 21:24:43 -0600 Subject: [Freeswitch-users] TLS support on debian lenny In-Reply-To: <33c87fa30911161915p3a2a082ame20d5427ea0bd4c4@mail.gmail.com> References: <33c87fa30911161505y6b59312cm2d631dae65cb531d@mail.gmail.com> <33c87fa30911161915p3a2a082ame20d5427ea0bd4c4@mail.gmail.com> Message-ID: <2503E4FE-A1DB-486D-B818-CD9B04D3D955@freeswitch.org> Mark update and try again... we did some changes to the core odbc stuff today and waiting on the dust to settle. Thanks, Brian On Nov 16, 2009, at 9:15 PM, Mark Campbell-Smith wrote: > I installed libcurl4-openssl-dev, but this automatically removed > libcurl4-gnutls-dev, which is required by mod_dingaling. Now > mod_dingaling fails to build with: > Compiling mod_dingaling.c ... > mod_dingaling.c:309:78: error: macro > "switch_odbc_handle_callback_exec" requires 5 arguments, but only 4 > given > mod_dingaling.c: In function ???mdl_execute_sql_callback???: > mod_dingaling.c:309: error: ???switch_odbc_handle_callback_exec??? > undeclared (first use in this function) > mod_dingaling.c:309: error: (Each undeclared identifier is reported > only once > mod_dingaling.c:309: error: for each function it appears in.) > make[6]: *** [mod_dingaling.lo] Error 1 > > Anyone know which package should be installed so that TLS works on > Debian? > > On Tue, Nov 17, 2009 at 10:05 AM, Mark Campbell-Smith > wrote: >> Hi! >> >> I am trying to enable SSL support in FS. I have followed the wiki at >> http://wiki.freeswitch.org/wiki/SIP_TLS >> >> I already had libssl-dev installed, so I thought support should >> already have been compiled into FS, however enabling >> Internal_ssl_enable=true in vars.xml results in FS internal profile >> to >> not start: >> >> 2009-11-17 09:31:48.593240 [NOTICE] sofia.c:3016 Started Profile >> internal [sofia_reg_internal] >> 2009-11-17 09:31:48.907740 [ERR] sofia.c:1006 Error Creating SIP UA >> for profile: internal >> >> Checking freeswitch/libs/sofia-sip/config.log I see the following, >> which I assume means TLS has not been compiled with support: >> configure:27892: checking openssl/tls1.h usability >> configure:27909: gcc -c -DSU_DEBUG=0 -g -ggdb conftest.c >&5 >> conftest.c:156:26: error: openssl/tls1.h: No such file or directory >> >> What package should I have installed prior to compiling FS on debian? >> There is no OpenSSL-Dev. Is it libcurl4-openssl-dev? >> >> Thanks >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From elihayun at gmail.com Tue Nov 17 01:03:48 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 17 Nov 2009 11:03:48 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? Message-ID: <4B0266F4.8070602@savion.huji.ac.il> Hi We have more then one profile. To make a call I have to enter : bridge sofia/profile/number at ip The problem is when I use : "${use_profile}" I am getting the caller profile, and I need the destination profile. How do I get this information? Thanks Eli From dujinfang at gmail.com Tue Nov 17 04:17:15 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 17 Nov 2009 20:17:15 +0800 Subject: [Freeswitch-users] prefix Freeswitch-users vs. FreeSWITCH-Users Message-ID: <23f91030911170417y2e857124m5796565d5f24b329@mail.gmail.com> Would it be better to change the list subject prefix from [Freeswitch-users] to FreeSWITCH-Users? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/61b92fe8/attachment-0002.html From mattdfong at gmail.com Tue Nov 17 05:28:46 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 17 Nov 2009 20:28:46 +0700 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel Message-ID: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> I'm trying performing a uuid_record command immediately after a uuid_bridge, but receive a "Can not record session. Media not enabled on channel" error. proxy_media and bypass_media are both set to false. The uuid_record however works if I use sched_api +1 uuid_record... but if I do this, I of course loose the first second of conversation. Does anyone have any ideas on how I might be able to solve this? I've turned on DEBUG mode, but nothing out of the ordinary appears. http://pastebin.freeswitch.org/11141 --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/f0c222f1/attachment-0002.html From regs at kinetix.gr Tue Nov 17 05:42:01 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 17 Nov 2009 15:42:01 +0200 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp Message-ID: <4B02A829.7080708@kinetix.gr> I am trying to use switch_r_sdp to rewrite the SDP. The problem I am facing has to do with the way of doing it. Let's say I have: v=0 o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx s=Opal SIP Session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 5144 RTP/AVP 18 3 101 120 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:120 NSE/8000 a=fmtp:120 192-193 who to I set the switch_r_sdp variable in xml? Obviously this doesn't work : Do I have to escape any special characters and how? I tried using escaped quotes, escaped spaces, escaped tabs etc. Nothing worked. Any suggestions? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From nicolas at medularis.com Tue Nov 17 06:18:16 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 17 Nov 2009 11:18:16 -0300 Subject: [Freeswitch-users] need desperate help with zombie channels In-Reply-To: <27c25bc40911130513u405062c7kb775e14d04761fd4@mail.gmail.com> References: <27c25bc40911130513u405062c7kb775e14d04761fd4@mail.gmail.com> Message-ID: <1b46b4e80911170618x669312cdrc334067eb5c20ec4@mail.gmail.com> Hi Juan, A similar thing happened to me. I was creating channels and bridging them with a JS script. I had to add a session.hangup(); statement at the end of the script. That solved my problem. Cheers, Nico On Fri, Nov 13, 2009 at 10:13 AM, Juan Backson wrote: > Hi, > > I am having difficulty trying to figure out why there are bunch of zombie > channels in my system. It seems to me that these zombies come from > apr_thread pool. > > Does anyone have any idea what may be the cause of these problems? > > > freeswitch at internal> show channels > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure > b789468a-4412-490b-bc66-32f149ba4d1d,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 > ,,,XML,default,,,,, > 7e1ecaaa-b2d8-47a0-9982-25cd44186d4e,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_REPORTING,a88999001,a88999001,192.168.1.116,999100 at 192.168.1.116:9342 > ,,,XML,default,,,,, > 01fa2ff6-f807-4ef0-b988-70a9fe8c4536,outbound,2009-11-13 > 20:15:35,1258114535,sofia/external/999100 at 192.168.1.116:9342 > ,CS_EXCHANGE_MEDIA,a88999001,a88999001,192.168.1.116, > 999100 at 192.168.1.116:9342,incre_call_stat,125 165 182 235 13 3184093 > 0,XML,default,,,,, > 0271541f-f0b5-482c-b05d-b196f85121be,inbound,2009-11-13 > 20:15:35,1258114535,sofia/external/88999001 at 192.168.1.116:7342 > ,CS_EXECUTE,sipp,88999001,192.168.1.116,88999100,hangup,NORMAL_CLEARING,XML,default,,,,, > 7e4ccfec-a4ad-4817-9a82-f1166b34576f,outbound,2009-11-13 > 20:15:35,1258114536,sofia/external/999100 at 192.168.1.116:9342 > ,CS_CONSUME_MEDIA,a88999001,a88999001,192.168.1.116, > 999100 at 192.168.1.116:9342,,,XML,default,,,,, > > 5 total. > > freeswitch at internal> uuid_kill b789468a-4412-490b-bc66-32f149ba4d1d > -ERR No Such Channel! > > These channels actually do not exist in the system! > > > Here is my gcore output with 5 zombies out of 100K test calls : > > > Thread 21 (process 8946): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 128000} > #2 0x00002b3cb3bfb8ca in switch_console_loop () at > src/switch_console.c:819 > arg = 1 > thread = (switch_thread_t *) 0x2aaab00320d0 > thd_attr = (switch_threadattr_t *) 0x2aaab0032070 > pool = (switch_memory_pool_t *) 0x2aaab0031f88 > __func__ = "switch_console_loop" > __PRETTY_FUNCTION__ = "switch_console_loop" > #3 0x0000000000402884 in main (argc=1, argv=) > at src/switch.c:753 > pid_path = "/usr/local/freeswitch/log/freeswitch.pid", '\0' > > pid_buffer = "8946", '\0' > old_pid_buffer = '\0' > pid_len = 4 > old_pid_len = 4198811 > err = 0x2b3cb3cec77d "Success" > ---Type to continue, or q to quit--- > nf = 0 > runas_user = > runas_group = > nc = 0 > pid = > x = > opts = > opts_str = '\0' > local_argv = {0x7ffff6f08c15 "./freeswitch", 0x0 times>} > arg_argv = {0x0 } > alt_dirs = 0 > known_opt = > high_prio = 0 > flags = 65 > ret = > destroy_status = > fd = (switch_file_t *) 0xb6293e0 > pool = (switch_memory_pool_t *) 0xb629368 > rlp = {rlim_cur = 245760, rlim_max = 245760} > waste = 0 > __PRETTY_FUNCTION__ = "main" > > Thread 20 (process 20699): > ---Type to continue, or q to quit--- > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002aaaab35e926 in read_packet (listener=0x2aaae7523d08, > event=0x2aab3b5ab058, timeout=0) at mod_event_socket.c:1255 > do_sleep = 1 '\001' > mlen = 0 > bytes = 0 > mbuf = '\0' > buf = '\0' > len = 123 > status = SWITCH_STATUS_BREAK > count = > start = 1258117263 > pop = (void *) 0x2aaad12f6540 > ptr = 0x2aab3b5a98a0 "" > crcount = 0 '\0' > channel = (switch_channel_t *) 0x0 > clen = > __func__ = "read_packet" > __PRETTY_FUNCTION__ = "read_packet" > ---Type to continue, or q to quit--- > #3 0x00002aaaab36347a in listener_run (thread=, > obj=0x2aaae7523d08) at mod_event_socket.c:2093 > listener = (listener_t *) 0x0 > buf = '\0' > len = 1024 > status = > event = (switch_event_t *) 0x0 > reply = "\000OK log level [7]", '\0' > session = (switch_core_session_t *) 0x0 > channel = > revent = (switch_event_t *) 0x0 > var = > __PRETTY_FUNCTION__ = "listener_run" > __func__ = "listener_run" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 19 (process 14505): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaaf49798, > data=0x7afe0080) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( > thread=, obj=) > at src/switch_event.c:248 > pop = (void *) 0x0 > event = (switch_event_t *) 0x0 > queue = (switch_queue_t *) 0x2aaaaaf49798 > my_id = 1 > __func__ = "switch_event_dispatch_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 18 (process 9334): > #0 0x0000003054e0d2cb in read () from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3cd50c8 in read_char (el=0x2aaab0028180, cp=0x4027002f "") > at read.c:294 > num_read = 1076297860 > tried = 0 > ---Type to continue, or q to quit--- > #2 0x00002b3cb3cd4ceb in el_gets (el=0x2aaab0028180, nread=0x40270084) > at read.c:241 > cmdnum = 112 'p' > num = -1321754256 > ch = 0 '\0' > #3 0x00002b3cb3bfc4bb in console_thread (thread=, > obj=) at src/switch_console.c:464 > arg = 1 > count = 1 > line = 0x2aaab0034e70 "\n" > pool = (switch_memory_pool_t *) 0x2aaab0031f88 > __func__ = "console_thread" > __PRETTY_FUNCTION__ = "console_thread" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 17 (process 9333): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > ---Type to continue, or q to quit--- > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002b3cb3c53895 in softtimer_runtime () at src/switch_time.c:464 > current_ms = 692 > x = 690 > tick = 292 > ts = > last = 1258117283599783 > fwd_errs = 0 > rev_errs = 0 > __func__ = "softtimer_runtime" > #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x0, obj=0x0) > at src/switch_loadable_module.c:94 > status = > ts = (switch_core_thread_session_t *) 0x0 > module = (switch_loadable_module_t *) 0xb6c4e00 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 16 (process 9332): > ---Type to continue, or q to quit--- > #0 0x0000003054e0d4eb in accept () from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c707a4 in apr_socket_accept (new=0x416b4020, > sock=0xbcfde38, > connection_context=0x2aaacda27718) at network_io/unix/sockets.c:187 > No locals. > #2 0x00002aaaab35f889 in mod_event_socket_runtime () > at mod_event_socket.c:2324 > pool = (switch_memory_pool_t *) 0xbcfdc88 > listener_pool = (switch_memory_pool_t *) 0x2aaacda27718 > rv = > sa = (switch_sockaddr_t *) 0xbcfdd68 > inbound_socket = (switch_socket_t *) 0x2aaacda277f8 > listener = > x = > __func__ = "mod_event_socket_runtime" > #3 0x00002b3cb3c1a347 in switch_loadable_module_exec (thread=0x14f, > obj=0x2aaacda27948) at src/switch_loadable_module.c:94 > status = > ts = (switch_core_thread_session_t *) 0x2aaacda27948 > module = (switch_loadable_module_t *) 0x2aaaac0058c0 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #4 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > ---Type to continue, or q to quit--- > No symbol table info available. > #5 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 15 (process 9330): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 55000} > #2 0x00002aaab503cc4c in node_thread_run (thread=, > obj=) at mod_fifo.c:580 > val = (void *) 0x0 > var = (const void *) 0x0 > idle_consumers = > hi = (switch_hash_index_t *) 0x0 > ppl_waiting = 0 > consumer_total = 1087699264 > node = > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > ---Type to continue, or q to quit--- > > Thread 14 (process 9329): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100} > #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( > thread=, obj=) at sofia.c:763 > profile = (sofia_profile_t *) 0xbce2310 > ireg_loops = 18 > gateway_loops = 0 > loops = 72 > qsize = 4294966782 > pop = (void *) 0x0 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 13 (process 9328): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > ---Type to continue, or q to quit--- > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 0} > #2 0x00002aaab44d77be in sofia_profile_worker_thread_run ( > thread=, obj=) at sofia.c:763 > profile = (sofia_profile_t *) 0x2aaab000eb10 > ireg_loops = 5 > gateway_loops = 0 > loops = 93 > qsize = 4294966782 > pop = (void *) 0x0 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 12 (process 9327): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce71c0, > tout=1000) > at su_epoll_port.c:495 > ---Type to continue, or q to quit--- > j = 198076976 > n = 0 > events = 0 > index = 10922 > version = 3 > M = 4 > ev = 0x41204ef0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce71c0) > at su_base_port.c:349 > tout = 1000 > tout2 = 0 > __PRETTY_FUNCTION__ = "su_base_port_run" > #3 0x00002aaab45c6c51 in su_port_run (self=0xbce71c0) at su_port.h:326 > base = (su_virtual_port_t *) 0xbce71c0 > #4 0x00002aaab45c6c29 in su_root_run (self=0xbce72a0) at su_root.c:819 > __PRETTY_FUNCTION__ = "su_root_run" > #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404f7ac0) > at su_pthread_port.c:324 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0xbce71c0, sut_root = 0xbce72a0}} > zap = 1 > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > ---Type to continue, or q to quit--- > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 11 (process 9326): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce78b0, > tout=1000) > at su_epoll_port.c:495 > j = -1342070512 > n = 10922 > events = 0 > index = 10922 > version = 3 > M = 4 > ev = 0x411c8ef0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d1079 in su_base_port_run (self=0xbce78b0) > at su_base_port.c:349 > tout = 1000 > tout2 = 0 > __PRETTY_FUNCTION__ = "su_base_port_run" > #3 0x00002aaab45c6c51 in su_port_run (self=0xbce78b0) at su_port.h:326 > ---Type to continue, or q to quit--- > base = (su_virtual_port_t *) 0xbce78b0 > #4 0x00002aaab45c6c29 in su_root_run (self=0x2aaab001a060) at > su_root.c:819 > __PRETTY_FUNCTION__ = "su_root_run" > #5 0x00002aaab45d8d58 in su_pthread_port_clone_main (varg=0x404bbac0) > at su_pthread_port.c:324 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0xbce78b0, sut_root = 0x2aaab001a060}} > zap = 1 > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 10 (process 9325): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbce6c30, > tout=1000) > at su_epoll_port.c:495 > j = -1268971119 > n = 10922 > events = 0 > index = 0 > version = 1 > ---Type to continue, or q to quit--- > M = 4 > ev = 0x404f7c40 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbce6c30, tout=1000) > at su_base_port.c:467 > now = {tv_sec = 3467106082, tv_usec = 971475} > __PRETTY_FUNCTION__ = "su_base_port_step" > #3 0x00002aaab45c6d6a in su_port_step (self=0xbce6c30, tout=1000) > at su_port.h:340 > base = (su_virtual_port_t *) 0xbce6c30 > #4 0x00002aaab45c6d32 in su_root_step (self=0xbce4650, tout=1000) > at su_root.c:858 > __PRETTY_FUNCTION__ = "su_root_step" > #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( > thread=, obj=) at sofia.c:973 > profile = (sofia_profile_t *) 0xbce2310 > pool = > node = (sip_alias_node_t *) 0x0 > s_event = (switch_event_t *) 0x0 > sanity = > worker_thread = (switch_thread_t *) 0xbce36a0 > st = SWITCH_STATUS_SUCCESS > __func__ = "sofia_profile_thread_run" > ---Type to continue, or q to quit--- > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 9 (process 9324): > #0 0x00000030542d3368 in epoll_wait () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002aaab45c9c9c in su_epoll_port_wait_events (self=0xbcdffb0, > tout=1000) > at su_epoll_port.c:495 > j = -1268971119 > n = 10922 > events = 0 > index = 0 > version = 1 > M = 4 > ev = 0x404bbc40 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #2 0x00002aaab45d11d4 in su_base_port_step (self=0xbcdffb0, tout=1000) > at su_base_port.c:467 > now = {tv_sec = 3467106083, tv_usec = 525146} > __PRETTY_FUNCTION__ = "su_base_port_step" > ---Type to continue, or q to quit--- > #3 0x00002aaab45c6d6a in su_port_step (self=0xbcdffb0, tout=1000) > at su_port.h:340 > base = (su_virtual_port_t *) 0xbcdffb0 > #4 0x00002aaab45c6d32 in su_root_step (self=0xbcdfe00, tout=1000) > at su_root.c:858 > __PRETTY_FUNCTION__ = "su_root_step" > #5 0x00002aaab44e5c3a in sofia_profile_thread_run ( > thread=, obj=) at sofia.c:973 > profile = (sofia_profile_t *) 0x2aaab000eb10 > pool = > node = (sip_alias_node_t *) 0x0 > s_event = (switch_event_t *) 0x0 > sanity = > worker_thread = (switch_thread_t *) 0x2aaab000fea0 > st = SWITCH_STATUS_SUCCESS > __func__ = "sofia_profile_thread_run" > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #6 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #7 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 8 (process 8999): > ---Type to continue, or q to quit--- > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 444000} > #2 0x00002b3cb3c14e2a in switch_scheduler_task_thread ( > thread=, obj=) > at src/switch_scheduler.c:171 > __func__ = "switch_scheduler_task_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 7 (process 8998): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100} > #2 0x00002b3cb3c054f5 in switch_core_sql_thread ( > thread=, obj=) > at src/switch_core_sqldb.c:220 > ---Type to continue, or q to quit--- > pop = (void *) 0x2aaabf3d6220 > itterations = 0 > trans = 0 '\0' > nothing_in_queue = 1 '\001' > len = 100 > sql_len = 4844546 > sqlbuf = 0x2aab135c7010 "" > sql = > newlen = > lc = 0 > __PRETTY_FUNCTION__ = "switch_core_sql_thread" > __func__ = "switch_core_sql_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 6 (process 8995): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb64c158, data=0x40893088) > at misc/apr_queue.c:276 > ---Type to continue, or q to quit--- > rv = 0 > #2 0x00002b3cb3c48ff1 in log_thread (t=, > obj=) at src/switch_log.c:288 > pop = (void *) 0x0 > node = (switch_log_node_t *) 0x0 > binding = (switch_log_binding_t *) 0x0 > __func__ = "log_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 5 (process 8951): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaac355a8, > data=0x40bec070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > ---Type to continue, or q to quit--- > queue = (switch_queue_t *) 0x2aaaaac355a8 > index = 0 > my_id = 2 > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 4 (process 8950): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaab705a8, > data=0x4060a070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > queue = (switch_queue_t *) 0x2aaaaab705a8 > index = 0 > my_id = 1 > ---Type to continue, or q to quit--- > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 3 (process 8949): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0xb638fa8, data=0x405ce070) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c1fb14 in switch_event_thread (thread= out>, > obj=) at src/switch_event.c:291 > pop = (void *) 0x0 > event = > queue = (switch_queue_t *) 0xb638fa8 > index = 0 > my_id = 0 > __func__ = "switch_event_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > Thread 2 (process 8948): > #0 0x0000003054e0a899 in pthread_cond_wait@@GLIBC_2.3.2 () > from /lib64/libpthread.so.0 > No symbol table info available. > #1 0x00002b3cb3c63b42 in apr_queue_pop (queue=0x2aaaaacfa5a8, > data=0x40592080) > at misc/apr_queue.c:276 > rv = 0 > #2 0x00002b3cb3c206be in switch_event_dispatch_thread ( > thread=, obj=) > at src/switch_event.c:248 > pop = (void *) 0x0 > event = (switch_event_t *) 0x0 > queue = (switch_queue_t *) 0x2aaaaacfa5a8 > my_id = 0 > __func__ = "switch_event_dispatch_thread" > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > > ---Type to continue, or q to quit--- > Thread 1 (process 8947): > #0 0x00000030542cc4c2 in select () from /lib64/libc.so.6 > No symbol table info available. > #1 0x00002b3cb3c72df5 in apr_sleep (t=) > at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 451000} > #2 0x00002b3cb3c00c95 in pool_thread (thread=, > obj=) at src/switch_core_memory.c:490 > x = > #3 0x0000003054e06367 in start_thread () from /lib64/libpthread.so.0 > No symbol table info available. > #4 0x00000030542d2f7d in clone () from /lib64/libc.so.6 > No symbol table info available. > (gdb) > (gdb) > (gdb) > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/75b3fb79/attachment-0002.html From brian at freeswitch.org Tue Nov 17 06:41:19 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 08:41:19 -0600 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp In-Reply-To: <4B02A829.7080708@kinetix.gr> References: <4B02A829.7080708@kinetix.gr> Message-ID: <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> Why are you needing to rewrite it? /b On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: > > I am trying to use switch_r_sdp to rewrite the SDP. > The problem I am facing has to do with the way of doing it. > > Let's say I have: > > v=0 > o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx > s=Opal SIP Session > c=IN IP4 xxx.xxx.xxx.xxx > t=0 0 > m=audio 5144 RTP/AVP 18 3 101 120 > c=IN IP4 xxx.xxx.xxx.xxx > a=rtpmap:18 G729/8000/1 > a=fmtp:18 annexb=no > a=rtpmap:3 gsm/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16,32,36 > a=rtpmap:120 NSE/8000 > a=fmtp:120 192-193 > > who to I set the switch_r_sdp variable in xml? > > Obviously this doesn't work : > > > > Do I have to escape any special characters and how? > I tried using escaped quotes, escaped spaces, escaped tabs etc. > Nothing worked. > > Any suggestions? > > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Tue Nov 17 06:43:15 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 08:43:15 -0600 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: <4B0266F4.8070602@savion.huji.ac.il> References: <4B0266F4.8070602@savion.huji.ac.il> Message-ID: Why do you need to know the destination profile like that? You get to pick that on your own so you should already know that before hand. /b On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote: > Hi > We have more then one profile. To make a call I have to enter : bridge > sofia/profile/number at ip > The problem is when I use : "${use_profile}" I am getting the caller > profile, and I need the destination profile. > > How do I get this information? > > Thanks > > Eli From abeka at greatiam.com Tue Nov 17 07:12:19 2009 From: abeka at greatiam.com (Sam Abekah-Mensah) Date: Tue, 17 Nov 2009 15:12:19 +0000 Subject: [Freeswitch-users] Registration Error - 408 timeout and now 403 In-Reply-To: <4B01A8AE.7070708@greatiam.com> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> <4B01A8AE.7070708@greatiam.com> Message-ID: <4B02BD53.5040203@greatiam.com> Hello I have tried the same setup but this time using a windows build FS1.0.4 on an XP machine and all is fine. The sample 1001 and 1002 IDs work without any tweaking at all. Could the problem be with the linux build 1.0.4.? I am running on an FC11 machine. On the FC11 box I used the svn link to build using the ff: bootstarp.sh configure withoout libcurl to eliinate the spidermonkey lib error make make install Did I miss anything ? In a nutshell I can get the sample test to work using a windows-based version Thanks d Samuel 'Otis' Abekah-Mensah wrote: >
Hello > > thanks so much. The machines are on the same lan , 2 have static IP > with one on DHCP just for variation . I do get there errors on stating FS > 1. Error stacksize too large 4194303 offers advise to run ./freeswitch > -wate > 2. Error checking for PMP [GENERAL ERROR] > and > 3. [WARNING] sofia_reg.c:1788: Can't register a pointer > I do not know if any of this is could help The 2 boxes I run X-lite > from are windows 2k service pack 4 > > Oh I ahve had a go and I am now getting Error 403 - Forbidden on the > Xlite clients side. > > I have also tried using Zoiper but it seems to register but then comes > up with an error "bearercapability " > > Thanks for your time, Michael and may thanks Brian. I am not sure if > the iptables bit has caused the change from error 408 to error 403. > > Thanks; I apperecitae your help > . > > > > Michael Collins wrote: >> >> >> On Sat, Nov 14, 2009 at 5:18 PM, Samuel Abekah-Mensah >> > wrote: >> >> Hello >> >> Please pardon me if the solution to this is somewhere already that I >> have been unable to locate. I have just got a straight >> out-of-the-box >> build of FS. According to the wiki, I should be able to test using >> user >> IDs 1001 and 1002. However, I am get the above error. If I, >> however, >> un-tick register with domain I do net get the error but does not >> communicate either. Is there a conf that I should have done ? >> >> I am using X-lite3 >> >> >> Is NAT involved or are the x-lite clients on the same LAN? Also, you >> might want to turn on a SIP trace at the console to see if there are >> any clues. >> -MC >> > > > >
> From jerry.richards at teotech.com Tue Nov 17 08:22:36 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 17 Nov 2009 08:22:36 -0800 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com><1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl><1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> Message-ID: MC, We would like the dialplan to route the call based on Presence, which is a database lookup. I should be able to do this in Lua, true? Jerry _____ From: Michael Collins [mailto:msc at freeswitch.org] Sent: Monday, November 16, 2009 11:33 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Accessing Config Info From Database On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards wrote: I have a bit of confusion about Lua scripting. When a script is invoked, should it always return an XML string that is used by FS? Or as in the case of dialplan examples, does it actually execute the dialplan (e.g. "session:answer();")? Best Regards, Jerry Jerry, A Lua script that is explicitly called from the dialplan will indeed execute dialplan-ish stuff. For example, let's say you had this in conf/dialplan/default.xml: Then myluascript.lua has something like: --Sample Lua script session:answer() session:sleep(1000) session:streamFile("/path/to/file.wav") session:hangup() Assuming an otherwise default install, the above Lua script would execute when a caller dialed 9876, or if a call was x-ferred to 9876. However, if you're wanting to use Lua to serve up a dialplan then it's totally different. Lua is not called from the dialplan; Lua provides the dialplan to FreeSWITCH. This latter case is the scenario discussed in the wiki section you referenced. (http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration) Are you trying to use Lua scripting for serving up a dynamic configuration of some sort? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/acf65320/attachment-0002.html From mrene_lists at avgs.ca Tue Nov 17 08:43:12 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 08:43:12 -0800 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel In-Reply-To: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> References: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> Message-ID: You can't record until media is present. You could trigger it with execute_on_answer and the record_session application Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: > I'm trying performing a uuid_record command immediately after a > uuid_bridge, but receive a "Can not record session. Media not > enabled on channel" error. proxy_media and bypass_media are both set > to false. > > The uuid_record however works if I use sched_api +1 uuid_record... > but if I do this, I of course loose the first second of conversation. > > Does anyone have any ideas on how I might be able to solve this? > I've turned on DEBUG mode, but nothing out of the ordinary appears. > > http://pastebin.freeswitch.org/11141 > > --matt > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/c460ef84/attachment-0002.html From yehavi.bourvine at gmail.com Tue Nov 17 08:51:00 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 17 Nov 2009 18:51:00 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: References: <4B0266F4.8070602@savion.huji.ac.il> Message-ID: Hello Brian, the situation is as follows: Our PBX machine has more than one interface, each one has a profile. Some phones are registered via one interface and tje others on the other. The call should be sent usinbg the profile of the destination as if not, the IP address of the server in the SIP message is incorrect (the other interface) thus the phone cannot answer. When a call is processed you know the originator profile name; we need also the destination profile name... Thanks! __yehavi: 2009/11/17 Brian West > Why do you need to know the destination profile like that? You get to > pick that on your own so you should already know that before hand. > > > /b > > On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote: > > > Hi > > We have more then one profile. To make a call I have to enter : bridge > > sofia/profile/number at ip > > The problem is when I use : "${use_profile}" I am getting the caller > > profile, and I need the destination profile. > > > > How do I get this information? > > > > Thanks > > > > Eli > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/9461e9bc/attachment-0002.html From mattdfong at gmail.com Tue Nov 17 08:57:54 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 17 Nov 2009 23:57:54 +0700 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel In-Reply-To: References: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> Message-ID: <4256bf830911170857o302b6f41i998cbe974e4d4008@mail.gmail.com> The media should be there, when I uuid_bridge both sessions are parked and should have already had media sent. I'm using ignore_early_media=true --matt On Tue, Nov 17, 2009 at 11:43 PM, Mathieu Rene wrote: > You can't record until media is present. You could trigger it with > execute_on_answer and the record_session application > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: > > I'm trying performing a uuid_record command immediately after a > uuid_bridge, but receive a "Can not record session. Media not enabled on > channel" error. proxy_media and bypass_media are both set to false. > > The uuid_record however works if I use sched_api +1 uuid_record... but if I > do this, I of course loose the first second of conversation. > > Does anyone have any ideas on how I might be able to solve this? I've > turned on DEBUG mode, but nothing out of the ordinary appears. > > http://pastebin.freeswitch.org/11141 > > --matt > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/dac32fc8/attachment-0002.html From krzysiez at go2.pl Tue Nov 17 06:05:07 2009 From: krzysiez at go2.pl (=?UTF-8?Q?Christopher_Z.?=) Date: Tue, 17 Nov 2009 15:05:07 +0100 Subject: [Freeswitch-users] =?utf-8?q?Compilation_problem?= Message-ID: <32c1b333.68a1501e.4b02ad93.bbfcd@go2.pl> Hi, I've got this error after make: http://pastebin.freeswitch.org/11145 Any idea how to fix this error ? Thanks. From krzysiez at go2.pl Tue Nov 17 09:18:32 2009 From: krzysiez at go2.pl (=?UTF-8?Q?Krzysztof_Zimnicki?=) Date: Tue, 17 Nov 2009 18:18:32 +0100 Subject: [Freeswitch-users] =?utf-8?q?=5Bfreeswitch-users=5DCompilation_pr?= =?utf-8?q?oblem?= Message-ID: <3201712c.1d4430a1.4b02dae8.49cdb@go2.pl> Hi, I've got this error after make: http://pastebin.freeswitch.org/11145 Any idea how to fix this error ? Thanks. From freeswitch-users-list at metik.com Tue Nov 17 09:32:05 2009 From: freeswitch-users-list at metik.com (Metik) Date: Tue, 17 Nov 2009 12:32:05 -0500 Subject: [Freeswitch-users] Using uuid_transfer with uuid_hold Message-ID: <4B02DE15.3090207@metik.com> Using the API, any caveats with transferring a call (via uuid_transfer) that has been placed on hold (via uuid_hold) without using "uuild_hold off" before doing so? Is it even possible? -metik From mrene_lists at avgs.ca Tue Nov 17 09:32:52 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 09:32:52 -0800 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel In-Reply-To: <4256bf830911170857o302b6f41i998cbe974e4d4008@mail.gmail.com> References: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> <4256bf830911170857o302b6f41i998cbe974e4d4008@mail.gmail.com> Message-ID: Ah I see what happens, switch_ivr_uuid_bridge will reset the session's read codec (which will be re-initialized as soon as the actual bridge takes place) and uuid_record will think it hasnt been initialized yet. You can set the following vars to execute an application right before bridge starts exchanging audio bridge_pre_execute_aleg_app bridge_pre_execute_aleg_arg Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 8:57 AM, Matthew Fong wrote: > The media should be there, when I uuid_bridge both sessions are > parked and should have already had media sent. I'm using > ignore_early_media=true > > --matt > > On Tue, Nov 17, 2009 at 11:43 PM, Mathieu Rene > wrote: > You can't record until media is present. You could trigger it with > execute_on_answer and the record_session application > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: > >> I'm trying performing a uuid_record command immediately after a >> uuid_bridge, but receive a "Can not record session. Media not >> enabled on channel" error. proxy_media and bypass_media are both >> set to false. >> >> The uuid_record however works if I use sched_api +1 uuid_record... >> but if I do this, I of course loose the first second of conversation. >> >> Does anyone have any ideas on how I might be able to solve this? >> I've turned on DEBUG mode, but nothing out of the ordinary appears. >> >> http://pastebin.freeswitch.org/11141 >> >> --matt >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/f4063684/attachment-0002.html From mrene_lists at avgs.ca Tue Nov 17 09:35:14 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 09:35:14 -0800 Subject: [Freeswitch-users] Using uuid_transfer with uuid_hold In-Reply-To: <4B02DE15.3090207@metik.com> References: <4B02DE15.3090207@metik.com> Message-ID: Shouldnt be a problem, but I think you really want to uuid_park it, not hold it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 9:32 AM, Metik wrote: > Using the API, any caveats with transferring a call (via > uuid_transfer) > that has been placed on hold (via uuid_hold) without using "uuild_hold > off" before doing so? Is it even possible? > > -metik > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Nov 17 09:41:49 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 11:41:49 -0600 Subject: [Freeswitch-users] Using uuid_transfer with uuid_hold In-Reply-To: References: <4B02DE15.3090207@metik.com> Message-ID: <44ABEBCF-FD9C-41C7-9AC1-9FC94FFD509C@freeswitch.org> uuid_hold will send a HOLD indication to the end you're talking to ... it will NOT put the person your talking to on HOLD... I think that is the confusion of uuid_hold. Example: Phone -> FS1 -> FS2(uses uuid_hold) will cause the FS1 box to play hold music to the phone. /b On Nov 17, 2009, at 11:35 AM, Mathieu Rene wrote: > Shouldnt be a problem, but I think you really want to uuid_park it, > not hold it. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/2020cc30/attachment-0002.html From brian at freeswitch.org Tue Nov 17 09:42:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 11:42:35 -0600 Subject: [Freeswitch-users] [freeswitch-users]Compilation problem In-Reply-To: <3201712c.1d4430a1.4b02dae8.49cdb@go2.pl> References: <3201712c.1d4430a1.4b02dae8.49cdb@go2.pl> Message-ID: <68899D9C-6D0C-4576-B6CA-68C04A013C79@freeswitch.org> You 'make current' and stop cross posting.. /b On Nov 17, 2009, at 11:18 AM, Krzysztof Zimnicki wrote: > Hi, > > I've got this error after make: > > http://pastebin.freeswitch.org/11145 > > Any idea how to fix this error ? > > Thanks. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mattdfong at gmail.com Tue Nov 17 09:51:52 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 18 Nov 2009 00:51:52 +0700 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel In-Reply-To: References: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> <4256bf830911170857o302b6f41i998cbe974e4d4008@mail.gmail.com> Message-ID: <4256bf830911170951x643ae03egce257f13982b7af6@mail.gmail.com> Hi Mathieu, This makes sense! Thanks. Since bypass_proxy = false and proxy_media = false, why is it trying to renegotiate a codec on a uuid_bridge? --matt On Wed, Nov 18, 2009 at 12:32 AM, Mathieu Rene wrote: > Ah I see what happens, switch_ivr_uuid_bridge will reset the session's read > codec (which will be re-initialized as soon as the actual bridge takes > place) and uuid_record will think it hasnt been initialized yet. > > You can set the following vars to execute an application right before > bridge starts exchanging audio > bridge_pre_execute_aleg_app > bridge_pre_execute_aleg_arg > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 8:57 AM, Matthew Fong wrote: > > The media should be there, when I uuid_bridge both sessions are parked and > should have already had media sent. I'm using ignore_early_media=true > > --matt > > On Tue, Nov 17, 2009 at 11:43 PM, Mathieu Rene wrote: > >> You can't record until media is present. You could trigger it with >> execute_on_answer and the record_session application >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: >> >> I'm trying performing a uuid_record command immediately after a >> uuid_bridge, but receive a "Can not record session. Media not enabled on >> channel" error. proxy_media and bypass_media are both set to false. >> >> The uuid_record however works if I use sched_api +1 uuid_record... but if >> I do this, I of course loose the first second of conversation. >> >> Does anyone have any ideas on how I might be able to solve this? I've >> turned on DEBUG mode, but nothing out of the ordinary appears. >> >> http://pastebin.freeswitch.org/11141 >> >> --matt >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/c2605d2d/attachment-0002.html From mrene_lists at avgs.ca Tue Nov 17 09:54:11 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 09:54:11 -0800 Subject: [Freeswitch-users] uuid_record immediatly after uuid_bridge - Can not record session. Media not enabled on channel In-Reply-To: <4256bf830911170951x643ae03egce257f13982b7af6@mail.gmail.com> References: <4256bf830911170528k2fb922efm627a2766728d4462@mail.gmail.com> <4256bf830911170857o302b6f41i998cbe974e4d4008@mail.gmail.com> <4256bf830911170951x643ae03egce257f13982b7af6@mail.gmail.com> Message-ID: Its not re-negotiating it, its re-initializing it so the codec's internal state gets reset. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 9:51 AM, Matthew Fong wrote: > Hi Mathieu, > > This makes sense! Thanks. > > Since bypass_proxy = false and proxy_media = false, why is it trying > to renegotiate a codec on a uuid_bridge? > > --matt > > On Wed, Nov 18, 2009 at 12:32 AM, Mathieu Rene > wrote: > Ah I see what happens, switch_ivr_uuid_bridge will reset the > session's read codec (which will be re-initialized as soon as the > actual bridge takes place) and uuid_record will think it hasnt been > initialized yet. > > You can set the following vars to execute an application right > before bridge starts exchanging audio > bridge_pre_execute_aleg_app > bridge_pre_execute_aleg_arg > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 8:57 AM, Matthew Fong wrote: > >> The media should be there, when I uuid_bridge both sessions are >> parked and should have already had media sent. I'm using >> ignore_early_media=true >> >> --matt >> >> On Tue, Nov 17, 2009 at 11:43 PM, Mathieu Rene >> wrote: >> You can't record until media is present. You could trigger it with >> execute_on_answer and the record_session application >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 17-Nov-09, at 5:28 AM, Matthew Fong wrote: >> >>> I'm trying performing a uuid_record command immediately after a >>> uuid_bridge, but receive a "Can not record session. Media not >>> enabled on channel" error. proxy_media and bypass_media are both >>> set to false. >>> >>> The uuid_record however works if I use sched_api +1 uuid_record... >>> but if I do this, I of course loose the first second of >>> conversation. >>> >>> Does anyone have any ideas on how I might be able to solve this? >>> I've turned on DEBUG mode, but nothing out of the ordinary appears. >>> >>> http://pastebin.freeswitch.org/11141 >>> >>> --matt >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/36d677c6/attachment-0002.html From freeswitch-users-list at metik.com Tue Nov 17 10:03:03 2009 From: freeswitch-users-list at metik.com (Metik) Date: Tue, 17 Nov 2009 13:03:03 -0500 Subject: [Freeswitch-users] Using uuid_transfer with uuid_hold In-Reply-To: <44ABEBCF-FD9C-41C7-9AC1-9FC94FFD509C@freeswitch.org> References: <4B02DE15.3090207@metik.com> <44ABEBCF-FD9C-41C7-9AC1-9FC94FFD509C@freeswitch.org> Message-ID: <4B02E557.8050608@metik.com> Brian, That explains what I have been seeing... The console would freeze or the xml-rpc request would never receive a response (trunk rev 15463). -metik Brian West wrote: > uuid_hold will send a HOLD indication to the end you're talking to ... > it will NOT put the person your talking to on HOLD... I think that is > the confusion of uuid_hold. > > > Example: > > Phone -> FS1 -> FS2(uses uuid_hold) will cause the FS1 box to play > hold music to the phone. > > /b > > > On Nov 17, 2009, at 11:35 AM, Mathieu Rene wrote: > >> Shouldnt be a problem, but I think you really want to uuid_park it, >> not hold it. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From tina at a2unlimited.com Tue Nov 17 10:04:23 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Tue, 17 Nov 2009 13:04:23 -0500 Subject: [Freeswitch-users] ESL: No matching function... In-Reply-To: References: Message-ID: MC, Yes, I tried "make && make perlmod", which did not fix the error. Just finished deploying an instance of the application on another server that did not produce the error (exact same configuration). Not sure what is causing it, or how to fix it. Bizarre. - Tina > On Mon, Nov 16, 2009 at 3:22 PM, wrote: > >> I have three FreeSWITCH servers currently setup with perl modules using >> ESL to send call instructions and monitor events. On two of the >> servers, >> my modules execute without error, but on a third, I keep getting the >> following error: >> >> No matching function for overloaded 'new_ESLconnection' at >> /usr/lib64/perl5/site_perl/5.8.8/x86_64-linux-thread-multi/ESL.pm line >> 116. >> >> Is this something in ESL that I'm doing wrong, or is it an issue related >> to the perl configuration on the server? >> >> As far as I can tell, the three servers have been setup the same. >> >> - Tina >> >> > Can you cd into libs/esl and do "make && make perlmod" and see if any > errors > show up? > -MC > From info at daccii.it Tue Nov 17 10:48:30 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Tue, 17 Nov 2009 19:48:30 +0100 Subject: [Freeswitch-users] Help testing a new startskype.sh script Message-ID: <4B02EFFE.1030705@daccii.it> Hi, this morning i've started to work on a new startskype.sh script, for mod_skypiax and, finally, it works as it should! I've done some preliminary testing, can someone help me to test it better with many users? Here jira report http://jira.freeswitch.org/browse/MODSKYPIAX-59 Thank you Best Regards, Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/59bd11dc/attachment-0002.vcf From msc at freeswitch.org Tue Nov 17 10:56:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Nov 2009 10:56:58 -0800 Subject: [Freeswitch-users] ESL: No matching function... In-Reply-To: References: Message-ID: <87f2f3b90911171056j5468436bqa765831665109bec@mail.gmail.com> On Tue, Nov 17, 2009 at 10:04 AM, wrote: > MC, > > Yes, I tried "make && make perlmod", which did not fix the error. > > Just finished deploying an instance of the application on another server > that did not produce the error (exact same configuration). > > Not sure what is causing it, or how to fix it. > > Bizarre. > > - Tina > > Is the offending machine a 32 or 64 bit machine? Just curious if there is something physically different about this machine than the others. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/48d02983/attachment-0002.html From msc at freeswitch.org Tue Nov 17 11:02:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Nov 2009 11:02:35 -0800 Subject: [Freeswitch-users] Accessing Config Info From Database In-Reply-To: References: <9478A66A6D6048BD977C80B34F766085@greyhawk.tonecommander.com> <1258151335.15402.16.camel@desk.bofh.scarlet-internet.nl> <1FDE686D97124E6A8D8D0C2F16ED4D74@greyhawk.tonecommander.com> <87f2f3b90911161133o552cc1d6xabb52222d1ddb371@mail.gmail.com> Message-ID: <87f2f3b90911171102s553de3b8lb1b8c36155d9fd51@mail.gmail.com> On Tue, Nov 17, 2009 at 8:22 AM, Jerry Richards wrote: > MC, > > We would like the dialplan to route the call based on Presence, which is a > database lookup. I should be able to do this in Lua, true? > > Jerry > > Yes, you can use Lua for this if you wish to do so, HOWEVER, luasql has a bug so tread carefully. Check with Chad/hunmonk for thoughts on doing db lookups from Lua. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/ca588669/attachment-0002.html From msc at freeswitch.org Tue Nov 17 11:05:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Nov 2009 11:05:16 -0800 Subject: [Freeswitch-users] Registration Error - 408 timeout and now 403 In-Reply-To: <4B02BD53.5040203@greatiam.com> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> <4B01A8AE.7070708@greatiam.com> <4B02BD53.5040203@greatiam.com> Message-ID: <87f2f3b90911171105s7fb2fea3l316fc2777cbc051a@mail.gmail.com> Try doing this: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install -MC On Tue, Nov 17, 2009 at 7:12 AM, Sam Abekah-Mensah wrote: > Hello > > I have tried the same setup but this time using a windows build FS1.0.4 > on an XP machine and all is fine. The sample 1001 and 1002 IDs work > without any tweaking at all. Could the problem be with the linux build > 1.0.4.? I am running on an FC11 machine. > > On the FC11 box I used the svn link to build using the ff: > > bootstarp.sh > configure withoout libcurl to eliinate the spidermonkey lib error > make > make install > > Did I miss anything ? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/6dbbd142/attachment-0002.html From kristian.kielhofner at gmail.com Tue Nov 17 11:21:01 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Nov 2009 14:21:01 -0500 Subject: [Freeswitch-users] Build FS without spandsp or libtiff Message-ID: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> Hello everyone, I'm trying to keep my build as small as possible (for AstLinux). Is there anyway to build without spandsp or libtiff? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Tue Nov 17 11:24:23 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 13:24:23 -0600 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> Message-ID: Don't build mod_fax /b On Nov 17, 2009, at 1:21 PM, Kristian Kielhofner wrote: > Hello everyone, > > I'm trying to keep my build as small as possible (for AstLinux). Is > there anyway to build without spandsp or libtiff? > > Thanks! > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/bee1c3cb/attachment-0002.html From kristian.kielhofner at gmail.com Tue Nov 17 11:33:33 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Nov 2009 14:33:33 -0500 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> Message-ID: <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> I'm not... On Tue, Nov 17, 2009 at 2:24 PM, Brian West wrote: > Don't build mod_fax > /b -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Tue Nov 17 11:37:45 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Nov 2009 13:37:45 -0600 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> Message-ID: <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> OH you need spandsp for VoipCodecs. No way around that one. /b On Nov 17, 2009, at 1:33 PM, Kristian Kielhofner wrote: > I'm not... > > On Tue, Nov 17, 2009 at 2:24 PM, Brian West > wrote: >> Don't build mod_fax >> /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/66db7e27/attachment-0002.html From regs at kinetix.gr Tue Nov 17 11:44:48 2009 From: regs at kinetix.gr (regs at kinetix.gr) Date: Tue, 17 Nov 2009 21:44:48 +0200 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp In-Reply-To: <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> References: <4B02A829.7080708@kinetix.gr> <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> Message-ID: <4B02FD30.8050502@kinetix.gr> I am trying to achieve something similar to that : http://wiki.freeswitch.org/wiki/Codec_negotiation#Modifying_the_codec_when_using_proxy_media_mode but I am using xml_curl to create the dialplan (i.e. the web server that serves the dialplan makes the decision about the SDP). So I need a way to write the new SDP in the XML dialplan response. However, in the above example due to the regex manipulation the user is not facing the problem that I am with setting the switch_r_sdp to a complex value that contains =, spaces, new lines etc. Brian West wrote: > Why are you needing to rewrite it? > > /b > > On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: > > >> I am trying to use switch_r_sdp to rewrite the SDP. >> The problem I am facing has to do with the way of doing it. >> >> Let's say I have: >> >> v=0 >> o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx >> s=Opal SIP Session >> c=IN IP4 xxx.xxx.xxx.xxx >> t=0 0 >> m=audio 5144 RTP/AVP 18 3 101 120 >> c=IN IP4 xxx.xxx.xxx.xxx >> a=rtpmap:18 G729/8000/1 >> a=fmtp:18 annexb=no >> a=rtpmap:3 gsm/8000/1 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16,32,36 >> a=rtpmap:120 NSE/8000 >> a=fmtp:120 192-193 >> >> who to I set the switch_r_sdp variable in xml? >> >> Obviously this doesn't work : >> >> >> >> Do I have to escape any special characters and how? >> I tried using escaped quotes, escaped spaces, escaped tabs etc. >> Nothing worked. >> >> Any suggestions? >> >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Tue Nov 17 12:15:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Nov 2009 14:15:02 -0600 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp In-Reply-To: <4B02FD30.8050502@kinetix.gr> References: <4B02A829.7080708@kinetix.gr> <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> <4B02FD30.8050502@kinetix.gr> Message-ID: <191c3a030911171215s336b7d7bk7b1959744da3d2d3@mail.gmail.com> you can do On Tue, Nov 17, 2009 at 1:44 PM, regs at kinetix.gr wrote: > I am trying to achieve something similar to that : > > http://wiki.freeswitch.org/wiki/Codec_negotiation#Modifying_the_codec_when_using_proxy_media_mode > > but I am using xml_curl to create the dialplan (i.e. the web server that > serves the dialplan makes the decision about the SDP). So I need a way > to write > the new SDP in the XML dialplan response. However, in the above example > due to the regex manipulation the user is not facing the problem that I am > with setting the switch_r_sdp to a complex value that contains =, > spaces, new lines etc. > > Brian West wrote: > > Why are you needing to rewrite it? > > > > /b > > > > On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: > > > > > >> I am trying to use switch_r_sdp to rewrite the SDP. > >> The problem I am facing has to do with the way of doing it. > >> > >> Let's say I have: > >> > >> v=0 > >> o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx > >> s=Opal SIP Session > >> c=IN IP4 xxx.xxx.xxx.xxx > >> t=0 0 > >> m=audio 5144 RTP/AVP 18 3 101 120 > >> c=IN IP4 xxx.xxx.xxx.xxx > >> a=rtpmap:18 G729/8000/1 > >> a=fmtp:18 annexb=no > >> a=rtpmap:3 gsm/8000/1 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16,32,36 > >> a=rtpmap:120 NSE/8000 > >> a=fmtp:120 192-193 > >> > >> who to I set the switch_r_sdp variable in xml? > >> > >> Obviously this doesn't work : > >> > >> > >> > >> Do I have to escape any special characters and how? > >> I tried using escaped quotes, escaped spaces, escaped tabs etc. > >> Nothing worked. > >> > >> Any suggestions? > >> > >> > >> > >> > >> -- > >> ------------------------------------------- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: regs at kinetix.gr > >> ------------------------------------------- > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/726426fb/attachment-0002.html From anthony.minessale at gmail.com Tue Nov 17 12:17:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Nov 2009 14:17:04 -0600 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp In-Reply-To: <191c3a030911171215s336b7d7bk7b1959744da3d2d3@mail.gmail.com> References: <4B02A829.7080708@kinetix.gr> <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> <4B02FD30.8050502@kinetix.gr> <191c3a030911171215s336b7d7bk7b1959744da3d2d3@mail.gmail.com> Message-ID: <191c3a030911171217q14acd9c3la4427fcfa7ccc250@mail.gmail.com> I should have said On Tue, Nov 17, 2009 at 2:15 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you can do > > > ]]> > > > > On Tue, Nov 17, 2009 at 1:44 PM, regs at kinetix.gr wrote: > >> I am trying to achieve something similar to that : >> >> http://wiki.freeswitch.org/wiki/Codec_negotiation#Modifying_the_codec_when_using_proxy_media_mode >> >> but I am using xml_curl to create the dialplan (i.e. the web server that >> serves the dialplan makes the decision about the SDP). So I need a way >> to write >> the new SDP in the XML dialplan response. However, in the above example >> due to the regex manipulation the user is not facing the problem that I >> am >> with setting the switch_r_sdp to a complex value that contains =, >> spaces, new lines etc. >> >> Brian West wrote: >> > Why are you needing to rewrite it? >> > >> > /b >> > >> > On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: >> > >> > >> >> I am trying to use switch_r_sdp to rewrite the SDP. >> >> The problem I am facing has to do with the way of doing it. >> >> >> >> Let's say I have: >> >> >> >> v=0 >> >> o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx >> >> s=Opal SIP Session >> >> c=IN IP4 xxx.xxx.xxx.xxx >> >> t=0 0 >> >> m=audio 5144 RTP/AVP 18 3 101 120 >> >> c=IN IP4 xxx.xxx.xxx.xxx >> >> a=rtpmap:18 G729/8000/1 >> >> a=fmtp:18 annexb=no >> >> a=rtpmap:3 gsm/8000/1 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16,32,36 >> >> a=rtpmap:120 NSE/8000 >> >> a=fmtp:120 192-193 >> >> >> >> who to I set the switch_r_sdp variable in xml? >> >> >> >> Obviously this doesn't work : >> >> >> >> >> >> >> >> Do I have to escape any special characters and how? >> >> I tried using escaped quotes, escaped spaces, escaped tabs etc. >> >> Nothing worked. >> >> >> >> Any suggestions? >> >> >> >> >> >> >> >> >> >> -- >> >> ------------------------------------------- >> >> Apostolos Pantsiopoulos >> >> Kinetix Tele.com R & D >> >> email: regs at kinetix.gr >> >> ------------------------------------------- >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >> users >> >> http://www.freeswitch.org >> >> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/cf8ff333/attachment-0002.html From kristian.kielhofner at gmail.com Tue Nov 17 12:19:15 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Nov 2009 15:19:15 -0500 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> Message-ID: <2d9149cd0911171219y744cb81cwf53c5d25ea26c05e@mail.gmail.com> Ah yes, using spandsp instead of libvoipcodecs. I'm not going to question the wisdom of that move but it appears that spandsp (as-is) doesn't cross compile properly (make_at_dictionary is built using the cross compiler and can't run on the host). Once that error is fixed (I hacked it for now) it still bombs as shown here: http://pastebin.freeswitch.org/11149 spandsp + libtiff are almost certainly *much* larger than libvoipcodecs but if that means that I can also build mod_fax I guess it's worth it ;). On Tue, Nov 17, 2009 at 2:37 PM, Brian West wrote: > OH you need spandsp for VoipCodecs. ?No way around that one. > /b -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From tina at a2unlimited.com Tue Nov 17 12:21:22 2009 From: tina at a2unlimited.com (tina at a2unlimited.com) Date: Tue, 17 Nov 2009 15:21:22 -0500 Subject: [Freeswitch-users] ESL: No matching function... In-Reply-To: References: Message-ID: <3a74a318fe704f2015db18186913d71d.squirrel@emailmg.ipower.com> Confirmed 64-bit machine. > On Tue, Nov 17, 2009 at 10:04 AM, wrote: > >> MC, >> >> Yes, I tried "make && make perlmod", which did not fix the error. >> >> Just finished deploying an instance of the application on another server >> that did not produce the error (exact same configuration). >> >> Not sure what is causing it, or how to fix it. >> >> Bizarre. >> >> - Tina >> >> > Is the offending machine a 32 or 64 bit machine? Just curious if there is > something physically different about this machine than the others. > -MC > From stevendt at primrosebank.net Tue Nov 17 13:02:19 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 17 Nov 2009 21:02:19 -0000 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 Message-ID: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> Hi, I have just about got FreeSwitch working with a Cisco 7940 Phone. After much reading, I worked out that I needed a TFTP server on the network that would supply the phone with it's SIP personality and config etc. I have been able to get the phone working and realise that the TFTP server needs to be available every time the phone loses power etc. At the moment, I have the TFTP server running on a temporary machine but it would be neater if it ran on the same machine as FreeSwitch. This will be a very small FreeSwitch installation, so, ....... Is there any reason why I should not try to run FreeSwitch and the SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't think the server should put much load on the machine but wondered if there were any other reasons why this is a bad idea ? In addition, while I have the phone working - I get a status message on boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf Can anyone tell me how to locate the errors in this file please ? (I have posted it to the Pastebin) Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/56bad71a/attachment-0002.html From lists at tigertech.com Tue Nov 17 14:18:42 2009 From: lists at tigertech.com (Robert L Mathews) Date: Tue, 17 Nov 2009 14:18:42 -0800 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time Message-ID: <4B032142.1000308@tigertech.com> I'm using FreeSWITCH 1.0.4. When I make a call from a SIP phone to either a conference or an echo test on the FreeSWITCH server, the latency ("lag") starts off very low -- a fraction of a second. But as several minutes of time goes by, the lag increases. After, say, 15 minutes, the lag will reach a couple of seconds, making conference calls unusable. This does not happen on pure SIP-to-SIP calls, even when FreeSWITCH is handling the RTP media. If I hang up and immediately call back in (even to the same conference), the lag is reset to almost zero. If I put the call on "hold" and take it off hold, the lag is also gone. During testing, I've found that this may be related to the freeswitch app on the server not getting all the CPU time it wants. If I suspend the freeswitch process for two seconds and then resume it, the sound stops for two seconds, as I'd expect. But the echo/conference calls that were active are then lagged by two seconds until they hang up (or get put on hold), even after freeswitch is resumed and getting all the CPU time it needs. This is easily reproduced by making a SIP call to the echo test module, then: pkill -STOP freeswitch; sleep 2; pkill -CONT freeswitch Any echo test or conference call that was in progress will then be permanently lagged by two seconds. However, any SIP-to-SIP calls that were in progress will not become lagged. Of course, killing it with -STOP is an artificially nasty thing to do. But it effectively just prevents it from being scheduled on the CPU for a short period of time, and I can duplicate the same behavior (more gradually) by just increasing the load on the machine to the point that the freeswitch app isn't getting much CPU time. Just for the record, I get the same results from several different phones and several different Internet connections, all of which have a ping latency of under 40 ms to the server. This problem does not happen using the same phones and network connections to an asterisk server. Throwing out an even more complicated example that I've encountered: If I have a SIP-to-SIP call going from party A to party B and I stop the process for two seconds, it doesn't permanently introduce lag to that call, as I mentioned. But if a third person (party C) starts eavesdropping on the call and presses "3" to make it a three way call, and then I suspend it for two seconds, the call between A and B isn't lagged, but what party C hears and sends *is* lagged. Any ideas on how to fix this? Do other people see the same thing happening? As I said, the gradual increase in lag over a long period of time makes long conferences unusable, unfortunately. -- Rob From jlenk at frontiernet.net Tue Nov 17 18:38:17 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 17 Nov 2009 18:38:17 -0800 (PST) Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> Message-ID: <1258511897776-4023012.post@n2.nabble.com> Hi I run the SolarWinds TFTP server alongside FS on my small installation - works nicely! Jeff Dave Stevenson wrote: > > Hi, > > I have just about got FreeSwitch working with a Cisco 7940 Phone. After > much reading, I worked out that I needed a TFTP server on the network that > would supply the phone with it's SIP personality and config etc. I have > been able to get the phone working and realise that the TFTP server needs > to be available every time the phone loses power etc. At the moment, I > have the TFTP server running on a temporary machine but it would be neater > if it ran on the same machine as FreeSwitch. This will be a very small > FreeSwitch installation, so, ....... > > Is there any reason why I should not try to run FreeSwitch and the > SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't think > the server should put much load on the machine but wondered if there were > any other reasons why this is a bad idea ? > > In addition, while I have the phone working - I get a status message on > boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf > > Can anyone tell me how to locate the errors in this file please ? (I have > posted it to the Pastebin) > > Regards > Dave > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/TFTP-Server-Cisco-7540-tp4021305p4023012.html Sent from the freeswitch-users mailing list archive at Nabble.com. From elihayun at gmail.com Tue Nov 17 21:36:49 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 18 Nov 2009 07:36:49 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: References: <4B0266F4.8070602@savion.huji.ac.il> Message-ID: <4B0387F1.7070105@savion.huji.ac.il> Brian West wrote: > Why do you need to know the destination profile like that? You get to > pick that on your own so you should already know that before hand. > > > /b > > On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote: > > >> Hi >> We have more then one profile. To make a call I have to enter : bridge >> sofia/profile/number at ip >> The problem is when I use : "${use_profile}" I am getting the caller >> profile, and I need the destination profile. >> >> How do I get this information? >> >> Thanks >> >> Eli >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Thanks for your answer. The problem is when I call to that number that the phone hook to other server, I cannot make the call. Is there is a variable that can tell me the destination profile? Lets say the other profile called "ph1" I have to dial sofia/ph1/xxxxx at host to make the call. Is there other way to do that? Thanks Eli -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/7224c4ea/attachment-0002.html From ujjval at simplesignal.com Tue Nov 17 21:49:02 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Tue, 17 Nov 2009 21:49:02 -0800 Subject: [Freeswitch-users] Changing User-Agent String Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#User_Agent_.5Buser-agent-string.5D As per the above link, we can change the User Agent String, but I added this param name but does not seem to work. [user at freeswitch autoload_configs]$ vi sofia.conf.xml -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/5697880f/attachment-0002.html From mrene_lists at avgs.ca Tue Nov 17 21:52:01 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 21:52:01 -0800 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> Message-ID: It needs to go in the profile, not in sofia's global config. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 9:49 PM, Ujjval Karihaloo wrote: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#User_Agent_.5Buser-agent-string.5D > > As per the above link, we can change the User Agent String, but I > added this param name but does not seem to work. > > [user at freeswitch autoload_configs]$ vi sofia.conf.xml > > > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091117/c3ad8dcc/attachment-0002.html From mcampbellsmith at gmail.com Tue Nov 17 21:53:27 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 18 Nov 2009 16:53:27 +1100 Subject: [Freeswitch-users] application="info" Message-ID: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> HI All, pretty basic question and I feel a bit stupid asking this, but what are the prerequisites for the INFO to be displayed when is called in a dialplan? ie are there requirements on the loglevel, does the INFO command have to be put at a certain place in the dialplan etc? The reason i ask is that I have a dialplan and the is not getting triggered on the fs_cli output. Is there some other debbugging level that needs to be set? Thanks! From mrene_lists at avgs.ca Tue Nov 17 21:56:09 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 21:56:09 -0800 Subject: [Freeswitch-users] application="info" In-Reply-To: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> References: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> Message-ID: <89E59F4D-6335-4D51-A17B-9A040EE2ACAC@avgs.ca> If you press F8 (or do /log 7), you will see what the dialplan is executing, try to see if you see the info app in there And, your global loglevel has to be <= INFO too... fsctl loglevel debug Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote: > HI All, > > pretty basic question and I feel a bit stupid asking this, but what > are the prerequisites for the INFO to be displayed when application="info"/> is called in a dialplan? > > ie are there requirements on the loglevel, does the INFO command have > to be put at a certain place in the dialplan etc? > > The reason i ask is that I have a dialplan and the application="info"/> is not getting triggered on the fs_cli output. > Is there some other debbugging level that needs to be set? > > Thanks! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Tue Nov 17 22:05:41 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 18 Nov 2009 17:05:41 +1100 Subject: [Freeswitch-users] application="info" In-Reply-To: <89E59F4D-6335-4D51-A17B-9A040EE2ACAC@avgs.ca> References: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> <89E59F4D-6335-4D51-A17B-9A040EE2ACAC@avgs.ca> Message-ID: <33c87fa30911172205m75e5ab91m3e54a4647d098e2@mail.gmail.com> I had console loglevel set to DEBUG, so that should be fine. And I do see that FS is executing the exact extension where I have put the INFO application - still no info on the console.... I'm using FreeSWITCH Version 1.0.trunk (15490) On Wed, Nov 18, 2009 at 4:56 PM, Mathieu Rene wrote: > If you press F8 (or do /log 7), you will see what the dialplan is > executing, try to see if you see the info app in there > > And, your global loglevel has to be <= INFO too... fsctl loglevel debug > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote: > >> HI All, >> >> pretty basic question and I feel a bit stupid asking this, but what >> are the prerequisites for the INFO to be displayed when > application="info"/> is called in a dialplan? >> >> ie are there requirements on the loglevel, does the INFO command have >> to be put at a certain place in the dialplan etc? >> >> The reason i ask is that I have a dialplan and the > application="info"/> is not getting triggered on the fs_cli output. >> Is there some other debbugging level that needs to be set? >> >> Thanks! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mrene_lists at avgs.ca Tue Nov 17 22:08:59 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Nov 2009 22:08:59 -0800 Subject: [Freeswitch-users] application="info" In-Reply-To: <33c87fa30911172205m75e5ab91m3e54a4647d098e2@mail.gmail.com> References: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> <89E59F4D-6335-4D51-A17B-9A040EE2ACAC@avgs.ca> <33c87fa30911172205m75e5ab91m3e54a4647d098e2@mail.gmail.com> Message-ID: Console loglevel only sets the loglevel on the console, not on fs_cli or other event_socket client programs. You have to do /log 7 on fs_cli. fsctl loglevel is the global system loglevel, if its at warning, you wont see anything below warning ANYWHERE (console/event socket log files) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 10:05 PM, Mark Campbell-Smith wrote: > I had console loglevel set to DEBUG, so that should be fine. > > And I do see that FS is executing the exact extension where I have put > the INFO application - still no info on the console.... > > I'm using FreeSWITCH Version 1.0.trunk (15490) > > > > On Wed, Nov 18, 2009 at 4:56 PM, Mathieu Rene > wrote: >> If you press F8 (or do /log 7), you will see what the dialplan is >> executing, try to see if you see the info app in there >> >> And, your global loglevel has to be <= INFO too... fsctl loglevel >> debug >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote: >> >>> HI All, >>> >>> pretty basic question and I feel a bit stupid asking this, but what >>> are the prerequisites for the INFO to be displayed when >> application="info"/> is called in a dialplan? >>> >>> ie are there requirements on the loglevel, does the INFO command >>> have >>> to be put at a certain place in the dialplan etc? >>> >>> The reason i ask is that I have a dialplan and the >> application="info"/> is not getting triggered on the fs_cli output. >>> Is there some other debbugging level that needs to be set? >>> >>> Thanks! >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Tue Nov 17 22:17:31 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 18 Nov 2009 17:17:31 +1100 Subject: [Freeswitch-users] application="info" In-Reply-To: References: <33c87fa30911172153r28c753a8kc07351ea00fcd07a@mail.gmail.com> <89E59F4D-6335-4D51-A17B-9A040EE2ACAC@avgs.ca> <33c87fa30911172205m75e5ab91m3e54a4647d098e2@mail.gmail.com> Message-ID: <33c87fa30911172217p5f22ae24j1721c92e060538d8@mail.gmail.com> ahha... great. Thanks Mathieu. On Wed, Nov 18, 2009 at 5:08 PM, Mathieu Rene wrote: > Console loglevel only sets the loglevel on the console, not on fs_cli > or other event_socket client programs. You have to do /log 7 on fs_cli. > > fsctl loglevel is the global system loglevel, if its at warning, you > wont see anything below warning ANYWHERE (console/event socket log > files) > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Nov-09, at 10:05 PM, Mark Campbell-Smith wrote: > >> I had console loglevel set to DEBUG, so that should be fine. >> >> And I do see that FS is executing the exact extension where I have put >> the INFO application - still no info on the console.... >> >> I'm using FreeSWITCH Version 1.0.trunk (15490) >> >> >> >> On Wed, Nov 18, 2009 at 4:56 PM, Mathieu Rene >> wrote: >>> If you press F8 (or do /log 7), you will see what the dialplan is >>> executing, try to see if you see the info app in there >>> >>> And, your global loglevel has to be <= INFO too... fsctl loglevel >>> debug >>> >>> Mathieu Rene >>> Avant-Garde Solutions Inc >>> Office: + 1 (514) 664-1044 x100 >>> Cell: +1 (514) 664-1044 x200 >>> mrene at avgs.ca >>> >>> >>> >>> >>> On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote: >>> >>>> HI All, >>>> >>>> pretty basic question and I feel a bit stupid asking this, but what >>>> are the prerequisites for the INFO to be displayed when >>> application="info"/> is called in a dialplan? >>>> >>>> ie are there requirements on the loglevel, does the INFO command >>>> have >>>> to be put at a certain place in the dialplan etc? >>>> >>>> The reason i ask is that I have a dialplan and the >>> application="info"/> is not getting triggered on the fs_cli output. >>>> Is there some other debbugging level that needs to be set? >>>> >>>> Thanks! >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jonas.gauffin at gmail.com Wed Nov 18 00:05:34 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 18 Nov 2009 09:05:34 +0100 Subject: [Freeswitch-users] acl configuration Message-ID: Hello, What should my acl.conf.xml look like if I want to allow ALL calls on the external profile and use only digest authentication on all other profiles? Thanks, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/8df807b4/attachment-0002.html From regs at kinetix.gr Wed Nov 18 00:13:42 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Wed, 18 Nov 2009 10:13:42 +0200 Subject: [Freeswitch-users] Rewriting SDP with switch_r_sdp In-Reply-To: <191c3a030911171217q14acd9c3la4427fcfa7ccc250@mail.gmail.com> References: <4B02A829.7080708@kinetix.gr> <07EA3C0C-C650-4492-A78A-6F42FAA144CC@freeswitch.org> <4B02FD30.8050502@kinetix.gr> <191c3a030911171215s336b7d7bk7b1959744da3d2d3@mail.gmail.com> <191c3a030911171217q14acd9c3la4427fcfa7ccc250@mail.gmail.com> Message-ID: <4B03ACB6.8060302@kinetix.gr> Thanx! That worked fine. Anthony Minessale wrote: > I should have said > > > ]]> > > > > On Tue, Nov 17, 2009 at 2:15 PM, Anthony Minessale > > wrote: > > you can do > > > ]]> > > > > On Tue, Nov 17, 2009 at 1:44 PM, regs at kinetix.gr > > > wrote: > > I am trying to achieve something similar to that : > http://wiki.freeswitch.org/wiki/Codec_negotiation#Modifying_the_codec_when_using_proxy_media_mode > > but I am using xml_curl to create the dialplan (i.e. the web > server that > serves the dialplan makes the decision about the SDP). So I need > a way > to write > the new SDP in the XML dialplan response. However, in the above > example > due to the regex manipulation the user is not facing the > problem that I am > with setting the switch_r_sdp to a complex value that contains =, > spaces, new lines etc. > > Brian West wrote: > > Why are you needing to rewrite it? > > > > /b > > > > On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: > > > > > >> I am trying to use switch_r_sdp to rewrite the SDP. > >> The problem I am facing has to do with the way of doing it. > >> > >> Let's say I have: > >> > >> v=0 > >> o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx > >> s=Opal SIP Session > >> c=IN IP4 xxx.xxx.xxx.xxx > >> t=0 0 > >> m=audio 5144 RTP/AVP 18 3 101 120 > >> c=IN IP4 xxx.xxx.xxx.xxx > >> a=rtpmap:18 G729/8000/1 > >> a=fmtp:18 annexb=no > >> a=rtpmap:3 gsm/8000/1 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16,32,36 > >> a=rtpmap:120 NSE/8000 > >> a=fmtp:120 192-193 > >> > >> who to I set the switch_r_sdp variable in xml? > >> > >> Obviously this doesn't work : > >> > >> > >> > >> Do I have to escape any special characters and how? > >> I tried using escaped quotes, escaped spaces, escaped tabs etc. > >> Nothing worked. > >> > >> Any suggestions? > >> > >> > >> > >> > >> -- > >> ------------------------------------------- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: regs at kinetix.gr > >> ------------------------------------------- > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From stevendt at primrosebank.net Wed Nov 18 01:03:08 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 18 Nov 2009 09:03:08 -0000 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> <1258511897776-4023012.post@n2.nabble.com> Message-ID: <1015DE92916E4605A896A02360B21E27@bp1.ad.bp.com> Hi Jeff, thanks lot for the feedback, sounds like it'll do just what I want without breaking FreeSwitch, or needing to have a dedicated server just for a few personality and supporting files. I have been messing with some custom ring tones. As I'm sure you know, but others might not, the Cisco phone downloads the sound file from the TFTP server everytime you change the ringtone from one of the default ones so the TFTP server needs to be on all the time so running it on the FreeSwitch machine is ideal for me, regards Dave ----- Original Message ----- From: "Jeff Lenk" To: Sent: Wednesday, November 18, 2009 2:38 AM Subject: Re: [Freeswitch-users] TFTP Server & Cisco 7540 > > Hi > > I run the SolarWinds TFTP server alongside FS on my small installation - > works nicely! > > Jeff > > > > Dave Stevenson wrote: >> >> Hi, >> >> I have just about got FreeSwitch working with a Cisco 7940 Phone. After >> much reading, I worked out that I needed a TFTP server on the network >> that >> would supply the phone with it's SIP personality and config etc. I have >> been able to get the phone working and realise that the TFTP server needs >> to be available every time the phone loses power etc. At the moment, I >> have the TFTP server running on a temporary machine but it would be >> neater >> if it ran on the same machine as FreeSwitch. This will be a very small >> FreeSwitch installation, so, ....... >> >> Is there any reason why I should not try to run FreeSwitch and the >> SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't >> think >> the server should put much load on the machine but wondered if there were >> any other reasons why this is a bad idea ? >> >> In addition, while I have the phone working - I get a status message on >> boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf >> >> Can anyone tell me how to locate the errors in this file please ? (I have >> posted it to the Pastebin) >> >> Regards >> Dave >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/TFTP-Server-Cisco-7540-tp4021305p4023012.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mustafa.pk at gmail.com Wed Nov 18 01:20:45 2009 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Wed, 18 Nov 2009 14:20:45 +0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <1015DE92916E4605A896A02360B21E27@bp1.ad.bp.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> <1258511897776-4023012.post@n2.nabble.com> <1015DE92916E4605A896A02360B21E27@bp1.ad.bp.com> Message-ID: <8213d6070911180120r2f8c2908x25825c4df96c6229@mail.gmail.com> i don't really think you need to erect a dedicated server for serving configs and binaries over tftp protocol, when installation size is not _very_ large. On Wed, Nov 18, 2009 at 2:03 PM, Dave Stevenson wrote: > Hi Jeff, > > thanks ?lot for the feedback, sounds like it'll do just what I want without > breaking FreeSwitch, or needing to have a dedicated server just for a few > personality and supporting files. > > I have been messing with some custom ring tones. As I'm sure you know, but > others might not, the Cisco phone downloads the sound file from the TFTP > server everytime you change the ringtone from one of the default ones so the > TFTP server needs to be on all the time so running it on the FreeSwitch > machine is ideal for me, > > regards > Dave > > > > ----- Original Message ----- > From: "Jeff Lenk" > To: > Sent: Wednesday, November 18, 2009 2:38 AM > Subject: Re: [Freeswitch-users] TFTP Server & Cisco 7540 > > >> >> Hi >> >> I run the SolarWinds TFTP server alongside FS on my small installation - >> works nicely! >> >> Jeff >> >> >> >> Dave Stevenson wrote: >>> >>> Hi, >>> >>> I have just about got FreeSwitch working with a Cisco 7940 Phone. After >>> much reading, I worked out that I needed a TFTP server on the network >>> that >>> would supply the phone with it's SIP personality and config etc. I have >>> been able to get the phone working and realise that the TFTP server needs >>> to be available every time the phone loses power etc. At the moment, I >>> have the TFTP server running on a temporary machine but it would be >>> neater >>> if it ran on the same machine as FreeSwitch. This will be a very small >>> FreeSwitch installation, so, ....... >>> >>> Is there any reason why I should not try to run FreeSwitch and the >>> SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't >>> think >>> the server should put much load on the machine but wondered if there were >>> any other reasons why this is a bad idea ? >>> >>> In addition, while I have the phone working - I get a status message on >>> boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf >>> >>> Can anyone tell me how to locate the errors in this file please ? (I have >>> posted it to the Pastebin) >>> >>> Regards >>> Dave >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/TFTP-Server-Cisco-7540-tp4021305p4023012.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com From abeka at greatiam.com Wed Nov 18 02:43:15 2009 From: abeka at greatiam.com (Sam Abekah-Mensah) Date: Wed, 18 Nov 2009 10:43:15 +0000 Subject: [Freeswitch-users] Registration Error - 408 timeout and now 403 In-Reply-To: <87f2f3b90911171105s7fb2fea3l316fc2777cbc051a@mail.gmail.com> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> <4B01A8AE.7070708@greatiam.com> <4B02BD53.5040203@greatiam.com> <87f2f3b90911171105s7fb2fea3l316fc2777cbc051a@mail.gmail.com> Message-ID: <4B03CFC3.7040501@greatiam.com> Thank you so much for your responses. I have resolved the problem somehow. I copied the default.xml from the root conf folder, the sample 1001 and 1002 .xml on a windows build to the Fedora 11 machine and that worked. I guessed the rejection was with the configuration on theFedora box.even though it was more straight -out-of-the-box. No one is seeking help on this so it must be something I did. I am reinstalling FC11 from scartch and see if I can reproduce the error after FS-1.0.4 install. Thanks folks . Michael Collins wrote: > Try doing this: > http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install > > -MC > > On Tue, Nov 17, 2009 at 7:12 AM, Sam Abekah-Mensah > wrote: > > Hello > > I have tried the same setup but this time using a windows build > FS1.0.4 > on an XP machine and all is fine. The sample 1001 and 1002 IDs work > without any tweaking at all. Could the problem be with the linux > build > 1.0.4.? I am running on an FC11 machine. > > On the FC11 box I used the svn link to build using the ff: > > bootstarp.sh > configure withoout libcurl to eliinate the spidermonkey lib error > make > make install > > Did I miss anything ? > From elihayun at gmail.com Wed Nov 18 04:56:25 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 18 Nov 2009 14:56:25 +0200 Subject: [Freeswitch-users] change event value Message-ID: <4B03EEF9.7070802@savion.huji.ac.il> Hi Is there is a way to intercept an event (for example : REGISTER) and change one of its parameters (for example: the extension number) and fire up the corrected event? I need it to set the speedial of the phone value to be "**xxxxx" but to make it register as "xxxxx" Thanks Eli From siniypin at gmail.com Wed Nov 18 05:07:31 2009 From: siniypin at gmail.com (RobertT) Date: Wed, 18 Nov 2009 16:07:31 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> Message-ID: <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> I've tried to add ;transport=tcp in dialplan bridge application and it has ended up with error on FS with message "can't find registered extension * called_extension*%external_call;transport=tcp" whereas this extension is registered in FS via tcp. Also I tried to reproduce the same scenario with public SIP server and everything worked fine from what I can draw that problem is with my FS configuration or maybe with some FS host's network configuration. Any help will be appretiated. Best regards, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/74eb06b3/attachment-0002.html From brian at freeswitch.org Wed Nov 18 06:24:27 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 08:24:27 -0600 Subject: [Freeswitch-users] acl configuration In-Reply-To: References: Message-ID: <08832B55-1243-498B-88D7-D85E6509AAEF@freeswitch.org> You wouldn't use ACL's at all. You just set auth-calls=false and remove all references to ACL's /b On Nov 18, 2009, at 2:05 AM, Jonas Gauffin wrote: > Hello, > > What should my acl.conf.xml look like if I want to allow ALL calls > on the external profile and use only digest authentication on all > other profiles? > > Thanks, > Jonas > __________ From brian at freeswitch.org Wed Nov 18 06:25:27 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 08:25:27 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> Message-ID: <4EF7A0AD-A76B-4B2D-BA16-EE49C2AB9467@freeswitch.org> AH you are hiding the fact you use FreeSWITCH... Great... but why? /b On Nov 17, 2009, at 11:52 PM, Mathieu Rene wrote: > It needs to go in the profile, not in sofia's global config. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/0e42a604/attachment-0002.html From brian at freeswitch.org Wed Nov 18 06:27:31 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 08:27:31 -0600 Subject: [Freeswitch-users] Registration Error - 408 timeout and now 403 In-Reply-To: <4B03CFC3.7040501@greatiam.com> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> <4B01A8AE.7070708@greatiam.com> <4B02BD53.5040203@greatiam.com> <87f2f3b90911171105s7fb2fea3l316fc2777cbc051a@mail.gmail.com> <4B03CFC3.7040501@greatiam.com> Message-ID: <4271AC30-877E-4E56-975B-77F71C3BD466@freeswitch.org> 403 is Forbidden, So its not really an error you're just getting told NO. You should follow the guide on the wiki on how to debug. 1. Turn on SIP Trace. sofia profile xxx siptrace on 2. Press F8 The error logs are very verbose and usually point to the problem. /b On Nov 18, 2009, at 4:43 AM, Sam Abekah-Mensah wrote: > Thank you so much for your responses. > > I have resolved the problem somehow. > I copied the default.xml from the root conf folder, the sample 1001 > and > 1002 .xml on a windows build to the Fedora 11 machine and that worked. > I guessed the rejection was with the configuration on theFedora > box.even > though it was more straight -out-of-the-box. No one is seeking help > on > this so it must be something I did. > > I am reinstalling FC11 from scartch and see if I can reproduce the > error > after FS-1.0.4 install. > > Thanks folks From brian at freeswitch.org Wed Nov 18 06:27:59 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 08:27:59 -0600 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> Message-ID: <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> How exactly are you doing this? /b On Nov 18, 2009, at 7:07 AM, RobertT wrote: > I've tried to add ;transport=tcp in dialplan bridge application and > it has ended up with error on FS with message "can't find registered > extension called_extension%external_call;transport=tcp" whereas this > extension is registered in FS via tcp. Also I tried to reproduce the > same scenario with public SIP server and everything worked fine from > what I can draw that problem is with my FS configuration or maybe > with some FS host's network configuration. > Any help will be appretiated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/03e9c623/attachment-0002.html From ujjval at simplesignal.com Wed Nov 18 07:17:04 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Wed, 18 Nov 2009 07:17:04 -0800 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> Not sure I am the only one changing User-Agent....but I just want a way for our Customers to know the purpose of the server when they talk to it. There is FreeSwitch written into the SDP "o" line as well...which I don't care about, I want to have something in there that identifies the purpose of the server. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Tuesday, November 17, 2009 10:52 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Changing User-Agent String It needs to go in the profile, not in sofia's global config. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Nov-09, at 9:49 PM, Ujjval Karihaloo wrote: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#User_Agent_.5Buser-agent-string.5D As per the above link, we can change the User Agent String, but I added this param name but does not seem to work. [user at freeswitch autoload_configs]$ vi sofia.conf.xml _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/70896d2f/attachment-0002.html From brian at freeswitch.org Wed Nov 18 07:21:55 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 09:21:55 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> Message-ID: <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> you do realize that is NOT the purpose of the user-agent string... changing might break things in some people's configs due to some assumptions made about the user agent on the far side for interop purposes... its your choice to change it but it servers NO purpose doing so. /b On Nov 18, 2009, at 9:17 AM, Ujjval Karihaloo wrote: > Not sure I am the only one changing User-Agent?.but I just want a > way for our Customers to know the purpose of the server when they > talk to it. There is FreeSwitch written into the SDP ?o? line as > well?which I don?t care about, I want to have something in there > that identifies the purpose of the server. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/1fc50096/attachment-0002.html From kristian.kielhofner at gmail.com Wed Nov 18 07:39:24 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 18 Nov 2009 10:39:24 -0500 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> Message-ID: <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> That's *exactly* why I change the User-Agent string. If they're changing their behavior in some way based on the assumption I'm using FreeSWITCH I want to know about it :). On Wed, Nov 18, 2009 at 10:21 AM, Brian West wrote: > you do realize that is NOT the purpose of the user-agent string... changing > might break things in some people's configs due to some assumptions made > about the user agent on the far side for interop purposes... its your choice > to change it but it servers NO purpose doing so. > /b -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Wed Nov 18 08:08:48 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 10:08:48 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> Message-ID: <684091CA-4793-41C6-88D5-BE09146E5713@freeswitch.org> There are a few behaviors in FreeSWITCH that get triggered based on the remote side... you know all about those? :P I'm just saying its usually not wise to change it just in case. /b On Nov 18, 2009, at 9:39 AM, Kristian Kielhofner wrote: > That's *exactly* why I change the User-Agent string. If they're > changing their behavior in some way based on the assumption I'm using > FreeSWITCH I want to know about it :). From oscav at hotmail.fr Wed Nov 18 08:07:42 2009 From: oscav at hotmail.fr (Oscav) Date: Wed, 18 Nov 2009 08:07:42 -0800 (PST) Subject: [Freeswitch-users] sched_broadcast doesn't execute Message-ID: <26408422.post@talk.nabble.com> Hi, I'm writing a script in Javascript that plays a message during a bridge. I'm trying to use a sched_broadcast to do it. The job is scheduled and then deleted but I never hear the wav file and I don't get the "OK Message Scheduled" in the log. It even doesn't display any error message if I specify a wrong file name. Someone could help me on this issue ?? new_session.execute("sched_broadcast", "+20 alloted_timeout ${uuid} playback:ivr-welcome_to_freeswitch.wav"); I already did some posts but I got no answer. This is very difficult to progress without help. Many thanks -- View this message in context: http://old.nabble.com/sched_broadcast-doesn%27t-execute-tp26408422p26408422.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mayamatakeshi at gmail.com Wed Nov 18 08:32:04 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Thu, 19 Nov 2009 01:32:04 +0900 Subject: [Freeswitch-users] mod_fifo and multi-tenancy Message-ID: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> About mod_fifo, it would be safe to use it in multi-tenancy scenarios where domains are created and deleted all the time and in consequence, fifos are created all the time? I mean, fifos are eventually destroyed by mod_fifo itself. Is this correct? br, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/0dfd376c/attachment-0002.html From robert.hadley at teotech.com Wed Nov 18 08:40:16 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Wed, 18 Nov 2009 08:40:16 -0800 Subject: [Freeswitch-users] Anybody interested in helping fix the -srcdir option? Message-ID: <77F5794BDB0A4FD4BD7592D70F4719D4@greyhawk.tonecommander.com> Hi All, Anybody interested in helping fix the -srcdir option? I am trying to build in a subdirectory off the Freeswitch source. I am working on it and finding issues. However, being a newbie at autoconf/automake and shell scripting I sometimes struggle at finding fixes. For example, the script command below is in bootstrap.sh, but might need to be moved or duplicated in configure.* to support using configure -srcdir option, as the modules.conf file also needs to be to the build destination folder. if [ ! -f modules.conf ]; then cp build/modules.conf.in modules.conf fi Thanks, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/80dfcf50/attachment-0002.html From kristian.kielhofner at gmail.com Wed Nov 18 08:42:10 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 18 Nov 2009 11:42:10 -0500 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <684091CA-4793-41C6-88D5-BE09146E5713@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> <684091CA-4793-41C6-88D5-BE09146E5713@freeswitch.org> Message-ID: <2d9149cd0911180842t547bc6fdt526bbc9399485141@mail.gmail.com> Brian, I see some things based on SDP originator (we all know what those are about) but nothing for SIP user agent... More curious than anything else, am I missing something? On Wed, Nov 18, 2009 at 11:08 AM, Brian West wrote: > There are a few behaviors in FreeSWITCH that get triggered based on > the remote side... you know all about those? ?:P ?I'm just saying its > usually not wise to change it just in case. > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From mike at jerris.com Wed Nov 18 08:43:35 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Nov 2009 11:43:35 -0500 Subject: [Freeswitch-users] Compilation problem In-Reply-To: <32c1b333.68a1501e.4b02ad93.bbfcd@go2.pl> References: <32c1b333.68a1501e.4b02ad93.bbfcd@go2.pl> Message-ID: <5377DF41-8B48-443D-AD65-E4C5F82E58D7@jerris.com> This issue is now fixed in trunk. Mike On Nov 17, 2009, at 9:05 AM, Christopher Z. wrote: > Hi, > > I've got this error after make: > > http://pastebin.freeswitch.org/11145 > > Any idea how to fix this error ? From abeka at greatiam.com Wed Nov 18 08:44:10 2009 From: abeka at greatiam.com (Samuel 'Otis' Abekah-Mensah) Date: Wed, 18 Nov 2009 16:44:10 +0000 Subject: [Freeswitch-users] Registration Error - 408 timeout and now 403 In-Reply-To: <4271AC30-877E-4E56-975B-77F71C3BD466@freeswitch.org> References: <4AFF5701.8010508@greatiam.com> <87f2f3b90911160921w6d75a1caoed8095fd5aca938a@mail.gmail.com> <4B01A8AE.7070708@greatiam.com> <4B02BD53.5040203@greatiam.com> <87f2f3b90911171105s7fb2fea3l316fc2777cbc051a@mail.gmail.com> <4B03CFC3.7040501@greatiam.com> <4271AC30-877E-4E56-975B-77F71C3BD466@freeswitch.org> Message-ID: <4B04245A.8080000@greatiam.com> Thanks. I will look up how to debug on the wiki. Kinda late now with my setup; got to learn it anyway. Thanks for the direction. Brian West wrote: >
403 is > Forbidden, So its not really an error you're just getting told NO. > You should follow the guide on the wiki on how to debug. > > 1. Turn on SIP Trace. sofia profile xxx siptrace on > 2. Press F8 > > The error logs are very verbose and usually point to the problem. > > /b > > On Nov 18, 2009, at 4:43 AM, Sam Abekah-Mensah wrote: > >> Thank you so much for your responses. >> >> I have resolved the problem somehow. >> I copied the default.xml from the root conf folder, the sample 1001 and >> 1002 .xml on a windows build to the Fedora 11 machine and that worked. >> I guessed the rejection was with the configuration on theFedora box.even >> though it was more straight -out-of-the-box. No one is seeking help on >> this so it must be something I did. >> >> I am reinstalling FC11 from scartch and see if I can reproduce the error >> after FS-1.0.4 install. >> >> Thanks folks > > > >
From brian at freeswitch.org Wed Nov 18 09:14:59 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 11:14:59 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <2d9149cd0911180842t547bc6fdt526bbc9399485141@mail.gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> <684091CA-4793-41C6-88D5-BE09146E5713@freeswitch.org> <2d9149cd0911180842t547bc6fdt526bbc9399485141@mail.gmail.com> Message-ID: <7E7166FF-1824-4F68-957E-898B4AEBB40E@freeswitch.org> We do in sofia_reg and sofia_sla, and some where we do updates so we don't send things to phones we know will piss their pants. /b On Nov 18, 2009, at 10:42 AM, Kristian Kielhofner wrote: > Brian, > > I see some things based on SDP originator (we all know what those > are about) but nothing for SIP user agent... From steveu at coppice.org Wed Nov 18 09:45:37 2009 From: steveu at coppice.org (Steve Underwood) Date: Thu, 19 Nov 2009 01:45:37 +0800 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <2d9149cd0911171219y744cb81cwf53c5d25ea26c05e@mail.gmail.com> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> <2d9149cd0911171219y744cb81cwf53c5d25ea26c05e@mail.gmail.com> Message-ID: <4B0432C1.7000006@coppice.org> On 11/18/2009 04:19 AM, Kristian Kielhofner wrote: > Ah yes, using spandsp instead of libvoipcodecs. I'm not going to > question the wisdom of that move but it appears that spandsp (as-is) > doesn't cross compile properly (make_at_dictionary is built using the > cross compiler and can't run on the host). Once that error is fixed > (I hacked it for now) it still bombs as shown here: > > http://pastebin.freeswitch.org/11149 > > spandsp + libtiff are almost certainly *much* larger than > libvoipcodecs but if that means that I can also build mod_fax I guess > it's worth it ;). > > On Tue, Nov 17, 2009 at 2:37 PM, Brian West wrote: > >> OH you need spandsp for VoipCodecs. No way around that one. >> /b >> > Over time more and more of spandsp will be used by Freeswitch, so its most certainly an integral part of FS going forward. In a few months it might be possible to not use libTIFF, depending how things go with some developments. spandsp builds OK for many cross compile setups. make_at_dictionary and make_modem_filters should be built using the host compiler, not the target compiler. This seems to work in the places I've tried it. The problem in your pastebin log seems to be a broken C99 environment, and not a spandsp problem. Steve From helmut.kuper at ewetel.de Wed Nov 18 10:21:38 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 18 Nov 2009 19:21:38 +0100 Subject: [Freeswitch-users] Question about odbc support Message-ID: <4B043B32.20802@ewetel.de> Hi, does anybody know how to check the affected rows caused by delete, insert or update sql statements in FS? To do this with sqlite3 there is a function called switch_core_db_changes(). regards helmut From kristian.kielhofner at gmail.com Wed Nov 18 10:53:22 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 18 Nov 2009 13:53:22 -0500 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <7E7166FF-1824-4F68-957E-898B4AEBB40E@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <2d9149cd0911180739p6141e504scdbbc889cc505312@mail.gmail.com> <684091CA-4793-41C6-88D5-BE09146E5713@freeswitch.org> <2d9149cd0911180842t547bc6fdt526bbc9399485141@mail.gmail.com> <7E7166FF-1824-4F68-957E-898B4AEBB40E@freeswitch.org> Message-ID: <2d9149cd0911181053u20644421q725809f0a6a40377@mail.gmail.com> Ah, that explains it. I don't do any registration or presence/sla with FS - yet ;). On Wed, Nov 18, 2009 at 12:14 PM, Brian West wrote: > We do in sofia_reg and sofia_sla, and some where we do updates so we > don't send things to phones we know will piss their pants. > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From kristian.kielhofner at gmail.com Wed Nov 18 11:02:44 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 18 Nov 2009 14:02:44 -0500 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <4B0432C1.7000006@coppice.org> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> <2d9149cd0911171219y744cb81cwf53c5d25ea26c05e@mail.gmail.com> <4B0432C1.7000006@coppice.org> Message-ID: <2d9149cd0911181102h92309e8recc7f46e8f81bd88@mail.gmail.com> On Wed, Nov 18, 2009 at 12:45 PM, Steve Underwood wrote: > Over time more and more of spandsp will be used by Freeswitch, so its > most certainly an integral part of FS going forward. In a few months it > might be possible to not use libTIFF, depending how things go with some > developments. > > spandsp builds OK for many cross compile setups. make_at_dictionary and > make_modem_filters should be built using the host compiler, not the > target compiler. This seems to work in the places I've tried it. > > The problem in your pastebin log seems to be a broken C99 environment, > and not a spandsp problem. > > Steve > make_at_dictionary is not build using the host compiler. I had to hack it (manually passing CC and LIBTOOL to make) to get the build to proceed to the next error... This may be specific to the integration with the rest of the FreeSWITCH build system. I'm using uClibc (as are most other embedded environments) and I've had other C99 issues before. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Wed Nov 18 11:28:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Nov 2009 13:28:44 -0600 Subject: [Freeswitch-users] sched_broadcast doesn't execute In-Reply-To: <26408422.post@talk.nabble.com> References: <26408422.post@talk.nabble.com> Message-ID: <191c3a030911181128g35ba0652w9fc575d5586367dc@mail.gmail.com> is that your exact code? ${uuid} will not be expanded by javascript var uuid = session.getVariable(uuid); new_session.execute("sched_broadcast", "+20 alloted_timeout " + uuid + " playback:ivr-welcome_to_freeswitch.wav"); On Wed, Nov 18, 2009 at 10:07 AM, Oscav wrote: > > Hi, > > I'm writing a script in Javascript that plays a message during a bridge. > I'm > trying to use a sched_broadcast to do it. The job is scheduled and then > deleted but I never hear the wav file and I don't get the "OK Message > Scheduled" in the log. It even doesn't display any error message if I > specify a wrong file name. Someone could help me on this issue ?? > > new_session.execute("sched_broadcast", "+20 alloted_timeout ${uuid} > playback:ivr-welcome_to_freeswitch.wav"); > > I already did some posts but I got no answer. This is very difficult to > progress without help. > > Many thanks > -- > View this message in context: > http://old.nabble.com/sched_broadcast-doesn%27t-execute-tp26408422p26408422.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/eb531511/attachment-0002.html From william.nishio at voicetechnology.com.br Wed Nov 18 11:38:24 2009 From: william.nishio at voicetechnology.com.br (William Kendi ...) Date: Wed, 18 Nov 2009 17:38:24 -0200 Subject: [Freeswitch-users] mod dptools record problem - hangup channel with invalid file path Message-ID: <67d615ac0911181138m30f3064ci1a2dad6732354e35@mail.gmail.com> While I was testing the "mod dptools record" application using invalid file paths, i noted that the "mod dptools record" application terminated the call. I am currently looking for a way to change this behaviour. Any suggestions? Can this be done? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/3d073d11/attachment-0002.html From mike at jerris.com Wed Nov 18 11:40:11 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Nov 2009 14:40:11 -0500 Subject: [Freeswitch-users] Build FS without spandsp or libtiff In-Reply-To: <2d9149cd0911181102h92309e8recc7f46e8f81bd88@mail.gmail.com> References: <2d9149cd0911171121k2711d38fj8257a73c28e7889d@mail.gmail.com> <2d9149cd0911171133t74f12384lba9432961c723dd3@mail.gmail.com> <31D20BD6-74A4-423E-938C-72B2C9D676A2@freeswitch.org> <2d9149cd0911171219y744cb81cwf53c5d25ea26c05e@mail.gmail.com> <4B0432C1.7000006@coppice.org> <2d9149cd0911181102h92309e8recc7f46e8f81bd88@mail.gmail.com> Message-ID: <9E79AB11-C3CB-4418-9360-69497550F053@jerris.com> Kristian, catch up with me somewhere that I can get remote access to this build environment so that we can sort this out. Mike On Nov 18, 2009, at 2:02 PM, Kristian Kielhofner wrote: > On Wed, Nov 18, 2009 at 12:45 PM, Steve Underwood wrote: >> Over time more and more of spandsp will be used by Freeswitch, so its >> most certainly an integral part of FS going forward. In a few months it >> might be possible to not use libTIFF, depending how things go with some >> developments. >> >> spandsp builds OK for many cross compile setups. make_at_dictionary and >> make_modem_filters should be built using the host compiler, not the >> target compiler. This seems to work in the places I've tried it. >> >> The problem in your pastebin log seems to be a broken C99 environment, >> and not a spandsp problem. >> >> Steve >> > > make_at_dictionary is not build using the host compiler. I had to > hack it (manually passing CC and LIBTOOL to make) to get the build to > proceed to the next error... This may be specific to the integration > with the rest of the FreeSWITCH build system. > > I'm using uClibc (as are most other embedded environments) and I've > had other C99 issues before. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Nov 18 11:45:43 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 13:45:43 -0600 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <4B032142.1000308@tigertech.com> References: <4B032142.1000308@tigertech.com> Message-ID: <4256FA16-DE2A-4871-A398-30CC0E8E6D4C@freeswitch.org> Can you tell us what kind of phone you're using? And have you tried this on SVN trunk? /b On Nov 17, 2009, at 4:18 PM, Robert L Mathews wrote: > I'm using FreeSWITCH 1.0.4. > > When I make a call from a SIP phone to either a conference or an echo > test on the FreeSWITCH server, the latency ("lag") starts off very low > -- a fraction of a second. But as several minutes of time goes by, > the > lag increases. After, say, 15 minutes, the lag will reach a couple of > seconds, making conference calls unusable. > > This does not happen on pure SIP-to-SIP calls, even when FreeSWITCH is > handling the RTP media. > > If I hang up and immediately call back in (even to the same > conference), > the lag is reset to almost zero. If I put the call on "hold" and > take it > off hold, the lag is also gone. > > During testing, I've found that this may be related to the freeswitch > app on the server not getting all the CPU time it wants. > > If I suspend the freeswitch process for two seconds and then resume > it, > the sound stops for two seconds, as I'd expect. But the echo/ > conference > calls that were active are then lagged by two seconds until they > hang up > (or get put on hold), even after freeswitch is resumed and getting all > the CPU time it needs. > > This is easily reproduced by making a SIP call to the echo test > module, > then: > > pkill -STOP freeswitch; sleep 2; pkill -CONT freeswitch > > Any echo test or conference call that was in progress will then be > permanently lagged by two seconds. However, any SIP-to-SIP calls that > were in progress will not become lagged. > > Of course, killing it with -STOP is an artificially nasty thing to do. > But it effectively just prevents it from being scheduled on the CPU > for > a short period of time, and I can duplicate the same behavior (more > gradually) by just increasing the load on the machine to the point > that > the freeswitch app isn't getting much CPU time. > > Just for the record, I get the same results from several different > phones and several different Internet connections, all of which have a > ping latency of under 40 ms to the server. This problem does not > happen > using the same phones and network connections to an asterisk server. > > Throwing out an even more complicated example that I've encountered: > If > I have a SIP-to-SIP call going from party A to party B and I stop the > process for two seconds, it doesn't permanently introduce lag to that > call, as I mentioned. But if a third person (party C) starts > eavesdropping on the call and presses "3" to make it a three way call, > and then I suspend it for two seconds, the call between A and B isn't > lagged, but what party C hears and sends *is* lagged. > > Any ideas on how to fix this? Do other people see the same thing > happening? As I said, the gradual increase in lag over a long period > of > time makes long conferences unusable, unfortunately. > > -- > Rob > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Wed Nov 18 11:46:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Nov 2009 13:46:47 -0600 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <4B032142.1000308@tigertech.com> References: <4B032142.1000308@tigertech.com> Message-ID: <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> Have you tried SVN trunk? 1.0.4 is several months old and will soon be replaced with 1.0.5 so if you can find similar problems on trunk it would be more helpful than possibly pointing out issues that we have already fixed. On Tue, Nov 17, 2009 at 4:18 PM, Robert L Mathews wrote: > I'm using FreeSWITCH 1.0.4. > > When I make a call from a SIP phone to either a conference or an echo > test on the FreeSWITCH server, the latency ("lag") starts off very low > -- a fraction of a second. But as several minutes of time goes by, the > lag increases. After, say, 15 minutes, the lag will reach a couple of > seconds, making conference calls unusable. > > This does not happen on pure SIP-to-SIP calls, even when FreeSWITCH is > handling the RTP media. > > If I hang up and immediately call back in (even to the same conference), > the lag is reset to almost zero. If I put the call on "hold" and take it > off hold, the lag is also gone. > > During testing, I've found that this may be related to the freeswitch > app on the server not getting all the CPU time it wants. > > If I suspend the freeswitch process for two seconds and then resume it, > the sound stops for two seconds, as I'd expect. But the echo/conference > calls that were active are then lagged by two seconds until they hang up > (or get put on hold), even after freeswitch is resumed and getting all > the CPU time it needs. > > This is easily reproduced by making a SIP call to the echo test module, > then: > > pkill -STOP freeswitch; sleep 2; pkill -CONT freeswitch > > Any echo test or conference call that was in progress will then be > permanently lagged by two seconds. However, any SIP-to-SIP calls that > were in progress will not become lagged. > > Of course, killing it with -STOP is an artificially nasty thing to do. > But it effectively just prevents it from being scheduled on the CPU for > a short period of time, and I can duplicate the same behavior (more > gradually) by just increasing the load on the machine to the point that > the freeswitch app isn't getting much CPU time. > > Just for the record, I get the same results from several different > phones and several different Internet connections, all of which have a > ping latency of under 40 ms to the server. This problem does not happen > using the same phones and network connections to an asterisk server. > > Throwing out an even more complicated example that I've encountered: If > I have a SIP-to-SIP call going from party A to party B and I stop the > process for two seconds, it doesn't permanently introduce lag to that > call, as I mentioned. But if a third person (party C) starts > eavesdropping on the call and presses "3" to make it a three way call, > and then I suspend it for two seconds, the call between A and B isn't > lagged, but what party C hears and sends *is* lagged. > > Any ideas on how to fix this? Do other people see the same thing > happening? As I said, the gradual increase in lag over a long period of > time makes long conferences unusable, unfortunately. > > -- > Rob > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/cf37b5ba/attachment-0002.html From mike at jerris.com Wed Nov 18 11:48:44 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Nov 2009 14:48:44 -0500 Subject: [Freeswitch-users] Anybody interested in helping fix the -srcdir option? In-Reply-To: <77F5794BDB0A4FD4BD7592D70F4719D4@greyhawk.tonecommander.com> References: <77F5794BDB0A4FD4BD7592D70F4719D4@greyhawk.tonecommander.com> Message-ID: Fixed in svn r15526 and other fixes in svn r15527. mike On Nov 18, 2009, at 11:40 AM, Robert Hadley wrote: > Hi All, > > Anybody interested in helping fix the ?srcdir option? I am trying to build in a subdirectory off the Freeswitch source. I am working on it and finding issues. However, being a newbie at autoconf/automake and shell scripting I sometimes struggle at finding fixes. > > For example, the script command below is in bootstrap.sh, but might need to be moved or duplicated in configure.* to support using configure ?srcdir option, as the modules.conf file also needs to be to the build destination folder. > > if [ ! -f modules.conf ]; then > cp build/modules.conf.in modules.conf > fi > > Thanks, > Robert > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/07a46d7e/attachment-0002.html From mike at jerris.com Wed Nov 18 11:53:05 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Nov 2009 14:53:05 -0500 Subject: [Freeswitch-users] mod dptools record problem - hangup channel with invalid file path In-Reply-To: <67d615ac0911181138m30f3064ci1a2dad6732354e35@mail.gmail.com> References: <67d615ac0911181138m30f3064ci1a2dad6732354e35@mail.gmail.com> Message-ID: <994A83CB-7069-4808-9055-30B8BD3CEA75@jerris.com> Okay, I'll ask the obvious question. Why are you passing record invalid file paths and why should it not fail if you do? Mike On Nov 18, 2009, at 2:38 PM, William Kendi ... wrote: > While I was testing the "mod dptools record" application using invalid file paths, i noted that the "mod dptools record" application terminated the call. > I am currently looking for a way to change this behaviour. > Any suggestions? Can this be done? From william.nishio at voicetechnology.com.br Wed Nov 18 12:26:00 2009 From: william.nishio at voicetechnology.com.br (William Kendi ...) Date: Wed, 18 Nov 2009 18:26:00 -0200 Subject: [Freeswitch-users] mod dptools record problem - hangup channel with invalid file path In-Reply-To: <994A83CB-7069-4808-9055-30B8BD3CEA75@jerris.com> References: <67d615ac0911181138m30f3064ci1a2dad6732354e35@mail.gmail.com> <994A83CB-7069-4808-9055-30B8BD3CEA75@jerris.com> Message-ID: <67d615ac0911181226y22b4fec6ndb8e622a24db101c@mail.gmail.com> Actually, I am integrating FreeSWITCH with a weird IVR Framework, and the current behaviour of the "mod dptools record" application breaks some rules of the weird IVR Framework that must be integrated with FreeSWITCH. In order to integrate FreeSWITCH with the weird IVR Framework, the "mod dptools record" application mustn't terminate the call when invalid file paths are passed, and a notification of the invalid file path through the event system of FreeSWITCH should be enough for me, like the behaviour of the "mod dptools playback" application when invalid file paths are passed. Thanks in advance. ** 2009/11/18 Michael Jerris > Okay, I'll ask the obvious question. Why are you passing record invalid > file paths and why should it not fail if you do? > > Mike > > On Nov 18, 2009, at 2:38 PM, William Kendi ... wrote: > > > While I was testing the "mod dptools record" application using invalid > file paths, i noted that the "mod dptools record" application terminated the > call. > > I am currently looking for a way to change this behaviour. > > Any suggestions? Can this be done? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/2d9c412e/attachment-0002.html From timuckun at gmail.com Wed Nov 18 12:39:18 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 19 Nov 2009 09:39:18 +1300 Subject: [Freeswitch-users] Hardware echo cancellation. Message-ID: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> I am about to build a new machine as a VOIP server. I am going to get either a quad core intel or a six core AMD processor with at least eight gigabytes of RAM in it. Given that much horsepower I am wondering if there is any need to purchase hardware with echo cancellation (I am thinking about redfone devices).. I can save some money by not getting the echo cancellation. So is it worth saving that money? Is it always better to have hardware echo cancellation? Is a quad core capable of dealing with echo cancellation needs of an IVR which is going to take lots of simultaneous calls? Thanks. From mrene_lists at avgs.ca Wed Nov 18 12:52:29 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 18 Nov 2009 12:52:29 -0800 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> Message-ID: <6A7DC321-F2E6-493F-ACFD-0373950D9659@avgs.ca> If you have TDM hardware, Buy the echo canceller. It's worth it. It's not required for standard SIP calls though, only when dealing with analog or TDM circuits. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 18-Nov-09, at 12:39 PM, Tim Uckun wrote: > I am about to build a new machine as a VOIP server. I am going to get > either a quad core intel or a six core AMD processor with at least > eight gigabytes of RAM in it. Given that much horsepower I am > wondering if there is any need to purchase hardware with echo > cancellation (I am thinking about redfone devices).. I can save some > money by not getting the echo cancellation. > > So is it worth saving that money? Is it always better to have hardware > echo cancellation? Is a quad core capable of dealing with echo > cancellation needs of an IVR which is going to take lots of > simultaneous calls? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sicfslist at gmail.com Wed Nov 18 13:00:09 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 18 Nov 2009 15:00:09 -0600 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> Message-ID: <4B046059.3090104@gmail.com> Given that it's an IVR system I think you'll find the DTMF results better with hardware based echo cans ... and to be frank the hardware based echo cancellations are really not that much more expensive on cards from folks like Sangoma. SDR Tim Uckun wrote: > I am about to build a new machine as a VOIP server. I am going to get > either a quad core intel or a six core AMD processor with at least > eight gigabytes of RAM in it. Given that much horsepower I am > wondering if there is any need to purchase hardware with echo > cancellation (I am thinking about redfone devices).. I can save some > money by not getting the echo cancellation. > > So is it worth saving that money? Is it always better to have hardware > echo cancellation? Is a quad core capable of dealing with echo > cancellation needs of an IVR which is going to take lots of > simultaneous calls? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dave at 3c.co.uk Wed Nov 18 13:04:09 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 18 Nov 2009 14:04:09 -0700 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> Message-ID: <1258578249.12820.264.camel@localhost.localdomain> Hi Tim, Here you go: http://old.nabble.com/echo-cancellation-on-PRI-cards-td22552605.html > I am about to build a new machine as a VOIP server. I am going to get > either a quad core intel or a six core AMD processor with at least > eight gigabytes of RAM in it. Given that much horsepower I am > wondering if there is any need to purchase hardware with echo > cancellation (I am thinking about redfone devices).. I can save some > money by not getting the echo cancellation. > > So is it worth saving that money? Is it always better to have hardware > echo cancellation? Is a quad core capable of dealing with echo > cancellation needs of an IVR which is going to take lots of > simultaneous calls? In (very) brief: maybe, no, and depends on the definition of 'lots'. --Dave From nicolas at medularis.com Wed Nov 18 13:13:33 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Wed, 18 Nov 2009 18:13:33 -0300 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> Message-ID: <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> I had a voip provider which wouldn't accept calls from Freeswitch because of the user-agent string. I had to change it to "Asterisk" and then everything worked. Nico On Wed, Nov 18, 2009 at 12:21 PM, Brian West wrote: > you do realize that is NOT the purpose of the user-agent string... changing > might break things in some people's configs due to some assumptions made > about the user agent on the far side for interop purposes... its your choice > to change it but it servers NO purpose doing so. > > /b > > On Nov 18, 2009, at 9:17 AM, Ujjval Karihaloo wrote: > > Not sure I am the only one changing *User-Agent*?.but I just want a way > for our Customers to know the purpose of the server when they talk to it. > There is FreeSwitch written into the SDP ?o? line as well?which I don?t care > about, I want to have something in there that identifies the purpose of the > server. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/bfd09469/attachment-0002.html From brian at freeswitch.org Wed Nov 18 13:20:37 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 15:20:37 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> Message-ID: Sounds like you need to take a baseball bat to their forehead. /b On Nov 18, 2009, at 3:13 PM, Nicolas Brenner wrote: > I had a voip provider which wouldn't accept calls from Freeswitch > because of the user-agent string. I had to change it to "Asterisk" > and then everything worked. > > Nico From rob4manhere at gmail.com Wed Nov 18 13:30:32 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Wed, 18 Nov 2009 15:30:32 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> Message-ID: lol! we have to play nice in the wiki but the mailing list is another story. On Nov 18, 2009, at 3:20 PM, Brian West wrote: > Sounds like you need to take a baseball bat to their forehead. > > /b > > On Nov 18, 2009, at 3:13 PM, Nicolas Brenner wrote: > >> I had a voip provider which wouldn't accept calls from Freeswitch >> because of the user-agent string. I had to change it to "Asterisk" >> and then everything worked. >> >> Nico > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lists at tigertech.com Wed Nov 18 13:33:30 2009 From: lists at tigertech.com (Robert L Mathews) Date: Wed, 18 Nov 2009 13:33:30 -0800 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> References: <4B032142.1000308@tigertech.com> <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> Message-ID: <4B04682A.6000309@tigertech.com> Anthony Minessale wrote: > Have you tried SVN trunk? No, I haven't. I'll try it. (But also, if others want to try seeing if it happens, it's trivially duplicated by calling the echo test app and sending the freeswitch server process a "-STOP" signal, sleeping for a second, then sending it a "-CONT" signal.) In any case, though, I have partially found the cause of the problem -- at least in the echo test module in 1.0.4. It's the same problem reported here: http://www.mail-archive.com/freeswitch-users%40lists.freeswitch.org/msg15800.html The two suggestions there, explicitly setting "rtp-autoflush-during-bridge" true and "rtp_timer_name=none", didn't fix it for me. (The first is no surprise because that's the default anyway.) However, setting the (undocumented?) parameter "rtp-autoflush" to true *did* fix it in the echo test (but not the conference). I think what's happening is that FreeSWITCH contains code to detect when we've "fallen behind" the RTP audio. It looks like this happens in rtp_common_read() of switch_rtp.c: if the code detects that there are "extra" RTP packets waiting during several consecutive rtp_common_read() calls, switch_core_timer_sync() is called to catch up. This code is apparently used during bridged calls when "rtp-autoflush-during-bridge" is true (the default), which explains why I don't have the problem during SIP-to-SIP calls. However, I'm guessing that echo test calls somehow aren't considered "bridged" by that code. Therefore the code isn't being used unless "rtp-autoflush" is true. That thread suggests that this is probably a phone or network problem, but it seems to me that even if all the timing is perfect, this problem will happen if the freeswitch server thread doesn't get enough CPU time to retrieve a packet before the next one arrives. For example, if packets arrive every 20 ms but high load on the server causes the process to sleep for 25 ms, so that two packets are waiting the next time the process runs, it will never catch up that extra packet -- the echo test or conference will now be permanently 20 ms behind. And if that happens again, it will now be 40 ms behind, and so on. That explains why the lag slowly increases over time. Does that make sense? I don't quite understand why the "catch up" code isn't always used for all RTP streams: if an RTP packet poll repeatedly shows that there are extra audio packets waiting, it seems to make sense that we'd always want to catch up. Anyway, as I said, I'm still having the conference problem, even with "rtp-autoflush" enabled. So I need to try the svn trunk version and see if it still happens, then track it down further if so. I will report back. -- Robert L Mathews, Tiger Technologies From timuckun at gmail.com Wed Nov 18 13:36:12 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 19 Nov 2009 10:36:12 +1300 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258578249.12820.264.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> Message-ID: <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> On Thu, Nov 19, 2009 at 10:04 AM, David Knell wrote: > Hi Tim, > > Here you go: > http://old.nabble.com/echo-cancellation-on-PRI-cards-td22552605.html > Thanks. That's almost exactly the same situation as the one I am going to find myself in. > > In (very) brief: maybe, no, and depends on the definition of 'lots'. > By lots I mean somewhere between 50 to a 100 but it's mostly an IVR application so all it will be doing is either playing prompts or recording messages. Almost no live conversations. From msc at freeswitch.org Wed Nov 18 13:36:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Nov 2009 13:36:34 -0800 Subject: [Freeswitch-users] mod_fifo and multi-tenancy In-Reply-To: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> References: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> Message-ID: <87f2f3b90911181336w2b0f6cb8s211bc235a6f9084d@mail.gmail.com> On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi wrote: > About mod_fifo, it would be safe to use it in multi-tenancy scenarios where > domains are created and deleted all the time and in consequence, fifos are > created all the time? I mean, fifos are eventually destroyed by mod_fifo > itself. Is this correct? > > br, > takeshi > > No, FIFOs are not "destroyed" automatically just because the last member goes away. Tony says that an empty FIFO takes up almost no memory so performance shouldn't be an issue. You can always issue the API command: *fifo reparse del_all* to clean everything out if you feel like things are getting out of hand. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/5d38dbaa/attachment-0002.html From msc at freeswitch.org Wed Nov 18 13:39:33 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Nov 2009 13:39:33 -0800 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> Message-ID: <87f2f3b90911181339w114f9bffl7837e03d30de1cf4@mail.gmail.com> On Wed, Nov 18, 2009 at 1:36 PM, Tim Uckun wrote: > On Thu, Nov 19, 2009 at 10:04 AM, David Knell wrote: > > Hi Tim, > > > > Here you go: > > http://old.nabble.com/echo-cancellation-on-PRI-cards-td22552605.html > > > Thanks. That's almost exactly the same situation as the one I am going > to find myself in. > > > > > In (very) brief: maybe, no, and depends on the definition of 'lots'. > > > > By lots I mean somewhere between 50 to a 100 but it's mostly an IVR > application so all it will be doing is either playing prompts or > recording messages. Almost no live conversations. > > Just get the HW echo can - all the COOL kids are doing it! :P -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/249b92cb/attachment-0002.html From brian at freeswitch.org Wed Nov 18 14:05:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 16:05:12 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> Message-ID: <51686C79-3122-4B3B-AD06-F9CEF175E023@freeswitch.org> Well this is a bit more informal vs the wiki where people take it as fact! :) Plus its a little humpday humor! /b On Nov 18, 2009, at 3:30 PM, Rob Forman wrote: > lol! > > we have to play nice in the wiki but the mailing list is another > story. > > > On Nov 18, 2009, at 3:20 PM, Brian West wrote: > >> Sounds like you need to take a baseball bat to their forehead. >> >> /b From hads at nice.net.nz Wed Nov 18 14:06:17 2009 From: hads at nice.net.nz (Hadley Rich) Date: Thu, 19 Nov 2009 11:06:17 +1300 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258578249.12820.264.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> Message-ID: <1258581977.6280.5.camel@sodium> On Wed, 2009-11-18 at 14:04 -0700, David Knell wrote: > In (very) brief: maybe, no, and depends on the definition of 'lots'. Most people say yes. -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier From dfansler at dv-fansler.com Wed Nov 18 13:49:02 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Wed, 18 Nov 2009 16:49:02 -0500 Subject: [Freeswitch-users] APT Utility Message-ID: <005a01ca6898$f16d99d0$d448cd70$@com> Greetings - I am trying to startup a freeSwitch on a P4 running Ubuntu 9.04 "Jaunty". I know very little about Linux. I decided to try this after reading the article in Linux Pro Magazine. I have been following the detailed instructions in the wiki for using Ubuntu Jaunty, however I have run into an unknown - "Use your favorite APT utility to get the needed packages". I am good at following direct instructions - but this statement is too vague for my minimal minimal - did I mention minimal - knowledge of Linux. Could someone please give me detailed instructions on how to use APT utility to get the needed packages - and what are the needed packages? Thanks kindly, David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/e357620b/attachment-0002.html From rob4manhere at gmail.com Wed Nov 18 14:52:51 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Wed, 18 Nov 2009 16:52:51 -0600 Subject: [Freeswitch-users] APT Utility In-Reply-To: <005a01ca6898$f16d99d0$d448cd70$@com> References: <005a01ca6898$f16d99d0$d448cd70$@com> Message-ID: Hi David, When using Apt, you would install packages with: apt-get install Or search for packages with apt-cache search If you're not root, you'll need to stick "sudo " in front of those command. Honestly, you might want to find a better tutorial with explicit command-by-command instructions. Good luck! Rob On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: > Greetings ? I am trying to startup a freeSwitch on a P4 running > Ubuntu 9.04 ?Jaunty?. I know very little about Linux. I decided to > try this after reading the article in Linux Pro Magazine. I have > been following the detailed instructions in the wiki for using > Ubuntu Jaunty, however I have run into an unknown ? ?Use your > favorite APT utility to get the needed packages?. > I am good at following direct instructions ? but this statement is > too vague for my minimal minimal ? did I mention minimal - knowledge > of Linux. > > Could someone please give me detailed instructions on how to use APT > utility to get the needed packages ? and what are the needed packages? > Thanks kindly, > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/e3da4325/attachment-0002.html From sicfslist at gmail.com Wed Nov 18 14:58:05 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 18 Nov 2009 16:58:05 -0600 Subject: [Freeswitch-users] APT Utility In-Reply-To: <005a01ca6898$f16d99d0$d448cd70$@com> References: <005a01ca6898$f16d99d0$d448cd70$@com> Message-ID: <4B047BFD.8000007@gmail.com> David, apt is the pack management system for the debian line of distros (including Ubuntu). Google apt-get or aptitude for more information on the utility. This probably isn't the best list for topics not related to FS (apt is a linux utility) ... but here is a brief rundown: apt-get install make flex patch gcc g++ autoconf automake libtool libncurses5-dev ncurses-dev python-MySQLdb subversion -y cd /usr/src ## DO ONE OF THE FOLLOWING TO GET THE SRC (TRUNK IS BEST FOR NOW -- NO STABLE RELEASE)## svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk or svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch cd freeswitch/ ./bootstrap.sh ./configure make make install Type that as you see it on the command line (do sudo -i and type in the root password first). I will say that FS requires some basic knowledge of Linux to get running ... and certainly to manage / maintain. Try digging around for some Ubuntu how to's for some basic info. Hope this helps. SDR David V. Fansler wrote: > > Greetings ? I am trying to startup a freeSwitch on a P4 running Ubuntu > 9.04 ?Jaunty?. I know very little about Linux. I decided to try this > after reading the article in Linux Pro Magazine. I have been following > the detailed instructions in the wiki for using Ubuntu Jaunty, however > I have run into an unknown ? ?Use your favorite *APT* utility to get > the needed packages?. > > I am good at following direct instructions ? but this statement is too > vague for my minimal minimal ? did I mention minimal - knowledge of Linux. > > Could someone please give me detailed instructions on how to use APT > utility to get the needed packages ? and what are the needed packages? > > Thanks kindly, > > David > > David V. Fansler > > s/v Annabelle > > dfansler at dv-fansler.com > > www.dv-fansler.com > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Wed Nov 18 15:10:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Nov 2009 17:10:27 -0600 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <51686C79-3122-4B3B-AD06-F9CEF175E023@freeswitch.org> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> <51686C79-3122-4B3B-AD06-F9CEF175E023@freeswitch.org> Message-ID: <191c3a030911181510i4c8d36brc03fa4063b088c93@mail.gmail.com> maybe you could send them 183 then 4 180's or send them an invite and pretend to deadlock and not send any more sip traffic as a way of identifying yourself On Wed, Nov 18, 2009 at 4:05 PM, Brian West wrote: > Well this is a bit more informal vs the wiki where people take it as > fact! :) Plus its a little humpday humor! > > /b > > On Nov 18, 2009, at 3:30 PM, Rob Forman wrote: > > > lol! > > > > we have to play nice in the wiki but the mailing list is another > > story. > > > > > > On Nov 18, 2009, at 3:20 PM, Brian West wrote: > > > >> Sounds like you need to take a baseball bat to their forehead. > >> > >> /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/3e07c844/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 18 15:28:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Nov 2009 17:28:30 -0600 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <4B04682A.6000309@tigertech.com> References: <4B032142.1000308@tigertech.com> <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> <4B04682A.6000309@tigertech.com> Message-ID: <191c3a030911181528j7a38ce32gb2fc6fdd585932a9@mail.gmail.com> I can promise you that much of your problems will be solved with latest SVN. There was a small change in the timer sync that was causing your symptoms in specific cases and it's been resolved for months. Did you answer the question about what phones? I'm going to guess Cisco based on the symptoms. non bridge calls use a timer to make sure the audio is coming in at a steady rate to ensure bursting RTP is played at the correct rate. Stopping it and restarting 2 seconds later will cause a delay by design because you have suspended the process but not the UDP stack. If you don't want to use rtp-timer you can disable it in the profile settings for rtp-timer-name by setting it to "none" or the channel variable rtp_timer_name=none on a per call basis, this is not necessarily recommended for everyone because it's a blocking read on the rtp socket which is usually undesirable in an asynchronous thing like a phone call. BTW, Conference counts as a bridge because it has 2 threads one for each direction On Wed, Nov 18, 2009 at 3:33 PM, Robert L Mathews wrote: > Anthony Minessale wrote: > > Have you tried SVN trunk? > > No, I haven't. I'll try it. (But also, if others want to try seeing if > it happens, it's trivially duplicated by calling the echo test app and > sending the freeswitch server process a "-STOP" signal, sleeping for a > second, then sending it a "-CONT" signal.) > > In any case, though, I have partially found the cause of the problem -- > at least in the echo test module in 1.0.4. It's the same problem > reported here: > > > http://www.mail-archive.com/freeswitch-users%40lists.freeswitch.org/msg15800.html > > The two suggestions there, explicitly setting > "rtp-autoflush-during-bridge" true and "rtp_timer_name=none", didn't fix > it for me. (The first is no surprise because that's the default anyway.) > > However, setting the (undocumented?) parameter "rtp-autoflush" to true > *did* fix it in the echo test (but not the conference). > > I think what's happening is that FreeSWITCH contains code to detect when > we've "fallen behind" the RTP audio. It looks like this happens in > rtp_common_read() of switch_rtp.c: if the code detects that there are > "extra" RTP packets waiting during several consecutive rtp_common_read() > calls, switch_core_timer_sync() is called to catch up. > > This code is apparently used during bridged calls when > "rtp-autoflush-during-bridge" is true (the default), which explains why > I don't have the problem during SIP-to-SIP calls. > > However, I'm guessing that echo test calls somehow aren't considered > "bridged" by that code. Therefore the code isn't being used unless > "rtp-autoflush" is true. > > That thread suggests that this is probably a phone or network problem, > but it seems to me that even if all the timing is perfect, this problem > will happen if the freeswitch server thread doesn't get enough CPU time > to retrieve a packet before the next one arrives. For example, if > packets arrive every 20 ms but high load on the server causes the > process to sleep for 25 ms, so that two packets are waiting the next > time the process runs, it will never catch up that extra packet -- the > echo test or conference will now be permanently 20 ms behind. And if > that happens again, it will now be 40 ms behind, and so on. That > explains why the lag slowly increases over time. > > Does that make sense? I don't quite understand why the "catch up" code > isn't always used for all RTP streams: if an RTP packet poll repeatedly > shows that there are extra audio packets waiting, it seems to make sense > that we'd always want to catch up. > > Anyway, as I said, I'm still having the conference problem, even with > "rtp-autoflush" enabled. So I need to try the svn trunk version and see > if it still happens, then track it down further if so. I will report back. > > -- > Robert L Mathews, Tiger Technologies > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/39ece30a/attachment-0002.html From dave at 3c.co.uk Wed Nov 18 15:39:02 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 18 Nov 2009 16:39:02 -0700 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> Message-ID: <1258587542.12820.275.camel@localhost.localdomain> Hi Tim, > > In (very) brief: maybe, no, and depends on the definition of 'lots'. > > > > By lots I mean somewhere between 50 to a 100 but it's mostly an IVR > application so all it will be doing is either playing prompts or > recording messages. Almost no live conversations. For the sort of box you're talking about (quad core++), this isn't lots; it's hardly any.. --Dave From brian at freeswitch.org Wed Nov 18 15:47:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Nov 2009 17:47:04 -0600 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258587542.12820.275.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> Message-ID: <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> It just doesn't belong in user space or kernel space in the machine for true performance you should do it in hardware... I'm pretty sure the poor box would die if you tried it on 32 E1's at the same time. /b On Nov 18, 2009, at 5:39 PM, David Knell wrote: > For the sort of box you're talking about (quad core++), this isn't > lots; > it's hardly any.. > > --Dave From anthony.minessale at gmail.com Wed Nov 18 15:47:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Nov 2009 17:47:51 -0600 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258587542.12820.275.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> Message-ID: <191c3a030911181547t7c92f306l4d8c1f2920b28688@mail.gmail.com> The important thing is that if you are using wanpipe native interface there is no software echo canceler. Sangoma only supports a hardware echo canceler so its not a matter of the cpu compensating for it, if you don't get the cards with the echo can you won't have any unless you run the card in zaptel mode, something we have aspired to avoid. On Wed, Nov 18, 2009 at 5:39 PM, David Knell wrote: > Hi Tim, > > > > In (very) brief: maybe, no, and depends on the definition of 'lots'. > > > > > > > By lots I mean somewhere between 50 to a 100 but it's mostly an IVR > > application so all it will be doing is either playing prompts or > > recording messages. Almost no live conversations. > > For the sort of box you're talking about (quad core++), this isn't lots; > it's hardly any.. > > --Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/aa4e8ff4/attachment-0002.html From mayamatakeshi at gmail.com Wed Nov 18 17:09:37 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Thu, 19 Nov 2009 10:09:37 +0900 Subject: [Freeswitch-users] mod_fifo and multi-tenancy In-Reply-To: <87f2f3b90911181336w2b0f6cb8s211bc235a6f9084d@mail.gmail.com> References: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> <87f2f3b90911181336w2b0f6cb8s211bc235a6f9084d@mail.gmail.com> Message-ID: <15b9404e0911181709i1057e2co9a06c24e2db1d267@mail.gmail.com> On Thu, Nov 19, 2009 at 6:36 AM, Michael Collins wrote: > > > On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi wrote: > >> About mod_fifo, it would be safe to use it in multi-tenancy scenarios >> where domains are created and deleted all the time and in consequence, fifos >> are created all the time? I mean, fifos are eventually destroyed by mod_fifo >> itself. Is this correct? >> >> br, >> takeshi >> >> > No, FIFOs are not "destroyed" automatically just because the last member > goes away. Tony says that an empty FIFO takes up almost no memory so > performance shouldn't be an issue. You can always issue the API command: *fifo > reparse del_all* to clean everything out if you feel like things are > getting out of hand. > Thanks, I have updated the mod_fifo wiki page. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/d1e43720/attachment-0002.html From msc at freeswitch.org Wed Nov 18 17:45:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Nov 2009 17:45:45 -0800 Subject: [Freeswitch-users] mod_fifo and multi-tenancy In-Reply-To: <15b9404e0911181709i1057e2co9a06c24e2db1d267@mail.gmail.com> References: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> <87f2f3b90911181336w2b0f6cb8s211bc235a6f9084d@mail.gmail.com> <15b9404e0911181709i1057e2co9a06c24e2db1d267@mail.gmail.com> Message-ID: <87f2f3b90911181745n15b89330wdc14f2c5e1dd4048@mail.gmail.com> On Wed, Nov 18, 2009 at 5:09 PM, mayamatakeshi wrote: > > > On Thu, Nov 19, 2009 at 6:36 AM, Michael Collins wrote: > >> >> >> On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi wrote: >> >>> About mod_fifo, it would be safe to use it in multi-tenancy scenarios >>> where domains are created and deleted all the time and in consequence, fifos >>> are created all the time? I mean, fifos are eventually destroyed by mod_fifo >>> itself. Is this correct? >>> >>> br, >>> takeshi >>> >>> >> No, FIFOs are not "destroyed" automatically just because the last member >> goes away. Tony says that an empty FIFO takes up almost no memory so >> performance shouldn't be an issue. You can always issue the API command: >> *fifo reparse del_all* to clean everything out if you feel like things >> are getting out of hand. >> > > Thanks, > I have updated the mod_fifo wiki page. > > FYI, I was doing some other research and I noticed this on the mod_fifo wiki page: *fifo_destroy_after_use*: FreeSWITCH automatically create a new FIFO when the first time use it, and keep in the memory hash. This var tell FreeSWITCH destroy it to save memory. Using for a one time FIFO. So... you do have that option as well. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/4230bb05/attachment-0002.html From mcampbellsmith at gmail.com Wed Nov 18 18:36:52 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 19 Nov 2009 13:36:52 +1100 Subject: [Freeswitch-users] Call from Secure RTP to non-secure RTP Message-ID: <33c87fa30911181836i75ec2945gc7e5782b38c14415@mail.gmail.com> Hi! How do I setup FS so that placing a call from an extension that only support SRTP (1002) to an extension that only supports RTP (1000)? I put this dialstring, from the wiki http://wiki.freeswitch.org/wiki/Tls, into the users xml file under directory/default I have also put a when 1000 is dialing 1002. However I never see crytpo sent in the RTP to 1002 and it responds with Bad Security Level What have I missed? Thanks From jim at evolutiontel.net Wed Nov 18 21:12:35 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 19 Nov 2009 16:12:35 +1100 Subject: [Freeswitch-users] Call from Secure RTP to non-secure RTP In-Reply-To: <33c87fa30911181836i75ec2945gc7e5782b38c14415@mail.gmail.com> References: <33c87fa30911181836i75ec2945gc7e5782b38c14415@mail.gmail.com> Message-ID: Does 1002 use TLS to transport SIP signalling? My experience is that TLS is required on some phones otherwise they will not do srtp and will reply with the responce you have mentioned. Sent from my iPhone On 19/11/2009, at 1:36 PM, Mark Campbell-Smith wrote: > Hi! > > How do I setup FS so that placing a call from an extension that only > support SRTP (1002) to an extension that only supports RTP (1000)? > > I put this dialstring, from the wiki > http://wiki.freeswitch.org/wiki/Tls, into the users xml file under > directory/default > > value="{sip_secure_media=${regex(${sofia_contact(${dialed_user}@$ > {dialed_domain})}|transport=tls)}, > presence_id=${dialed_user}@${dialed_domain}}${sofia_contact($ > {dialed_user}@${dialed_domain})}" > /> > > I have also put a data="sip_secure_media=true"/> when 1000 is dialing 1002. > > > > > > > However I never see crytpo sent in the RTP to 1002 and it responds > with Bad Security Level > > What have I missed? > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From dujinfang at gmail.com Wed Nov 18 21:22:04 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 19 Nov 2009 13:22:04 +0800 Subject: [Freeswitch-users] mod_fifo and multi-tenancy In-Reply-To: <87f2f3b90911181745n15b89330wdc14f2c5e1dd4048@mail.gmail.com> References: <15b9404e0911180832g6930f08k9c0f6dbe3b4e54b@mail.gmail.com> <87f2f3b90911181336w2b0f6cb8s211bc235a6f9084d@mail.gmail.com> <15b9404e0911181709i1057e2co9a06c24e2db1d267@mail.gmail.com> <87f2f3b90911181745n15b89330wdc14f2c5e1dd4048@mail.gmail.com> Message-ID: <23f91030911182122h4b6c94b9hab74132425b6b006@mail.gmail.com> I once wrote a patch for "fifo delete", but didn't submit to jira. If someone think it's useful to merge into trunk, I think I can still find the code, but sure need to test with the current trunk. 2009/11/19 Michael Collins > > > On Wed, Nov 18, 2009 at 5:09 PM, mayamatakeshi wrote: > >> >> >> On Thu, Nov 19, 2009 at 6:36 AM, Michael Collins wrote: >> >>> >>> >>> On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi wrote: >>> >>>> About mod_fifo, it would be safe to use it in multi-tenancy scenarios >>>> where domains are created and deleted all the time and in consequence, fifos >>>> are created all the time? I mean, fifos are eventually destroyed by mod_fifo >>>> itself. Is this correct? >>>> >>>> br, >>>> takeshi >>>> >>>> >>> No, FIFOs are not "destroyed" automatically just because the last member >>> goes away. Tony says that an empty FIFO takes up almost no memory so >>> performance shouldn't be an issue. You can always issue the API command: >>> *fifo reparse del_all* to clean everything out if you feel like things >>> are getting out of hand. >>> >> >> Thanks, >> I have updated the mod_fifo wiki page. >> >> > FYI, I was doing some other research and I noticed this on the mod_fifo > wiki page: > > *fifo_destroy_after_use*: FreeSWITCH automatically create a new FIFO when > the first time use it, and keep in the memory hash. This var tell FreeSWITCH > destroy it to save memory. Using for a one time FIFO. > > So... you do have that option as well. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/432e1b76/attachment-0002.html From mcampbellsmith at gmail.com Wed Nov 18 21:22:32 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 19 Nov 2009 16:22:32 +1100 Subject: [Freeswitch-users] Call from Secure RTP to non-secure RTP In-Reply-To: References: <33c87fa30911181836i75ec2945gc7e5782b38c14415@mail.gmail.com> Message-ID: <33c87fa30911182122g79354754t49d16f35db1f0d26@mail.gmail.com> Thanks Jim, Yep, 1002 does TLS and SRTP, 1000 does UDP and RTP. Cheers On Thu, Nov 19, 2009 at 4:12 PM, Jim Burke wrote: > Does 1002 use TLS to transport SIP signalling? My experience is that > TLS is required on some phones otherwise they will not do srtp and > will reply with the responce you have mentioned. > > Sent from my iPhone > > On 19/11/2009, at 1:36 PM, Mark Campbell-Smith > wrote: > >> Hi! >> >> How do I setup FS so that placing a call from an extension that only >> support SRTP (1002) to an extension that only supports RTP (1000)? >> >> I put this dialstring, from the wiki >> http://wiki.freeswitch.org/wiki/Tls, into the users xml file under >> directory/default >> >> > value="{sip_secure_media=${regex(${sofia_contact(${dialed_user}@$ >> {dialed_domain})}|transport=tls)}, >> ?presence_id=${dialed_user}@${dialed_domain}}${sofia_contact($ >> {dialed_user}@${dialed_domain})}" >> /> >> >> I have also put a > data="sip_secure_media=true"/> when 1000 is dialing 1002. >> ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> >> However I never see crytpo sent in the RTP to 1002 and it responds >> with Bad Security Level >> >> What have I missed? >> >> Thanks >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Wed Nov 18 21:28:39 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 19 Nov 2009 13:28:39 +0800 Subject: [Freeswitch-users] Changing User-Agent String In-Reply-To: <191c3a030911181510i4c8d36brc03fa4063b088c93@mail.gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4319@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC437C@EXMBXCLUS01.citservers.local> <21DDC28F-6D0D-4A5E-81FA-C4BAF9F91761@freeswitch.org> <1b46b4e80911181313x466fe26ek5ec0134310d95bfe@mail.gmail.com> <51686C79-3122-4B3B-AD06-F9CEF175E023@freeswitch.org> <191c3a030911181510i4c8d36brc03fa4063b088c93@mail.gmail.com> Message-ID: <23f91030911182128g11441fcasf9aff6e49e997f3@mail.gmail.com> lol: 2009/11/19 Anthony Minessale > maybe you could send them 183 then 4 180's or send them an invite and > pretend to deadlock and not send any more sip traffic as a way of > identifying yourself > > On Wed, Nov 18, 2009 at 4:05 PM, Brian West wrote: > >> Well this is a bit more informal vs the wiki where people take it as >> fact! :) Plus its a little humpday humor! >> >> /b >> >> On Nov 18, 2009, at 3:30 PM, Rob Forman wrote: >> >> > lol! >> > >> > we have to play nice in the wiki but the mailing list is another >> > story. >> > >> > >> > On Nov 18, 2009, at 3:20 PM, Brian West wrote: >> > >> >> Sounds like you need to take a baseball bat to their forehead. >> >> >> >> /b >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/b56af2c2/attachment-0002.html From ujjval at simplesignal.com Wed Nov 18 21:34:02 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Wed, 18 Nov 2009 21:34:02 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> I have used the following setting in ivr.conf.xml to setup conferencing with moderator. However, the issue I have is - the user enters 123456 and then say if it's a moderator they enter wrong Moderator PIN 3 times then it takes the user back to the main menu..."conference_menu" and asks for main conf pin (123456) once again. I would like the caller to be disconnected if they get into the Moderator menu and enter wrong Moderator PIN 3 times. Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO? 80112 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Thursday, November 05, 2009 7:52 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hi UK, From what I've done and read, the caller-controls (in conference.conf.xml) can be modified to almost anything you can think of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a caller control just for those with the moderator flag. So unless you want everyone able to mute/kick everyone then you can't do it. The wiki seems to indicate this as well: "Be aware that the caller-controls are applied across the entire conference. You cannot enter one member of the conference using caller- controls ABC and then enter a second member using caller-controls XYZ." http://wiki.freeswitch.org/wiki/Mod_conference I think this might be a limitation of mod_conference. Perhaps one of the pros can chime in if I'm off-base or there's some nifty way to accomplish this. Cheers, Rob On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > Any ideas on the below...has anyone implemented the below: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Ujjval Karihaloo > Sent: Monday, November 02, 2009 12:52 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Rob: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 9:34 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hm, strange. I haven't seen that before. Can you pastebin your logs > at debug level? > > On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > >> It's strange... a tcpdump tells me that there is no DTMF from my >> provider when using IVR, but when I call into a TN that goes >> directly into the Conference App, I see DTMF from the provider. >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 7:23 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I've never had any problem with that. Is your logging at debug level >> so you can see the RECV DTFM in the log/fs_cli? Are you calling from >> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >> isn't passing them through. >> >> Make sure your logging is turned up then try something simpler, like >> calling the echo application, and see if DTFM comes through. >> >> Rob >> >> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >> >>> Rob: >>> >>> For some reason, I don't see the DTMF appear on the fs_CLI when >>> using the below configuration....so it basically timesout. >>> >>> UK >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, October 26, 2009 9:21 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Thx a lot Rob, reading the wiki your way or using IVR seems >>> correct.. >>> =============== >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> ====================== >>> >>> I will try it out. >>> >>> Again thx a lot for your help. Will keep everyone posted. >>> >>> ________________________________________ >>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>> Sent: Friday, October 23, 2009 12:22 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I just re-tested with the pin in my dial plan: >>> >>> >>> >>> And it doesn't challenge me for the pin. I just drop right in. I >>> figured this is how it was intended, since the wiki says the pin is >>> set initially and only challenged in later attempts [by future >>> callers]: >>> >>> "The first time a conference name (confname) is used, it will be >>> created on demand, and the pin will be set to what ever is specified >>> at that time: the pin in the data string if specified, or if not, >>> the >>> "pin" setting in the conference profile, and if that is also >>> unspecified, then there is no pin protection. Any later attempt to >>> join the conference must specify the same pin number, if one existed >>> when it was created. " >>> >>> >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> >>> I'm sure there's a scenario where its used and useful, the wiki just >>> doesn't explain it. >>> >>> Rob >>> >>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>> >>>> Well first off you're not defining a pine here... >>>> >>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>> pin >>>> number] >>>> >>>> That might be why its not asking for a pin. >>>> >>>> /b >>>> >>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From rob4manhere at gmail.com Wed Nov 18 22:02:03 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 19 Nov 2009 00:02:03 -0600 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> Message-ID: <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> Hi again UK, IVR is designed to naturally return to previous or top menus. I don't think there's a way to change this default behavior. Maybe its time to move to a script-based pin validation system so you have the full control you need. http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR Rob On Nov 18, 2009, at 11:34 PM, Ujjval Karihaloo wrote: > I have used the following setting in ivr.conf.xml to setup > conferencing with moderator. > > However, the issue I have is - the user enters 123456 and then say > if it's a moderator they enter wrong Moderator PIN 3 times then it > takes the user back to the main menu..."conference_menu" and asks > for main conf pin (123456) once again. > > I would like the caller to be disconnected if they get into the > Moderator menu and enter wrong Moderator PIN 3 times. > > greet-long="welcome_please_enter_conference_pin.wav" > greet-short="check_and_try_again.wav" > invalid-sound="passcode_invalid.wav" > exit-sound="voicemail/vm-goodbye.wav" > timeout="10000" > inter-digit-timeout="5000" > max-failures="3" > max-timeouts="3" > digit-len="7"> > param="conference_123456_moderator_menu" /> > > > greet- > long > = > "conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant > .wav" > greet-short="check_moderator_pin_or_1_to_join.wav" > invalid-sound="invalid_moderator_pin.wav" > exit-sound="voicemail/vm-goodbye.wav" > timeout="10000" > inter-digit-timeout="5000" > max-failures="3" > max-timeouts="3" > digit-len="5"> > > > > > > > > > Ujjval Karihaloo > VP Voice Engineering > IP Phone: +13032428610 > E-Fax: +17202391690 > > SimpleSignal Inc. > 88 Inverness Circle East > Suite K105 > Englewood, CO 80112 > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Thursday, November 05, 2009 7:52 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hi UK, > > From what I've done and read, the caller-controls (in > conference.conf.xml) can be modified to almost anything you can think > of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a > caller control just for those with the moderator flag. So unless you > want everyone able to mute/kick everyone then you can't do it. > > The wiki seems to indicate this as well: > > "Be aware that the caller-controls are applied across the entire > conference. You cannot enter one member of the conference using > caller- > controls ABC and then enter a second member using caller-controls > XYZ." > > http://wiki.freeswitch.org/wiki/Mod_conference > > > I think this might be a limitation of mod_conference. Perhaps one of > the pros can chime in if I'm off-base or there's some nifty way to > accomplish this. > > Cheers, > Rob > > On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > >> Any ideas on the below...has anyone implemented the below: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, November 02, 2009 12:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Rob: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 9:34 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hm, strange. I haven't seen that before. Can you pastebin your logs >> at debug level? >> >> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >> >>> It's strange... a tcpdump tells me that there is no DTMF from my >>> provider when using IVR, but when I call into a TN that goes >>> directly into the Conference App, I see DTMF from the provider. >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 7:23 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I've never had any problem with that. Is your logging at debug >>> level >>> so you can see the RECV DTFM in the log/fs_cli? Are you calling >>> from >>> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >>> isn't passing them through. >>> >>> Make sure your logging is turned up then try something simpler, like >>> calling the echo application, and see if DTFM comes through. >>> >>> Rob >>> >>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>> >>>> Rob: >>>> >>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>> using the below configuration....so it basically timesout. >>>> >>>> UK >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Ujjval Karihaloo >>>> Sent: Monday, October 26, 2009 9:21 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>> Moderator >>>> >>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>> correct.. >>>> =============== >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> ====================== >>>> >>>> I will try it out. >>>> >>>> Again thx a lot for your help. Will keep everyone posted. >>>> >>>> ________________________________________ >>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>> Sent: Friday, October 23, 2009 12:22 PM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>> Moderator >>>> >>>> I just re-tested with the pin in my dial plan: >>>> >>>> >>>> >>>> And it doesn't challenge me for the pin. I just drop right in. I >>>> figured this is how it was intended, since the wiki says the pin is >>>> set initially and only challenged in later attempts [by future >>>> callers]: >>>> >>>> "The first time a conference name (confname) is used, it will be >>>> created on demand, and the pin will be set to what ever is >>>> specified >>>> at that time: the pin in the data string if specified, or if not, >>>> the >>>> "pin" setting in the conference profile, and if that is also >>>> unspecified, then there is no pin protection. Any later attempt to >>>> join the conference must specify the same pin number, if one >>>> existed >>>> when it was created. " >>>> >>>> >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> >>>> I'm sure there's a scenario where its used and useful, the wiki >>>> just >>>> doesn't explain it. >>>> >>>> Rob >>>> >>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>> >>>>> Well first off you're not defining a pine here... >>>>> >>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>> pin >>>>> number] >>>>> >>>>> That might be why its not asking for a pin. >>>>> >>>>> /b >>>>> >>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From elihayun at gmail.com Wed Nov 18 22:36:22 2009 From: elihayun at gmail.com (Eli Hayun) Date: Thu, 19 Nov 2009 08:36:22 +0200 Subject: [Freeswitch-users] change event value Message-ID: <4B04E766.8070706@savion.huji.ac.il> Hi Is there is a way to intercept an event (for example : REGISTER) and change one of its parameters (for example: the extension number) and fire up the corrected event? I need it to set the speedial of the phone value to be "**xxxxx" but to make it register as "xxxxx" Thanks Eli From lists at tigertech.com Wed Nov 18 23:09:22 2009 From: lists at tigertech.com (Robert L Mathews) Date: Wed, 18 Nov 2009 23:09:22 -0800 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <191c3a030911181528j7a38ce32gb2fc6fdd585932a9@mail.gmail.com> References: <4B032142.1000308@tigertech.com> <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> <4B04682A.6000309@tigertech.com> <191c3a030911181528j7a38ce32gb2fc6fdd585932a9@mail.gmail.com> Message-ID: <4B04EF22.1030404@tigertech.com> Anthony Minessale wrote: > I can promise you that much of your problems will be solved with > latest SVN. Okay, thanks! And in fact, I tried today's SVN, and it has fixed the problem with the conference, even without setting "rtp-autoflush". Conferences now discard packets and "catch up" when they gets behind, even with only the default "rtp-autoflush-during-bridge" set. The echo test still suffers from the same problem unless "rtp-autoflush" is used, which I assume is simply because it's not considered a bridged call. Eavesdropping on an existing bridged call, then pressing "3" to turn it into a conference call, also requires "rtp-autoflush" to avoid persistent lag on the third leg. > Did you answer the question about what phones? I'm going to guess Cisco > based on the symptoms. It happens with all phones, as far as I can tell. I've tried at least Grandstream GXP2000, Grandstream BT102, SJPhone, Twinkle, and Express Talk (none of them Cisco). I'm fairly positive the problem is unrelated to phones; it's caused by delays in CPU scheduling of the server process. > non bridge calls use a timer to make sure the audio is coming in at a > steady rate to ensure bursting RTP > is played at the correct rate. Stopping it and restarting 2 seconds > later will cause a delay by design because you have suspended the > process but not the UDP stack. Ummm.... well, a copy of FreeSWITCH running on any non-realtime operating system will occasionally not get scheduled for all the CPU time it wants. For example, it wouldn't be unusual for a thread to ask to sleep for 20 milliseconds but actually not wake up for 21, 25, or even 40 milliseconds because the server is busy with other things. Each time that happens, it's a smaller version of my artificial suspend test: the operating system has, of course "suspended the process but not the UDP stack". It doesn't necessarily mean there's bursty network traffic or phone timing issues. Should FreeSWITCH really lag by that much for the rest of the call? 20 milliseconds here, 20 milliseconds there, and pretty soon you're talking about real seconds. I'm assuming the reason for making it catch up on bridged calls, but not non-bridged calls, is that people talking to each other can't tolerate high latency, but people listening to an IVR or something can. But is that still true if it gets seconds behind? And should the third leg of an eavesdrop-converted-to-three-way-call be considered non-bridged for this purpose? Anyway, given that current svn trunk fixes the problem by default in conferences and any other bridged call, I'm satisfied. And if anyone complains about this problem for non-bridged calls, I suppose they can enable "rtp-autoflush" to get the same "catch-up" behavior there. I took a shot at documenting these parameters in the wiki on: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#rtp-autoflush-during-bridge Thanks for the help! -- Robert L Mathews, Tiger Technologies From helmut.kuper at ewetel.de Thu Nov 19 00:33:03 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 19 Nov 2009 09:33:03 +0100 Subject: [Freeswitch-users] Question about odbc support In-Reply-To: <4B043B32.20802@ewetel.de> References: <4B043B32.20802@ewetel.de> Message-ID: <4B0502BF.6050800@ewetel.de> Hello, hm kind of unclear Question. So I'm looking for a way to get the affected number of rows after executing a delete statement via ODBC. There is a function called "SQLRowCount()", but I didn't found a switch_odbc_* function in FS which allows me to call it. On 18.11.2009 19:21, Helmut Kuper wrote: > Hi, > > > does anybody know how to check the affected rows caused by delete, > insert or update sql statements in FS? > > To do this with sqlite3 there is a function called switch_core_db_changes(). > > > regards > helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Mit freundlichen Gr??en Helmut Kuper Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From helmut.kuper at ewetel.de Thu Nov 19 00:33:11 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 19 Nov 2009 09:33:11 +0100 Subject: [Freeswitch-users] Question about odbc support In-Reply-To: <4B043B32.20802@ewetel.de> References: <4B043B32.20802@ewetel.de> Message-ID: <4B0502C7.1060105@ewetel.de> Hello, hm kind of unclear Question. So I'm looking for a way to get the affected number of rows after executing a delete statement via ODBC. There is a function called "SQLRowCount()", but I didn't found a switch_odbc_* function in FS which allows me to call it. On 18.11.2009 19:21, Helmut Kuper wrote: > Hi, > > > does anybody know how to check the affected rows caused by delete, > insert or update sql statements in FS? > > To do this with sqlite3 there is a function called switch_core_db_changes(). > > > regards > helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From woodydickson at gmail.com Thu Nov 19 00:40:58 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Thu, 19 Nov 2009 16:40:58 +0800 Subject: [Freeswitch-users] store registration info in memcache Message-ID: Hi, Is there anyway to store registration info in memcache instead of sqlite? By doing that, it is possible for multiple freeswitch to share the same user registration info. Is there anyway I can intercept the registration success/failure event and write stuff to memcache myself? thanks, woody -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/623d720c/attachment-0002.html From samuelmukoti at gmail.com Thu Nov 19 00:48:03 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Thu, 19 Nov 2009 10:48:03 +0200 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> Message-ID: <2A0CB328-8C3A-4CD0-B2EC-D6952E7539C0@gmail.com> Greetings, I'm a new freeswitch user and am wondering what people do when setting options in the freeswitch config files. Do people use special tools, XML editors etc or is it just vi/emacs/Kate? I'm a developer and was thinking of putting together a small editor to manage my freeswitch server, something like freepbx, I know it's not an easy undertaking but I'm sure it's well worth it. Regards, Samuel Mukoti CEO Melivo Business Systems Mobile: +263912739405 Email: sam at melivo.com Skype: samuelmukoti Twitter: twitter.com/samuelmukoti On 19 Nov,2009, at 8:02 AM, Rob Forman wrote: > Hi again UK, > > IVR is designed to naturally return to previous or top menus. I don't > think there's a way to change this default behavior. Maybe its time > to move to a script-based pin validation system so you have the full > control you need. > > http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR > > Rob > > On Nov 18, 2009, at 11:34 PM, Ujjval Karihaloo wrote: > >> I have used the following setting in ivr.conf.xml to setup >> conferencing with moderator. >> >> However, the issue I have is - the user enters 123456 and then say >> if it's a moderator they enter wrong Moderator PIN 3 times then it >> takes the user back to the main menu..."conference_menu" and asks >> for main conf pin (123456) once again. >> >> I would like the caller to be disconnected if they get into the >> Moderator menu and enter wrong Moderator PIN 3 times. >> >> > greet-long="welcome_please_enter_conference_pin.wav" >> greet-short="check_and_try_again.wav" >> invalid-sound="passcode_invalid.wav" >> exit-sound="voicemail/vm-goodbye.wav" >> timeout="10000" >> inter-digit-timeout="5000" >> max-failures="3" >> max-timeouts="3" >> digit-len="7"> >> > param="conference_123456_moderator_menu" /> >> >> >> > greet- >> long >> = >> "conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant >> .wav" >> greet-short="check_moderator_pin_or_1_to_join.wav" >> invalid-sound="invalid_moderator_pin.wav" >> exit-sound="voicemail/vm-goodbye.wav" >> timeout="10000" >> inter-digit-timeout="5000" >> max-failures="3" >> max-timeouts="3" >> digit-len="5"> >> >> >> >> >> >> >> >> >> Ujjval Karihaloo >> VP Voice Engineering >> IP Phone: +13032428610 >> E-Fax: +17202391690 >> >> SimpleSignal Inc. >> 88 Inverness Circle East >> Suite K105 >> Englewood, CO 80112 >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Thursday, November 05, 2009 7:52 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hi UK, >> >> From what I've done and read, the caller-controls (in >> conference.conf.xml) can be modified to almost anything you can think >> of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a >> caller control just for those with the moderator flag. So unless you >> want everyone able to mute/kick everyone then you can't do it. >> >> The wiki seems to indicate this as well: >> >> "Be aware that the caller-controls are applied across the entire >> conference. You cannot enter one member of the conference using >> caller- >> controls ABC and then enter a second member using caller-controls >> XYZ." >> >> http://wiki.freeswitch.org/wiki/Mod_conference >> >> >> I think this might be a limitation of mod_conference. Perhaps one of >> the pros can chime in if I'm off-base or there's some nifty way to >> accomplish this. >> >> Cheers, >> Rob >> >> On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: >> >>> Any ideas on the below...has anyone implemented the below: >>> >>> Once I have the Moderator and Participants logged on, how do I >>> invoke the moderator previlidges, LIk esay muting everyone/someone >>> or kicking someone out of the Conf and the like? >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, November 02, 2009 12:52 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Rob: >>> >>> Once I have the Moderator and Participants logged on, how do I >>> invoke the moderator previlidges, LIk esay muting everyone/someone >>> or kicking someone out of the Conf and the like? >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 9:34 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Hm, strange. I haven't seen that before. Can you pastebin your >>> logs >>> at debug level? >>> >>> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >>> >>>> It's strange... a tcpdump tells me that there is no DTMF from my >>>> provider when using IVR, but when I call into a TN that goes >>>> directly into the Conference App, I see DTMF from the provider. >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman >>>> Sent: Friday, October 30, 2009 7:23 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>> Moderator >>>> >>>> I've never had any problem with that. Is your logging at debug >>>> level >>>> so you can see the RECV DTFM in the log/fs_cli? Are you calling >>>> from >>>> a SIP phone on the pbx, or via a PSTN provider? Maybe your >>>> provider >>>> isn't passing them through. >>>> >>>> Make sure your logging is turned up then try something simpler, >>>> like >>>> calling the echo application, and see if DTFM comes through. >>>> >>>> Rob >>>> >>>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>>> >>>>> Rob: >>>>> >>>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>>> using the below configuration....so it basically timesout. >>>>> >>>>> UK >>>>> >>>>> >>>>> >>>>> -----Original Message----- >>>>> From: freeswitch-users-bounces at lists.freeswitch.org >>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>>> ] On Behalf Of Ujjval Karihaloo >>>>> Sent: Monday, October 26, 2009 9:21 AM >>>>> To: freeswitch-users at lists.freeswitch.org >>>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>>> Moderator >>>>> >>>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>>> correct.. >>>>> =============== >>>>> The wiki also says that the wait-mod might be "used in >>>>> conjunction >>>>> with an IVR where the moderators are authenticated with an extra >>>>> pass- >>>>> code", which is what I did. I guess that's why I didn't >>>>> understand >>>>> the point of the +pin. >>>>> ====================== >>>>> >>>>> I will try it out. >>>>> >>>>> Again thx a lot for your help. Will keep everyone posted. >>>>> >>>>> ________________________________________ >>>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch- >>>>> users-bounces at lists.freeswitch.org >>>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>>> Sent: Friday, October 23, 2009 12:22 PM >>>>> To: freeswitch-users at lists.freeswitch.org >>>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>>> Moderator >>>>> >>>>> I just re-tested with the pin in my dial plan: >>>>> >>>>> >>>>> >>>>> And it doesn't challenge me for the pin. I just drop right in. I >>>>> figured this is how it was intended, since the wiki says the pin >>>>> is >>>>> set initially and only challenged in later attempts [by future >>>>> callers]: >>>>> >>>>> "The first time a conference name (confname) is used, it will be >>>>> created on demand, and the pin will be set to what ever is >>>>> specified >>>>> at that time: the pin in the data string if specified, or if not, >>>>> the >>>>> "pin" setting in the conference profile, and if that is also >>>>> unspecified, then there is no pin protection. Any later attempt to >>>>> join the conference must specify the same pin number, if one >>>>> existed >>>>> when it was created. " >>>>> >>>>> >>>>> The wiki also says that the wait-mod might be "used in >>>>> conjunction >>>>> with an IVR where the moderators are authenticated with an extra >>>>> pass- >>>>> code", which is what I did. I guess that's why I didn't >>>>> understand >>>>> the point of the +pin. >>>>> >>>>> I'm sure there's a scenario where its used and useful, the wiki >>>>> just >>>>> doesn't explain it. >>>>> >>>>> Rob >>>>> >>>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>>> >>>>>> Well first off you're not defining a pine here... >>>>>> >>>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>>> pin >>>>>> number] >>>>>> >>>>>> That might be why its not asking for a pin. >>>>>> >>>>>> /b >>>>>> >>>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>>> freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From jason at jasonjgw.net Thu Nov 19 01:07:09 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 19 Nov 2009 20:07:09 +1100 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <2A0CB328-8C3A-4CD0-B2EC-D6952E7539C0@gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> <2A0CB328-8C3A-4CD0-B2EC-D6952E7539C0@gmail.com> Message-ID: <20091119090709.GA26604@jdc.jasonjgw.net> Samuel Mukoti wrote: > I'm a new freeswitch user and am wondering what people do when setting > options in the freeswitch config files. Do people use special tools, > XML editors etc or is it just vi/emacs/Kate? Emacs has an XML editing mode; Vim may have extensions for handling XML as well. However, I have not found it necessary to invoke the XML features of an editor; just treating the configuration files as plain text is sufficient. From leon at scarlet-internet.nl Thu Nov 19 01:14:22 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 19 Nov 2009 10:14:22 +0100 Subject: [Freeswitch-users] store registration info in memcache In-Reply-To: References: Message-ID: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> Hi, Not that I know of, but you can use odbc to store registrations and share it that way.. regards, Leon On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: > Hi, > > Is there anyway to store registration info in memcache instead of > sqlite? > > By doing that, it is possible for multiple freeswitch to share the > same user registration info. > > Is there anyway I can intercept the registration success/failure > event and write stuff to memcache myself? > > thanks, > woody > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jaybinks at gmail.com Thu Nov 19 01:33:12 2009 From: jaybinks at gmail.com (jay binks) Date: Thu, 19 Nov 2009 19:33:12 +1000 Subject: [Freeswitch-users] store registration info in memcache In-Reply-To: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> References: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> Message-ID: I believe OBDC is the official way.. however id love look at doing this in a higher performance way, without the single point of failure.. local memcache, in front of OBDC or something ?? not 100% sure of it, but just using a single central database is a little bit of a concern in a carrier environment. Jay On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij wrote: > Hi, > > Not that I know of, but you can use odbc to store registrations and > share it that way.. > > regards, > > Leon > > On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: > > > Hi, > > > > Is there anyway to store registration info in memcache instead of > > sqlite? > > > > By doing that, it is possible for multiple freeswitch to share the > > same user registration info. > > > > Is there anyway I can intercept the registration success/failure > > event and write stuff to memcache myself? > > > > thanks, > > woody > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/2852b7b7/attachment-0002.html From leon at scarlet-internet.nl Thu Nov 19 02:07:30 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 19 Nov 2009 11:07:30 +0100 Subject: [Freeswitch-users] store registration info in memcache In-Reply-To: References: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> Message-ID: <72D493D6-F194-495A-8028-41362870305C@scarlet-internet.nl> Well, you can of course easily have a loadbalancer with failover in front of your sql servers and have them replicate to each other. Freeswitch will reconnect if a connection goes down. Perhaps failover is also possible directly through odbc ? Does anyone know if that's possible ? regards, Leon On Nov 19, 2009, at 10:33 AM, jay binks wrote: > I believe OBDC is the official way.. > however id love look at doing this in a higher performance way, > without the single point of failure.. > > local memcache, in front of OBDC or something ?? > > not 100% sure of it, but just using a single central database is a > little bit of a concern in a carrier environment. > > Jay > > > > On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij > wrote: > Hi, > > Not that I know of, but you can use odbc to store registrations and > share it that way.. > > regards, > > Leon > > On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: > > > Hi, > > > > Is there anyway to store registration info in memcache instead of > > sqlite? > > > > By doing that, it is possible for multiple freeswitch to share the > > same user registration info. > > > > Is there anyway I can intercept the registration success/failure > > event and write stuff to memcache myself? > > > > thanks, > > woody > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Sincerely > > Jay > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/55f40424/attachment-0002.html From lon at kickasspixels.com Thu Nov 19 02:23:08 2009 From: lon at kickasspixels.com (Lon Baker) Date: Thu, 19 Nov 2009 02:23:08 -0800 Subject: [Freeswitch-users] store registration info in memcache In-Reply-To: <72D493D6-F194-495A-8028-41362870305C@scarlet-internet.nl> References: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> <72D493D6-F194-495A-8028-41362870305C@scarlet-internet.nl> Message-ID: <5d3e0dc60911190223j8a32cf0m4d7ed2983a47987a@mail.gmail.com> If we could access mod_memcache for registration information that would be ideal and highly robust, since you can share memcache with external applications. Lon On Thu, Nov 19, 2009 at 2:07 AM, Leon de Rooij wrote: > Well, you can of course easily have a loadbalancer with failover in front of > your sql servers and have them replicate to each other.?Freeswitch will > reconnect if a connection goes down. Perhaps failover is also possible > directly through odbc ? Does anyone know if that's possible ? > regards, > Leon > > > On Nov 19, 2009, at 10:33 AM, jay binks wrote: > > I believe OBDC is the official way.. > however id love look at doing this in a higher performance way, without the > single point of failure.. > local memcache, in front of OBDC or something ?? > not 100% sure of it, but just using a single central database is a little > bit of a concern in a carrier?environment. > Jay > > > On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij > wrote: >> >> Hi, >> >> Not that I know of, but you can use odbc to store registrations and >> share it that way.. >> >> regards, >> >> Leon >> >> On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: >> >> > Hi, >> > >> > Is there anyway to store registration info in memcache instead of >> > sqlite? >> > >> > By doing that, it is possible for multiple freeswitch to share the >> > same user registration info. >> > >> > Is there anyway I can intercept the registration success/failure >> > event and write stuff to memcache myself? >> > >> > thanks, >> > woody >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Sincerely > > Jay > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From stevendt at primrosebank.net Thu Nov 19 03:33:46 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 11:33:46 -0000 Subject: [Freeswitch-users] Extension Configuration - XML File Entries for Group configuration Message-ID: <73DD76AE07884A1D9535EF27C5841DAD@bp1.ad.bp.com> Hi, Can someone please help me understand a little more about Group configuration ? I believe that Group Membership is configured in the \conf\directory\default.xml file I've done this and the caller groups seem to work fine. However, each extension in the \conf\directory\default directory, e.g., 111.xml also has an entry for "callgroup" Can someone explain what the difference in these two options is please ? regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/0e1d4b31/attachment-0002.html From rupa at rupa.com Thu Nov 19 04:52:40 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 19 Nov 2009 06:52:40 -0600 Subject: [Freeswitch-users] store registration info in memcache In-Reply-To: <5d3e0dc60911190223j8a32cf0m4d7ed2983a47987a@mail.gmail.com> References: <050B8131-D1E5-4814-9CF1-E01EBDAA57F0@scarlet-internet.nl> <72D493D6-F194-495A-8028-41362870305C@scarlet-internet.nl> <5d3e0dc60911190223j8a32cf0m4d7ed2983a47987a@mail.gmail.com> Message-ID: I'd have to double check all the sql used for registration, but I doubt memcache is expressive enough to act as the registration store. For instance, you can't get a list of registrations from it (sofia status profile internal). memcache is a keystore only. That being said, one could use memcache as a umm.. well cache like it is designed as a front end to the real odbc database. Consult memcache first then hit the db. Doing anything like that would require moving much of mod_memcache up into core, something I promised I would do at one point but never got around to doing -- lack of time and motivation and no strong use case IMO. On Thu, Nov 19, 2009 at 4:23 AM, Lon Baker wrote: > If we could access mod_memcache for registration information that > would be ideal and highly robust, since you can share memcache with > external applications. > > Lon > > On Thu, Nov 19, 2009 at 2:07 AM, Leon de Rooij wrote: >> Well, you can of course easily have a loadbalancer with failover in front of >> your sql servers and have them replicate to each other.?Freeswitch will >> reconnect if a connection goes down. Perhaps failover is also possible >> directly through odbc ? Does anyone know if that's possible ? >> regards, >> Leon >> >> >> On Nov 19, 2009, at 10:33 AM, jay binks wrote: >> >> I believe OBDC is the official way.. >> however id love look at doing this in a higher performance way, without the >> single point of failure.. >> local memcache, in front of OBDC or something ?? >> not 100% sure of it, but just using a single central database is a little >> bit of a concern in a carrier?environment. >> Jay >> >> >> On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij >> wrote: >>> >>> Hi, >>> >>> Not that I know of, but you can use odbc to store registrations and >>> share it that way.. >>> >>> regards, >>> >>> Leon >>> >>> On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: >>> >>> > Hi, >>> > >>> > Is there anyway to store registration info in memcache instead of >>> > sqlite? >>> > >>> > By doing that, it is possible for multiple freeswitch to share the >>> > same user registration info. >>> > >>> > Is there anyway I can intercept the registration success/failure >>> > event and write stuff to memcache myself? >>> > >>> > thanks, >>> > woody >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Sincerely >> >> Jay >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From rupa at rupa.com Thu Nov 19 05:03:51 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 19 Nov 2009 07:03:51 -0600 Subject: [Freeswitch-users] How to implement mod_lcr + mod_limit In-Reply-To: References: Message-ID: Using lcr_auto_route + limit isn't really possible at this point. It is on the list of things to do but is more complex than it seems on it's surface. mod_lcr just constructs dial strings, it doesn't do any call control. It does provide enough information to do what you want via a scripting language like lua. mod_lcr sets channel vars lcr_route_count which tells you how many routes there are. It also sets lcr_route_N (where N is 1 to lcr_route_count) which contains each lcr route. You can then iterate over the routes, set limit try to bridge and loop until success. Arguably this should be done from within FS so that you could just use lcr_auto_route (assuming mod_lcr can pull limit info from the routes db). That is "the plan" but a workable solution hasn't magically appeared yet. On Mon, Nov 16, 2009 at 1:29 AM, Ahmed Munir wrote: > Hi, > > I've worked on setup for carriers routing using mod_lcr + mod_nibble + > mod_xml_curl and mod_xml_cdr. The setup is working fine as I desired. Now I > want to include mod_limit in to my setup. > > As I read the wiki pages of mod_limit I want to know how can I limit the > calls per destination basis while running mod_lcr? Because LCR is routing to > different carriers, how can I call mod_limit in mod_lcr? > > Kindly advise this issue soon. > > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From samuelmukoti at gmail.com Thu Nov 19 05:41:16 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Thu, 19 Nov 2009 15:41:16 +0200 Subject: [Freeswitch-users] XML config file parsing Message-ID: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> Thx Jason for the reply, I realise i was quite unclear in what i'm hoping to achieve. I wanted to make a control panel for our office so that we can provision extensions at the same time as we do users. We have a system much like the "ubuntu ebox" that allows use to manage users for our organization and for virtual domains - it uses postgresql as a backend. I'm not aware of freeswitch's abilities or features when it comes to databases. Can freeswitch lookup SQL tables in realtime? I would love the ability to manage dialplans, voicemail accounts, and extensions/endpoints thru a database much like mysql or postgresql The reason i was discussing XML is for this very same purpose, i though i could write helper scripts that would 'spit' out some XML configuration files thus dynamically updating Freeswitch configuration from a web frontend.. almost similar to what the freepbx.org guys have done. regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/50eaa216/attachment-0002.html From rob4manhere at gmail.com Thu Nov 19 06:03:45 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 19 Nov 2009 08:03:45 -0600 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> References: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> Message-ID: <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> Hi Sam, Take a look at mod_xml_curl. Pretty sure it'll do everything you're looking for. http://wiki.freeswitch.org/wiki/Mod_xml_curl Also, I would browse the modules and look for other nifty functionality that already exists before setting out to write something new. http://wiki.freeswitch.org/wiki/Modules Good luck! Rob On Nov 19, 2009, at 7:41 AM, Samuel Mukoti wrote: > Thx Jason for the reply, > > I realise i was quite unclear in what i'm hoping to achieve. I > wanted to make a control panel for our office so that we can > provision extensions at the same time as we do users. We have a > system much like the "ubuntu ebox" that allows use to manage users > for our organization and for virtual domains - it uses postgresql as > a backend. > > I'm not aware of freeswitch's abilities or features when it comes to > databases. Can freeswitch lookup SQL tables in realtime? > > I would love the ability to manage dialplans, voicemail accounts, > and extensions/endpoints thru a database much like mysql or postgresql > > The reason i was discussing XML is for this very same purpose, i > though i could write helper scripts that would 'spit' out some XML > configuration files thus dynamically updating Freeswitch > configuration from a web frontend.. almost similar to what the > freepbx.org guys have done. > > regards, > > Sam > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/001f3e54/attachment-0002.html From john_platts at hotmail.com Wed Nov 18 18:54:43 2009 From: john_platts at hotmail.com (John Platts) Date: Wed, 18 Nov 2009 20:54:43 -0600 Subject: [Freeswitch-users] Need help configuring our FreeSWITCH instance Message-ID: I have installed FreeSWITCH on our server, and need some help configuring our FreeSWITCH instance. All of the numbers associated with our FreeSWITCH instance are in the format: 1NPANXXXXXX (where NPA is the area code, and NXXXXXX are the last 7 digits of the phone number). I need the following configuration: Calls coming from our IP to IP gateway into our FreeSWITCH instance needs to be routed to the endpoint that is registered with FreeSWITCHCalls coming from any of the registered SIP endpoints need to be sent to the appropriate destination. The appropriate destination for any number that is not registered with FreeSWITCH is our IP to IP gateway.Our IP to IP gateway does not require any SIP registration or authentication.G.729 (but not G.729 Annex B), G.711 mu-law, and G.711 A-law need to be enabledSIP registrar enabled for registering endpoints other than our IP-IP gatewaySIP traffic needs to be accepted to and from both the IP-IP gateway and from the registered SIP endpoints. How do I get the above configured in FreeSWITCH? _________________________________________________________________ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091118/d3082fed/attachment-0002.html From dave at 3c.co.uk Thu Nov 19 07:54:55 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 19 Nov 2009 08:54:55 -0700 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> Message-ID: <1258646095.12820.300.camel@localhost.localdomain> Hi Brian, > It just doesn't belong in user space or kernel space in the machine > for true performance you should do it in hardware... I'm pretty sure > the poor box would die if you tried it on 32 E1's at the same time. Disagree somewhat. The challenge that echo cancellers further from the hardware face is having some idea of the size of the buffers between the canceller and the wire; provided that this is known, or is small in comparison to the canceller's tail length, it can, in principle, go anywhere. All other things being equal, the right place for a software EC is in user space: can be done in a cross-platform way, can use FPU/MMX/SSE without guilt and voodoo, etc. And there is no reason why the same algorithm would perform differently if implemented in "hardware" or on the host CPU. And the OP only needed four E1s.. --Dave > > /b > > On Nov 18, 2009, at 5:39 PM, David Knell wrote: > > > For the sort of box you're talking about (quad core++), this isn't > > lots; > > it's hardly any.. > > > > --Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch-users-list at metik.com Thu Nov 19 08:01:11 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 19 Nov 2009 11:01:11 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <1258511897776-4023012.post@n2.nabble.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com> <1258511897776-4023012.post@n2.nabble.com> Message-ID: <4B056BC7.6030009@metik.com> If you are using Windows XP (or Vista for that matter), you may want to look at tftpd32. Its more compact and uses less memory than Solarwinds yet provides not only a tftp server but a dhcp and syslog server as well. In the past, I've use it to upgrade, install, and troubleshoot a variety of gear (dslams, routers, softswitches, SIP endpoints, etc.) when a dedicated server was not available. -metik Jeff Lenk wrote: > Hi > > I run the SolarWinds TFTP server alongside FS on my small installation - > works nicely! > > Jeff > > > > Dave Stevenson wrote: > >> Hi, >> >> I have just about got FreeSwitch working with a Cisco 7940 Phone. After >> much reading, I worked out that I needed a TFTP server on the network that >> would supply the phone with it's SIP personality and config etc. I have >> been able to get the phone working and realise that the TFTP server needs >> to be available every time the phone loses power etc. At the moment, I >> have the TFTP server running on a temporary machine but it would be neater >> if it ran on the same machine as FreeSwitch. This will be a very small >> FreeSwitch installation, so, ....... >> >> Is there any reason why I should not try to run FreeSwitch and the >> SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't think >> the server should put much load on the machine but wondered if there were >> any other reasons why this is a bad idea ? >> >> In addition, while I have the phone working - I get a status message on >> boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf >> >> Can anyone tell me how to locate the errors in this file please ? (I have >> posted it to the Pastebin) >> >> Regards >> Dave >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > From anthony.minessale at gmail.com Thu Nov 19 08:11:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Nov 2009 10:11:56 -0600 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <4B04EF22.1030404@tigertech.com> References: <4B032142.1000308@tigertech.com> <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> <4B04682A.6000309@tigertech.com> <191c3a030911181528j7a38ce32gb2fc6fdd585932a9@mail.gmail.com> <4B04EF22.1030404@tigertech.com> Message-ID: <191c3a030911190811w267162a2p35cf85bb7e62be40@mail.gmail.com> Like I said, The timer by default is designed to make sure that none of the audio is lost for situations like FAX etc. There are parameters you can configure to disable the timers that I mentioned in the last email which will cause all of the audio to be jammed into your ear like twiddlebugs if you did you sleep test and brought it back. We do not use sleep for the timers we have timer objects into the code derived from a high priority thread sending conditional broadcasts to the timer objects. There is certainly a place where this can begin to break down but it has proven to provide reliable timing to thousands of concurrent channels. The auto-flush can get false positives in jittery situations is not always the best answer. What kind of CPU are you using and what kind of hardware that you suspect you are getting delayed cpu scheduling on a small number of calls? I appreciate your theory and I am willing to investigate a little for you but you must be aware we have put more than a few hours of thought into the architecture here. There may be a bigger problem with the eavesdropping which we can have a look at today because that does not sound right. On Thu, Nov 19, 2009 at 1:09 AM, Robert L Mathews wrote: > Anthony Minessale wrote: > > > I can promise you that much of your problems will be solved with > > latest SVN. > > Okay, thanks! > > And in fact, I tried today's SVN, and it has fixed the problem with the > conference, even without setting "rtp-autoflush". Conferences now > discard packets and "catch up" when they gets behind, even with only the > default "rtp-autoflush-during-bridge" set. > > The echo test still suffers from the same problem unless "rtp-autoflush" > is used, which I assume is simply because it's not considered a bridged > call. > > Eavesdropping on an existing bridged call, then pressing "3" to turn it > into a conference call, also requires "rtp-autoflush" to avoid > persistent lag on the third leg. > > > > Did you answer the question about what phones? I'm going to guess Cisco > > based on the symptoms. > > It happens with all phones, as far as I can tell. I've tried at least > Grandstream GXP2000, Grandstream BT102, SJPhone, Twinkle, and Express > Talk (none of them Cisco). I'm fairly positive the problem is unrelated > to phones; it's caused by delays in CPU scheduling of the server process. > > > > non bridge calls use a timer to make sure the audio is coming in at a > > steady rate to ensure bursting RTP > > is played at the correct rate. Stopping it and restarting 2 seconds > > later will cause a delay by design because you have suspended the > > process but not the UDP stack. > > Ummm.... well, a copy of FreeSWITCH running on any non-realtime > operating system will occasionally not get scheduled for all the CPU > time it wants. For example, it wouldn't be unusual for a thread to ask > to sleep for 20 milliseconds but actually not wake up for 21, 25, or > even 40 milliseconds because the server is busy with other things. > > Each time that happens, it's a smaller version of my artificial suspend > test: the operating system has, of course "suspended the process but not > the UDP stack". It doesn't necessarily mean there's bursty network > traffic or phone timing issues. > > Should FreeSWITCH really lag by that much for the rest of the call? 20 > milliseconds here, 20 milliseconds there, and pretty soon you're talking > about real seconds. > > I'm assuming the reason for making it catch up on bridged calls, but not > non-bridged calls, is that people talking to each other can't tolerate > high latency, but people listening to an IVR or something can. But is > that still true if it gets seconds behind? And should the third leg of > an eavesdrop-converted-to-three-way-call be considered non-bridged for > this purpose? > > Anyway, given that current svn trunk fixes the problem by default in > conferences and any other bridged call, I'm satisfied. And if anyone > complains about this problem for non-bridged calls, I suppose they can > enable "rtp-autoflush" to get the same "catch-up" behavior there. > > I took a shot at documenting these parameters in the wiki on: > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#rtp-autoflush-during-bridge > > Thanks for the help! > > -- > Robert L Mathews, Tiger Technologies > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/2d51c0cc/attachment-0002.html From steveu at coppice.org Thu Nov 19 08:15:15 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 20 Nov 2009 00:15:15 +0800 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258646095.12820.300.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> <1258646095.12820.300.camel@localhost.localdomain> Message-ID: <4B056F13.6050106@coppice.org> On 11/19/2009 11:54 PM, David Knell wrote: > Hi Brian, > > >> It just doesn't belong in user space or kernel space in the machine >> for true performance you should do it in hardware... I'm pretty sure >> the poor box would die if you tried it on 32 E1's at the same time. >> > Disagree somewhat. The challenge that echo cancellers further from the > hardware face is having some idea of the size of the buffers between the > canceller and the wire; provided that this is known, or is small in > comparison to the canceller's tail length, it can, in principle, go > anywhere. All other things being equal, the right place for a software > EC is in user space: can be done in a cross-platform way, can use > FPU/MMX/SSE without guilt and voodoo, etc. And there is no reason why > the same algorithm would perform differently if implemented in > "hardware" or on the host CPU. > > And the OP only needed four E1s.. > The audio path between kernel and user space is not stable with any current PC based telephony system. At some point in the day the odd chunk of data is lost here and there, whether you use asterisk, callweaver, yate or FS, with dahdi or sangoma. This is the key problem for user space echo cancellation. When the path hiccups, the EC goes crazy, and howls. So far kernel space EC has been the only way to keep the path length rock solid. There is an Intel development platform which tries to do EC with OSLEC in user space. That's the only delivered system I know that tries to do this. Its very quirky. Steve From stevendt at primrosebank.net Thu Nov 19 08:46:50 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 16:46:50 -0000 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> Message-ID: Metik, thanks a lot for the tip, I will certainly look at it, particularly if it does DHCP too. At the moment, I use my ADSL Router to provide DHCP to the network but I've just discovered that you can't configure options in its DHCP server to point to the TFTP server for the phone. At the moment, I have to have the phone set to a static IP address to be able to configure the TFTP server address which is not as flexible as using DHCP. I had thought about changing over to use Windows Server DHCP services but it sounds like ttpd32 would do the trick. I just need to decide whether I want all of my machines to rely on getting their IP address from another PC - it feels like having DHCP in the router is more robust. Regards Dave ----- Original Message ----- From: "Metik" To: Sent: Thursday, November 19, 2009 4:01 PM Subject: Re: [Freeswitch-users] TFTP Server & Cisco 7540 > If you are using Windows XP (or Vista for that matter), you may want to > look at tftpd32. Its more compact and uses less memory than Solarwinds > yet provides not only a tftp server but a dhcp and syslog server as well. > > In the past, I've use it to upgrade, install, and troubleshoot a variety > of gear (dslams, routers, softswitches, SIP endpoints, etc.) when a > dedicated server was not available. > > -metik > > > Jeff Lenk wrote: >> Hi >> >> I run the SolarWinds TFTP server alongside FS on my small installation - >> works nicely! >> >> Jeff >> >> >> >> Dave Stevenson wrote: >> >>> Hi, >>> >>> I have just about got FreeSwitch working with a Cisco 7940 Phone. After >>> much reading, I worked out that I needed a TFTP server on the network >>> that >>> would supply the phone with it's SIP personality and config etc. I have >>> been able to get the phone working and realise that the TFTP server >>> needs >>> to be available every time the phone loses power etc. At the moment, I >>> have the TFTP server running on a temporary machine but it would be >>> neater >>> if it ran on the same machine as FreeSwitch. This will be a very small >>> FreeSwitch installation, so, ....... >>> >>> Is there any reason why I should not try to run FreeSwitch and the >>> SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't >>> think >>> the server should put much load on the machine but wondered if there >>> were >>> any other reasons why this is a bad idea ? >>> >>> In addition, while I have the phone working - I get a status message on >>> boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf >>> >>> Can anyone tell me how to locate the errors in this file please ? (I >>> have >>> posted it to the Pastebin) >>> >>> Regards >>> Dave >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Thu Nov 19 08:55:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 10:55:28 -0600 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> Message-ID: <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> Some Cisco phones need DHCP option 150. /b On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: > Metik, > > thanks a lot for the tip, I will certainly look at it, particularly > if it > does DHCP too. > > At the moment, I use my ADSL Router to provide DHCP to the network > but I've > just discovered that you can't configure options in its DHCP server > to point > to the TFTP server for the phone. At the moment, I have to have the > phone > set to a static IP address to be able to configure the TFTP server > address > which is not as flexible as using DHCP. I had thought about changing > over to > use Windows Server DHCP services but it sounds like ttpd32 would do > the > trick. > > I just need to decide whether I want all of my machines to rely on > getting > their IP address from another PC - it feels like having DHCP in the > router > is more robust. > > Regards > Dave From kjv at ken-ton.com Thu Nov 19 10:11:00 2009 From: kjv at ken-ton.com (Karl J. Vesterling) Date: Thu, 19 Nov 2009 13:11:00 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> Message-ID: <868A4E38-D947-4291-BBD7-4F4C9E5B239E@ken-ton.com> Yeah, roger that... Here is an excerpt from the page I did on the Cisco 7960G HowTo: http://wiki.freeswitch.org/wiki/Freeswitch_Cisco_7960G_Howto It's for Linux, but you'll get some good pointers on the TFTP option you're looking for. I haven't provisioned any 7540's... Good luck! Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Nov 19, 2009, at 11:55 AM, Brian West wrote: > Some Cisco phones need DHCP option 150. > > /b > > On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: > >> Metik, >> >> thanks a lot for the tip, I will certainly look at it, particularly >> if it >> does DHCP too. >> >> At the moment, I use my ADSL Router to provide DHCP to the network >> but I've >> just discovered that you can't configure options in its DHCP server >> to point >> to the TFTP server for the phone. At the moment, I have to have the >> phone >> set to a static IP address to be able to configure the TFTP server >> address >> which is not as flexible as using DHCP. I had thought about changing >> over to >> use Windows Server DHCP services but it sounds like ttpd32 would do >> the >> trick. >> >> I just need to decide whether I want all of my machines to rely on >> getting >> their IP address from another PC - it feels like having DHCP in the >> router >> is more robust. >> >> Regards >> Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Nov 19 10:17:48 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 12:17:48 -0600 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <868A4E38-D947-4291-BBD7-4F4C9E5B239E@ken-ton.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> <868A4E38-D947-4291-BBD7-4F4C9E5B239E@ken-ton.com> Message-ID: I don't think a 7540 exists. /b On Nov 19, 2009, at 12:11 PM, Karl J. Vesterling wrote: > I haven't provisioned any 7540's... Good luck! From stevendt at primrosebank.net Thu Nov 19 10:25:42 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 18:25:42 -0000 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com><4B056BC7.6030009@metik.com><921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> <868A4E38-D947-4291-BBD7-4F4C9E5B239E@ken-ton.com> Message-ID: Thanks Guys, I had not realised until the last couple of days that DHCP did more than just providing the IP address to the client. I have been happily just doing that for a few years now without anything other than my Router providing the DHCP function. It's only now that I have taken the plunge into IP telephony that I realise that it can do more and for Cisco phones, should provide the address of the TFTP server. My work-around at the moment is to used fixed IP addresses in the phone for it's own IP address and the TFTP server - not as neat as I would like, but it works. I will look at a better long term solution with a different DHCP server (as already mentioned earlier in this thread). Looking on the bright side, I have got the phone provisioned - though I'm still working out what all the options are, but it is working. As Brian has spotted - my reference to a 7540 was an error - I got in right in the body of the original post, but not when I edited the subject line - oooops - sorry. The phone is a 7940 ! regards Dave ----- Original Message ----- From: "Karl J. Vesterling" To: Sent: Thursday, November 19, 2009 6:11 PM Subject: Re: [Freeswitch-users] TFTP Server & Cisco 7540 > Yeah, roger that... > Here is an excerpt from the page I did on the Cisco 7960G HowTo: > > http://wiki.freeswitch.org/wiki/Freeswitch_Cisco_7960G_Howto > > It's for Linux, but you'll get some good pointers on the TFTP option > you're looking for. > I haven't provisioned any 7540's... Good luck! > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 > > On Nov 19, 2009, at 11:55 AM, Brian West wrote: > >> Some Cisco phones need DHCP option 150. >> >> /b >> >> On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: >> >>> Metik, >>> >>> thanks a lot for the tip, I will certainly look at it, particularly >>> if it >>> does DHCP too. >>> >>> At the moment, I use my ADSL Router to provide DHCP to the network >>> but I've >>> just discovered that you can't configure options in its DHCP server >>> to point >>> to the TFTP server for the phone. At the moment, I have to have the >>> phone >>> set to a static IP address to be able to configure the TFTP server >>> address >>> which is not as flexible as using DHCP. I had thought about changing >>> over to >>> use Windows Server DHCP services but it sounds like ttpd32 would do >>> the >>> trick. >>> >>> I just need to decide whether I want all of my machines to rely on >>> getting >>> their IP address from another PC - it feels like having DHCP in the >>> router >>> is more robust. >>> >>> Regards >>> Dave >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shiyanov at gmail.com Thu Nov 19 11:25:57 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 19 Nov 2009 22:25:57 +0300 Subject: [Freeswitch-users] mod dptools record problem - hangup channel with invalid file path In-Reply-To: <67d615ac0911181226y22b4fec6ndb8e622a24db101c@mail.gmail.com> References: <67d615ac0911181138m30f3064ci1a2dad6732354e35@mail.gmail.com> <994A83CB-7069-4808-9055-30B8BD3CEA75@jerris.com> <67d615ac0911181226y22b4fec6ndb8e622a24db101c@mail.gmail.com> Message-ID: I had almost the same problem- it was needed to record everything, even if the record path doesn't exist - it was requested to create the needed path. For this purpose I've used event_socket command "api system ...", precisely, api system mkdir -p path And after this command I've started recording. So, you may the same approach. On Wed, Nov 18, 2009 at 11:26 PM, William Kendi ... < william.nishio at voicetechnology.com.br> wrote: > Actually, I am integrating FreeSWITCH with a weird IVR Framework, and the > current behaviour of the "mod dptools record" application breaks some rules > of the weird IVR Framework that must be integrated with FreeSWITCH. > In order to integrate FreeSWITCH with the weird IVR Framework, the "mod > dptools record" application mustn't terminate the call when invalid file > paths are passed, and a notification of the invalid file path through the > event system of FreeSWITCH should be enough for me, like the behaviour of > the "mod dptools playback" application when invalid file paths are passed. > > Thanks in advance. > > ** > 2009/11/18 Michael Jerris > > Okay, I'll ask the obvious question. Why are you passing record invalid >> file paths and why should it not fail if you do? >> >> Mike >> >> On Nov 18, 2009, at 2:38 PM, William Kendi ... wrote: >> >> > While I was testing the "mod dptools record" application using invalid >> file paths, i noted that the "mod dptools record" application terminated the >> call. >> > I am currently looking for a way to change this behaviour. >> > Any suggestions? Can this be done? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/7e83bf30/attachment-0002.html From info at daccii.it Thu Nov 19 11:29:43 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Thu, 19 Nov 2009 20:29:43 +0100 Subject: [Freeswitch-users] Call doesn't work while registration work for a VOIP provider Message-ID: <4B059CA7.3040201@daccii.it> Hi, i'm trying to configure freeswitch with a VOIP provider, exsorsa, that uses OpenSER. Exsorsa use as own gateway, another provider, Eutelia, that it uses Cisco (or, at least, this appears in headers). Short story: ------------ If i try to setup my Eutelia account all works perfectly while if i try to setup Exsorsa account registration works fine while calling not: when fs send the ACK, as answer to a OK (sip code 200), that is sended from exsorsa as answer to an INVITE, exsorsa send back a BYE. Long story: ----------- I put call log on pastebin with debug and sip_trace enabled for external sip_profile and with log level on debug on fs console. Registration log, here all is ok (or at least it seems to be ok) http://pastebin.freeswitch.org/11176 Annoyng message that comes up every 30 seconds http://pastebin.freeswitch.org/11177 Call log http://pastebin.freeswitch.org/11178 As you can see from call log all works fine until fs send back the acknowledgment message (line 451 on last log). Can this depend on the annoyng message that comes up every 30 seconds? Here my external sip profile config http://pastebin.freeswitch.org/11180 while here exsorsa gateway config http://pastebin.freeswitch.org/11181 Any helps is really appreciated! I'm fought with it all the day!!! Best Regards, Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/712d6ef2/attachment-0002.vcf From brian at freeswitch.org Thu Nov 19 11:38:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 13:38:53 -0600 Subject: [Freeswitch-users] Call doesn't work while registration work for a VOIP provider In-Reply-To: <4B059CA7.3040201@daccii.it> References: <4B059CA7.3040201@daccii.it> Message-ID: I'm going to guess gw+exsorsa is what they don't like. try extensions- in-contact=true on the gateway config. /b On Nov 19, 2009, at 1:29 PM, Albano Daniele Salvatore - Lavoro wrote: > Hi, > > i'm trying to configure freeswitch with a VOIP provider, exsorsa, > that uses OpenSER. Exsorsa use as own gateway, another provider, > Eutelia, that it uses Cisco (or, at least, this appears in headers). > > Short story: > ------------ > If i try to setup my Eutelia account all works perfectly while if i > try to setup Exsorsa account registration works fine while calling > not: when fs send the ACK, as answer to a OK (sip code 200), that is > sended from exsorsa as answer to an INVITE, exsorsa send back a BYE. > > > Long story: > ----------- > I put call log on pastebin with debug and sip_trace enabled for > external sip_profile and with log level on debug on fs console. > > Registration log, here all is ok (or at least it seems to be ok) > http://pastebin.freeswitch.org/11176 > > Annoyng message that comes up every 30 seconds > http://pastebin.freeswitch.org/11177 > > Call log > http://pastebin.freeswitch.org/11178 > > As you can see from call log all works fine until fs send back the > acknowledgment message (line 451 on last log). > > Can this depend on the annoyng message that comes up every 30 seconds? > > Here my external sip profile config > http://pastebin.freeswitch.org/11180 > > while here exsorsa gateway config > http://pastebin.freeswitch.org/11181 > > > Any helps is really appreciated! I'm fought with it all the day!!! > > Best Regards, > Daniele > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From shiyanov at gmail.com Thu Nov 19 11:46:14 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 19 Nov 2009 22:46:14 +0300 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app Message-ID: Hi there! I've got annoying FS behavior: There are 2 channels executing the same Java application (application itself is an IVR). If I try to bridge them with uuid_bridged then both channels are killed. Here is a log from FS console: uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 (sofia/internal/ 1005 at 192.168.147.130) State Change CS_EXECUTE -> CS_HIBERNATE 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1005 at 192.168.147.130 [BREAK] 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 (sofia/internal/ 1001 at master.agent.starpoundtech.net) State Change CS_EXECUTE -> CS_HIBERNATE 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at master.agent.starpoundtec 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1005 at 192.168.147.130 [BREAK] 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] (FS version is 1.0.4) Any thoughts? Artem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/cec920e7/attachment-0002.html From jerry.richards at teotech.com Thu Nov 19 12:00:14 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 19 Nov 2009 12:00:14 -0800 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP Message-ID: <7C3068B7C76746E4A9ACC216574035B9@greyhawk.tonecommander.com> Hello, I just pasted a log in the Pastebin with Freeswitch logging enabled. Does anyone know a way to prevent FS from connecting the call prior to the callee answering? Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Thursday, November 05, 2009 3:50 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Want 183 w/SDP, but Get 200 w/SDP I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry From anthony.minessale at gmail.com Thu Nov 19 12:18:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Nov 2009 14:18:05 -0600 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP In-Reply-To: <7C3068B7C76746E4A9ACC216574035B9@greyhawk.tonecommander.com> References: <7C3068B7C76746E4A9ACC216574035B9@greyhawk.tonecommander.com> Message-ID: <191c3a030911191218m6fb6992eg5b4eaf338397ed0b@mail.gmail.com> set enable-3pcc to "proxy" instead of "true" On Thu, Nov 19, 2009 at 2:00 PM, Jerry Richards wrote: > > Hello, > > I just pasted a log in the Pastebin with Freeswitch logging enabled. Does > anyone know a way to prevent FS from connecting the call prior to the > callee > answering? > > Best Regards, > Jerry > > > -----Original Message----- > From: Jerry Richards [mailto:jerry.richards at teotech.com] > Sent: Thursday, November 05, 2009 3:50 PM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: Want 183 w/SDP, but Get 200 w/SDP > > > I am trying to make a call through a Gateway that sends the INVITE with no > SDP and ONLY wants the 200 OK w/SDP when the callee answers. > > For some reason, Freeswitch answers the call with 200 OK w/SDP even before > the callee answers the phone. Is this to provide ringback? Can I disable > that action? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/3f24680c/attachment-0002.html From msc at freeswitch.org Thu Nov 19 12:25:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Nov 2009 12:25:36 -0800 Subject: [Freeswitch-users] Extension Configuration - XML File Entries for Group configuration In-Reply-To: <73DD76AE07884A1D9535EF27C5841DAD@bp1.ad.bp.com> References: <73DD76AE07884A1D9535EF27C5841DAD@bp1.ad.bp.com> Message-ID: <87f2f3b90911191225l243a5cdflf8c61b8ebc76dfcb@mail.gmail.com> On Thu, Nov 19, 2009 at 3:33 AM, Dave Stevenson wrote: > Hi, > > Can someone please help me understand a little more about Group > configuration ? > > I believe that Group Membership is configured in the > \conf\directory\default.xml file > > I've done this and the caller groups seem to work fine. > > However, each extension in the \conf\directory\default directory, e.g., > 111.xml also has an entry for "callgroup" > > Can someone explain what the difference in these two options is please ? > > The groups defined in conf/directory/default.xml correspond to the "group" channel or group_call API as can be found in conf/dialplan/default.xml, extensions 2000, 2001, and 2002. Go to the fs_cli and type this: group_call sales at 1.1.1.1 (where 1.1.1.1 is your FS IP addr) You'll see that it returns a nicely formatted multiple dialstring for dialing everyone in the group. These have nothing to do with the "callgroup" variable that is defined on each user in the default directory. That is just a variable - it isn't required and doesn't have to be used, but it's available if you want it for some reason. (For example, it will show up in XML CDRs for auth'd calls from the user.) Bottom line: if you're trying to dial multiple users (i.e. "group call") then just use the group definitions in the directory and use either the group_call API (like in ext 2000) or use the "group" channel (like in ext 2001 and 2002). -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/f67b0def/attachment-0002.html From msc at freeswitch.org Thu Nov 19 12:28:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Nov 2009 12:28:39 -0800 Subject: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP In-Reply-To: <191c3a030911191218m6fb6992eg5b4eaf338397ed0b@mail.gmail.com> References: <7C3068B7C76746E4A9ACC216574035B9@greyhawk.tonecommander.com> <191c3a030911191218m6fb6992eg5b4eaf338397ed0b@mail.gmail.com> Message-ID: <87f2f3b90911191228i52c89598v78845d80abced257@mail.gmail.com> On Thu, Nov 19, 2009 at 12:18 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > set enable-3pcc to "proxy" instead of "true" > > FYI, the wiki entry is here: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#enable-3pcc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/cb8deb43/attachment-0002.html From msc at freeswitch.org Thu Nov 19 12:30:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Nov 2009 12:30:29 -0800 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: References: Message-ID: <87f2f3b90911191230u512878e3x122efa0357cf83c5@mail.gmail.com> On Thu, Nov 19, 2009 at 11:46 AM, Artem Shiyanov wrote: > Hi there! > > I've got annoying FS behavior: > There are 2 channels executing the same Java application (application > itself is an IVR). If I try to bridge them with uuid_bridged then both > channels are killed. Here is a log from FS console: > uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 > 7d6c08fc-62bf-4a6c-a9ae-763d607e43de > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 > (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> > CS_HIBERNATE > 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1005 at 192.168.147.130 [BREAK] > 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 > (sofia/internal/1001 at master.agent.starpoundtech.net) State Change > CS_EXECUTE -> CS_HIBERNATE > 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called > API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 > 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: > +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de > > freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] > switch_core_session.c:933 Send signal > sofia/internal/1001 at master.agent.starpoundtec > 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal > sofia/internal/1005 at 192.168.147.130 [BREAK] > > 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal > sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] > > (FS version is 1.0.4) > > Any thoughts? > > First, update to latest trunk - there are many behaviors that have been tweaked and repaired since early August when 1.0.4 came out. Try it on latest trunk and see if the behavior persists, is different, or is gone. Please report back and let us know how it all goes. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/f40d08e8/attachment-0002.html From mrene_lists at avgs.ca Thu Nov 19 12:33:17 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 19 Nov 2009 12:33:17 -0800 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: References: Message-ID: I don't see any hangups here, are you talking about the BREAK signals? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 19-Nov-09, at 11:46 AM, Artem Shiyanov wrote: > Hi there! > > I've got annoying FS behavior: > There are 2 channels executing the same Java application > (application itself is an IVR). If I try to bridge them with > uuid_bridged then both channels are killed. Here is a log from FS > console: > uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c- > a9ae-763d607e43de > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 (sofia/internal/1005 at 192.168.147.130 > ) State Change CS_EXECUTE -> CS_HIBERNATE > 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook > called > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done > playing file > 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done > playing file > 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/1005 at 192.168.147.130 [BREAK] > 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 (sofia/internal/1001 at master.agent.starpoundtech.net > ) State Change CS_EXECUTE -> CS_HIBERNATE > 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook > called > API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 > 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: > +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de > > freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] > switch_core_session.c:933 Send signal sofia/internal/1001 at master.agent.starpoundtec > 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send > signal sofia/internal/1005 at 192.168.147.130 [BREAK] > > 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking > stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send > signal sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] > > (FS version is 1.0.4) > > Any thoughts? > > > Artem > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/98559953/attachment-0002.html From stevendt at primrosebank.net Thu Nov 19 12:36:30 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 20:36:30 -0000 Subject: [Freeswitch-users] Extension Configuration - XML File Entriesfor Group configuration References: <73DD76AE07884A1D9535EF27C5841DAD@bp1.ad.bp.com> <87f2f3b90911191225l243a5cdflf8c61b8ebc76dfcb@mail.gmail.com> Message-ID: <843A417506F445D6ABCECFA05037BA2D@bp1.ad.bp.com> Thanks Michael, I think I've got it ! regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Thursday, November 19, 2009 8:25 PM Subject: Re: [Freeswitch-users] Extension Configuration - XML File Entriesfor Group configuration On Thu, Nov 19, 2009 at 3:33 AM, Dave Stevenson wrote: Hi, Can someone please help me understand a little more about Group configuration ? I believe that Group Membership is configured in the \conf\directory\default.xml file I've done this and the caller groups seem to work fine. However, each extension in the \conf\directory\default directory, e.g., 111.xml also has an entry for "callgroup" Can someone explain what the difference in these two options is please ? The groups defined in conf/directory/default.xml correspond to the "group" channel or group_call API as can be found in conf/dialplan/default.xml, extensions 2000, 2001, and 2002. Go to the fs_cli and type this: group_call sales at 1.1.1.1 (where 1.1.1.1 is your FS IP addr) You'll see that it returns a nicely formatted multiple dialstring for dialing everyone in the group. These have nothing to do with the "callgroup" variable that is defined on each user in the default directory. That is just a variable - it isn't required and doesn't have to be used, but it's available if you want it for some reason. (For example, it will show up in XML CDRs for auth'd calls from the user.) Bottom line: if you're trying to dial multiple users (i.e. "group call") then just use the group definitions in the directory and use either the group_call API (like in ext 2000) or use the "group" channel (like in ext 2001 and 2002). -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/2acca687/attachment-0002.html From stevendt at primrosebank.net Thu Nov 19 12:43:51 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 20:43:51 -0000 Subject: [Freeswitch-users] Another Group Question - on VoiceMail Message-ID: Hi again ! I have FreeSwitch configured such that if someone dials in from the PSTN line, a group of phones ring. If nobody answers, the group extension number (100) picks up the call and voice mail kicks in. So far, so good, each of the individual phones logs a missed call and anyone in the group can call into voice mail and go to the extension 100 mailbox to check if there are any messages but the individual phones are not notified that a Voice message is waiting. Is there any way that each extension in the group can be notified that a group Voice Mail is waiting to be picked up so that each phone shows the message waiting indication ? Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/44c0b35b/attachment-0002.html From brian at freeswitch.org Thu Nov 19 12:49:59 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 14:49:59 -0600 Subject: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv Message-ID: We have removed the two modules using the reference code from BroadVoice and added a lib with a new interface from Steve Underwood and mod_bv.c using this lib... We know their is ONE last bug to be fixed in the lib before its working so please do not open any jira's if you try to run it because it will crash right now. Thanks for your understanding and once this is fixed it'll work with aastra and x-lite on both 32bit and 64bit systems without any issues. Thanks, Brian West From msc at freeswitch.org Thu Nov 19 12:57:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Nov 2009 12:57:10 -0800 Subject: [Freeswitch-users] Need help configuring our FreeSWITCH instance In-Reply-To: References: Message-ID: <87f2f3b90911191257n3fbea0eagce482a363c3abb99@mail.gmail.com> On Wed, Nov 18, 2009 at 6:54 PM, John Platts wrote: > I have installed FreeSWITCH on our server, and need some help configuring > our FreeSWITCH instance. All of the numbers associated with our FreeSWITCH > instance are in the format: 1NPANXXXXXX (where NPA is the area code, and > NXXXXXX are the last 7 digits of the phone number). > > I need the following configuration: > > - Calls coming from our IP to IP gateway into our FreeSWITCH instance > needs to be routed to the endpoint that is registered with FreeSWITCH > - Calls coming from any of the registered SIP endpoints need to be sent > to the appropriate destination. The appropriate destination for any number > that is not registered with FreeSWITCH is our IP to IP gateway. > - Our IP to IP gateway does not require any SIP registration or > authentication. > - G.729 (but not G.729 Annex B), G.711 mu-law, and G.711 A-law need to > be enabled > - SIP registrar enabled for registering endpoints other than our IP-IP > gateway > - SIP traffic needs to be accepted to and from both the IP-IP gateway > and from the registered SIP endpoints. > > > How do I get the above configured in FreeSWITCH? > > I'd say you have two choices: roll up your sleeves and start learning or email consulting at freeswitch.org and get some paid help. All of the questions you asked are answered in the wiki (and in some cases, mailing list history) but the answers require some foundational knowledge for them to make sense. If you are not a VoIP user then I'd recommend going the paid route and getting a professional to assist you - it will be the fastest way to get up and running. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/a2a94f08/attachment-0002.html From msc at freeswitch.org Thu Nov 19 13:01:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Nov 2009 13:01:01 -0800 Subject: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv In-Reply-To: References: Message-ID: <87f2f3b90911191301q1083e23fs63fc11722bb60aa5@mail.gmail.com> On Thu, Nov 19, 2009 at 12:49 PM, Brian West wrote: > We have removed the two modules using the reference code from > BroadVoice and added a lib with a new interface from Steve Underwood > and mod_bv.c using this lib... We know their is ONE last bug to be > fixed in the lib before its working so please do not open any jira's > if you try to run it because it will crash right now. > > Thanks for your understanding and once this is fixed it'll work with > aastra and x-lite on both 32bit and 64bit systems without any issues. > > Thanks, Brian West > > Thanks to Brian, Tony, Mike, and Steve U. for all their hard work on this. Not only did they get this implemented quickly, they found a few bugs and reported back to the Broadcom guys. :) Excellent work all the way around. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/e4c46407/attachment-0002.html From lists at tigertech.com Thu Nov 19 13:08:08 2009 From: lists at tigertech.com (Robert L Mathews) Date: Thu, 19 Nov 2009 13:08:08 -0800 Subject: [Freeswitch-users] Call latency in conferences and echo test increases over time In-Reply-To: <191c3a030911190811w267162a2p35cf85bb7e62be40@mail.gmail.com> References: <4B032142.1000308@tigertech.com> <191c3a030911181146i17b75f76ia38be218acfdb95b@mail.gmail.com> <4B04682A.6000309@tigertech.com> <191c3a030911181528j7a38ce32gb2fc6fdd585932a9@mail.gmail.com> <4B04EF22.1030404@tigertech.com> <191c3a030911190811w267162a2p35cf85bb7e62be40@mail.gmail.com> Message-ID: <4B05B3B8.6000708@tigertech.com> Anthony Minessale wrote: > Like I said, > The timer by default is designed to make sure that none of the audio is > lost for situations like FAX etc. Right, that makes sense. I've updated the wiki entries I made to warn about this. > We do not use sleep for the timers we have timer objects into the code > derived from a high priority thread sending conditional broadcasts to > the timer objects. Sorry for not being clear. When I said it "sleeps", I just meant "the operating system isn't scheduling any FreeSWITCH threads to run for some period of time, for whatever reason". > What kind of CPU are you using and what kind of hardware that you > suspect you are getting delayed cpu scheduling on a small number of calls? Well, I'm using 2.4 GHz dual Xeons, but couldn't this situation happen on any hardware, if it also has non-FreeSWITCH processes consuming lots of CPU time? That's because the timer needs to make sure that rtp_common_read() is called at least once every 20 ms. If it can't be called that often, for any reason, then FreeSWITCH will fall behind the RTP stream. At that point, audio latency will certainly increase unless some of the packets are discarded. I could duplicate the latency on 1.0.4 by running many other non-FreeSWITCH processes on the same server, so that all the freeswitch threads get starved for CPU time. FreeSWITCH then can't read the RTP packets as fast as they come in, and since the 1.0.4 code didn't flush those extra packets in conferences, that caused noticeable latency. Imposing heavy server load is obviously a silly thing to do, but something similar could happen on any server that fires up lots of non-FreeSWITCH, CPU-hungry processes. (In my case it was virus scanners.) Not using a dedicated server is also silly if people care about call quality, but I was just initially using it for conferences, and I didn't care if some packets were dropped. But conference packet dropping didn't happen on 1.0.4. Instead, a noticeable lag developed, which I did care about. Since 1.0.5 *does* work the way I expect in conferences and other bridged calls (discarding packets), I'm *definitely* not complaining -- please consider this a resolved issue! I agree that it makes sense to preserve all packets for some RTP streams such as faxes and DTMF recognition, and basing that decision on whether the call is bridged makes as much sense as anything else I can think of (although perhaps that flag isn't getting set properly for the third leg of eavesdrop-converted-to-three-way calls). I've been impressed by the extremely high performance of FreeSWITCH. The conference latency I was hearing in 1.0.4 was caused by the fact that I'm stressing the server with separate, unrelated processes, which is a foolish thing to do if you care about audio quality. I was just hoping that FreeSWITCH could more gracefully deal with such foolishness in cases where people *don't* care about audio quality... and 1.0.5 does. That's perfect. Thanks again! -- Robert L Mathews, Tiger Technologies From dave at 3c.co.uk Thu Nov 19 13:15:21 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 19 Nov 2009 14:15:21 -0700 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <4B056F13.6050106@coppice.org> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> <1258646095.12820.300.camel@localhost.localdomain> <4B056F13.6050106@coppice.org> Message-ID: <1258665321.12582.6.camel@localhost.localdomain> On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote: > The audio path between kernel and user space is not stable with any > current PC based telephony system. At some point in the day the odd > chunk of data is lost here and there, whether you use asterisk, > callweaver, yate or FS, with dahdi or sangoma. This is the key problem > for user space echo cancellation. When the path hiccups, the EC goes > crazy, and howls. So far kernel space EC has been the only way to keep > the path length rock solid. Why do you think this is? Getting data from kernel space to user space isn't something which it's difficult to do reliably: the disk system manages it. Even if data is being lost, buffer overruns can be dealt with by using bigger buffers, or timestamping blocks of data on their way in so that missing blocks can be detected. --Dave From lon at kickasspixels.com Thu Nov 19 13:16:53 2009 From: lon at kickasspixels.com (Lon Baker) Date: Thu, 19 Nov 2009 13:16:53 -0800 Subject: [Freeswitch-users] Radius for registration Message-ID: <5d3e0dc60911191316g57a875fbqc15da46fc9847913@mail.gmail.com> Hi everyone, I want to verify what the wiki says, you can use a radius server as the data source for your registrations? Lon From JCasale at activenetwerx.com Thu Nov 19 14:26:02 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 19 Nov 2009 22:26:02 +0000 Subject: [Freeswitch-users] Another Group Question - on VoiceMail In-Reply-To: References: Message-ID: >Is there any way that each extension in the group can be notified that a >group Voice Mail is waiting to be picked up so that each phone shows the >message waiting indication ? Wouldn't this be simply accomplished by setting the vicemail as box 100 for each of the users (such as ext 101....1xx)? From stevendt at primrosebank.net Thu Nov 19 14:49:47 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 19 Nov 2009 22:49:47 -0000 Subject: [Freeswitch-users] Another Group Question - on VoiceMail References: Message-ID: Thanks Joseph, that would be one way, but it would mean that everyone had a common mailbox for all calls, I just wanted to do it for calls coming in on the PSTN line. Maybe that's not possible though ? regards Dave ----- Original Message ----- From: "Joseph L. Casale" To: Sent: Thursday, November 19, 2009 10:26 PM Subject: Re: [Freeswitch-users] Another Group Question - on VoiceMail > >Is there any way that each extension in the group can be notified that a >>group Voice Mail is waiting to be picked up so that each phone shows the >>message waiting indication ? > > Wouldn't this be simply accomplished by setting the vicemail as box 100 > for > each of the users (such as ext 101....1xx)? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch-users-list at metik.com Thu Nov 19 15:23:40 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 19 Nov 2009 18:23:40 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> Message-ID: <4B05D37C.4000607@metik.com> He should be able to just use "Additional Option" to add option 150 (and the associated IP address to which the TFTP server is bound). Brian West wrote: > Some Cisco phones need DHCP option 150. > > /b > > On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: > > >> Metik, >> >> thanks a lot for the tip, I will certainly look at it, particularly >> if it >> does DHCP too. >> >> At the moment, I use my ADSL Router to provide DHCP to the network >> but I've >> just discovered that you can't configure options in its DHCP server >> to point >> to the TFTP server for the phone. At the moment, I have to have the >> phone >> set to a static IP address to be able to configure the TFTP server >> address >> which is not as flexible as using DHCP. I had thought about changing >> over to >> use Windows Server DHCP services but it sounds like ttpd32 would do >> the >> trick. >> >> I just need to decide whether I want all of my machines to rely on >> getting >> their IP address from another PC - it feels like having DHCP in the >> router >> is more robust. >> >> Regards >> Dave >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Mailings at kh-dev.de Thu Nov 19 15:54:04 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Fri, 20 Nov 2009 00:54:04 +0100 Subject: [Freeswitch-users] Media got stuck after attended transfer... In-Reply-To: <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> References: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com> <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com> <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> Message-ID: Hi, one of my customers is willing to contribute for t38 integration. The basic idea is to connect HylaFAX to FS: t38modem <-> FreeSWITCH <-> Media Gateway with t38 support All this without media proxy. Another idea might be to implement t38 origination/termination with a class 1 modem input/output for use with HylaFAX. Do you know how much money we need to collect for t38 support? How much time is needed for implementing this? Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, October 16, 2009 2:10 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Media got stuck after attended transfer... On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga > wrote: hi, any clue when can t38 be added? "Eventually." :) Of course, if we could get more to add to the bounty it might grease the wheels of innovation. http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/c6eb7827/attachment-0002.html From jason at jasonjgw.net Thu Nov 19 17:15:42 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Nov 2009 12:15:42 +1100 Subject: [Freeswitch-users] RTP issues (possibly nat-related) Message-ID: <20091120011542.GA20754@jdc.jasonjgw.net> I have upgraded FreeSWITCH several times recently for testing purposes. Also, my router's configuration has changed slightly as I have moved from tunneled IPv6 to a new native IPv6-over-ADSL trial. However, the problem now is related to my ISP's IPv4-only SIP service, and the symptoms are as follows. 1. If I call a test number, sometimes it all works perfectly. 2. On other occasions (with no discernible pattern) the call connects but no audio is received from the remote end. When this occurs, tshark shows that rtp packets are being sent out to the correct IPv4 address of the server. I am using Stun to handle nat, as my router does not support any of the nat configuration protocols. I want to establish whether it's a router issue or a FreeSWITCH problem. The router is going to be replaced eventually with a small form-factor Linux box and an ADSL2+ card from Traverse Technologies (http://www.traverse.com.au/), but given my priorities at the moment, it won't happen until next year. I can compare SIP traces of that would help. From brian at freeswitch.org Thu Nov 19 17:25:59 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 19:25:59 -0600 Subject: [Freeswitch-users] RTP issues (possibly nat-related) In-Reply-To: <20091120011542.GA20754@jdc.jasonjgw.net> References: <20091120011542.GA20754@jdc.jasonjgw.net> Message-ID: <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> I think the fix for this is coming to an SVN repo near you... so give it a few and update. /b On Nov 19, 2009, at 7:15 PM, Jason White wrote: > I have upgraded FreeSWITCH several times recently for testing > purposes. Also, > my router's configuration has changed slightly as I have moved from > tunneled > IPv6 to a new native IPv6-over-ADSL trial. > > However, the problem now is related to my ISP's IPv4-only SIP > service, and the > symptoms are as follows. > > 1. If I call a test number, sometimes it all works perfectly. > > 2. On other occasions (with no discernible pattern) the call > connects but no > audio is received from the remote end. > > When this occurs, tshark shows that rtp packets are being sent out > to the > correct IPv4 address of the server. > > I am using Stun to handle nat, as my router does not support any of > the nat > configuration protocols. I want to establish whether it's a router > issue or a > FreeSWITCH problem. The router is going to be replaced eventually > with a small > form-factor Linux box and an ADSL2+ card from Traverse Technologies > (http://www.traverse.com.au/), but given my priorities at the > moment, it won't > happen until next year. > > I can compare SIP traces of that would help. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From steveu at coppice.org Thu Nov 19 17:57:34 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 20 Nov 2009 09:57:34 +0800 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <1258665321.12582.6.camel@localhost.localdomain> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> <1258646095.12820.300.camel@localhost.localdomain> <4B056F13.6050106@coppice.org> <1258665321.12582.6.camel@localhost.localdomain> Message-ID: <4B05F78E.3080007@coppice.org> On 11/20/2009 05:15 AM, David Knell wrote: > On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote: > > >> The audio path between kernel and user space is not stable with any >> current PC based telephony system. At some point in the day the odd >> chunk of data is lost here and there, whether you use asterisk, >> callweaver, yate or FS, with dahdi or sangoma. This is the key problem >> for user space echo cancellation. When the path hiccups, the EC goes >> crazy, and howls. So far kernel space EC has been the only way to keep >> the path length rock solid. >> > Why do you think this is? Getting data from kernel space to user space > isn't something which it's difficult to do reliably: the disk system > manages it. Even if data is being lost, buffer overruns can be dealt > with by using bigger buffers, or timestamping blocks of data on their > way in so that missing blocks can be detected. > Disk isn't audio. Audio is real time, and real time constraints are a harsh mistress. Big buffers are out of the question, due to latency. Some mitigation could be provided if you can detect where missing chunks occur and their exact size. Right now, the I/O schemes do not provide for that, and incorporating support would be tough. You'd need some out of band indication, like an ioctl or something, which would lead to more user space/kernel space exchanges, further increasing the problem. Steve From brian at freeswitch.org Thu Nov 19 18:17:54 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Nov 2009 20:17:54 -0600 Subject: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv In-Reply-To: <87f2f3b90911191301q1083e23fs63fc11722bb60aa5@mail.gmail.com> References: <87f2f3b90911191301q1083e23fs63fc11722bb60aa5@mail.gmail.com> Message-ID: <3E187F0F-66DC-4282-8643-90662F7863BA@freeswitch.org> It now works.. update and have fun! /b On Nov 19, 2009, at 3:01 PM, Michael Collins wrote: > > > On Thu, Nov 19, 2009 at 12:49 PM, Brian West > wrote: > We have removed the two modules using the reference code from > BroadVoice and added a lib with a new interface from Steve Underwood > and mod_bv.c using this lib... We know their is ONE last bug to be > fixed in the lib before its working so please do not open any jira's > if you try to run it because it will crash right now. > > Thanks for your understanding and once this is fixed it'll work with > aastra and x-lite on both 32bit and 64bit systems without any issues. > > Thanks, > Brian West > > Thanks to Brian, Tony, Mike, and Steve U. for all their hard work on > this. Not only did they get this implemented quickly, they found a > few bugs and reported back to the Broadcom guys. :) Excellent work > all the way around. > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/c2ea4953/attachment-0002.html From dfansler at dv-fansler.com Thu Nov 19 18:18:36 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Thu, 19 Nov 2009 21:18:36 -0500 Subject: [Freeswitch-users] APT Utility In-Reply-To: References: <005a01ca6898$f16d99d0$d448cd70$@com> Message-ID: <000301ca6987$c47edbb0$4d7c9310$@com> Thanks for your answers Rob and Shelby. I found more info on apt-get and ran it against all the missing dependences noted. I also ran through the sequence of commands Shelby suggested. In the end, running dpkg -checkbuilddeps I got the following in return dpkg-checkbuilddeps: Unnet builddependencies: debhelper (>=5) then followed the instructions for Ubuntu to enable freeswitch nano /etc/default/freeswitch FREESWITCH_ENABLE="true" And then tried invoke -rc.d freeswitch start but nothing obvious happened. I am only using Ubuntu since it came as a free DVD in the Linux Pro mag that the article about Freeswitch was in. Is there a better version of Linux to use? thanks David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Wednesday, November 18, 2009 5:53 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] APT Utility Hi David, When using Apt, you would install packages with: apt-get install Or search for packages with apt-cache search If you're not root, you'll need to stick "sudo " in front of those command. Honestly, you might want to find a better tutorial with explicit command-by-command instructions. Good luck! Rob On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: Greetings - I am trying to startup a freeSwitch on a P4 running Ubuntu 9.04 "Jaunty". I know very little about Linux. I decided to try this after reading the article in Linux Pro Magazine. I have been following the detailed instructions in the wiki for using Ubuntu Jaunty, however I have run into an unknown - "Use your favorite APT utility to get the needed packages". I am good at following direct instructions - but this statement is too vague for my minimal minimal - did I mention minimal - knowledge of Linux. Could someone please give me detailed instructions on how to use APT utility to get the needed packages - and what are the needed packages? Thanks kindly, David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/06a798be/attachment-0002.html From ujjval at simplesignal.com Thu Nov 19 18:37:06 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Thu, 19 Nov 2009 18:37:06 -0800 Subject: [Freeswitch-users] Setting up Conference with Moderator In-Reply-To: <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local> <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>, <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com> <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71702E7CD84@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF71703077A38@EXMBXCLUS01.citservers.local> <118F3AD6-E4CA-4933-970B-5A9C018FDFFE@gmail.com> <3C04B27FC880044F8FCD735D0D952FF7175DAC46C8@EXMBXCLUS01.citservers.local> <68CA7433-C8FE-4108-BA1C-529F28634772@gmail.com> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C5A@EXMBXCLUS01.citservers.local> Cool, I will explore that option when I have some time. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Wednesday, November 18, 2009 11:02 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hi again UK, IVR is designed to naturally return to previous or top menus. I don't think there's a way to change this default behavior. Maybe its time to move to a script-based pin validation system so you have the full control you need. http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR Rob On Nov 18, 2009, at 11:34 PM, Ujjval Karihaloo wrote: > I have used the following setting in ivr.conf.xml to setup > conferencing with moderator. > > However, the issue I have is - the user enters 123456 and then say > if it's a moderator they enter wrong Moderator PIN 3 times then it > takes the user back to the main menu..."conference_menu" and asks > for main conf pin (123456) once again. > > I would like the caller to be disconnected if they get into the > Moderator menu and enter wrong Moderator PIN 3 times. > > greet-long="welcome_please_enter_conference_pin.wav" > greet-short="check_and_try_again.wav" > invalid-sound="passcode_invalid.wav" > exit-sound="voicemail/vm-goodbye.wav" > timeout="10000" > inter-digit-timeout="5000" > max-failures="3" > max-timeouts="3" > digit-len="7"> > param="conference_123456_moderator_menu" /> > > > greet- > long > = > "conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant > .wav" > greet-short="check_moderator_pin_or_1_to_join.wav" > invalid-sound="invalid_moderator_pin.wav" > exit-sound="voicemail/vm-goodbye.wav" > timeout="10000" > inter-digit-timeout="5000" > max-failures="3" > max-timeouts="3" > digit-len="5"> > > > > > > > > > Ujjval Karihaloo > VP Voice Engineering > IP Phone: +13032428610 > E-Fax: +17202391690 > > SimpleSignal Inc. > 88 Inverness Circle East > Suite K105 > Englewood, CO 80112 > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Thursday, November 05, 2009 7:52 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hi UK, > > From what I've done and read, the caller-controls (in > conference.conf.xml) can be modified to almost anything you can think > of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a > caller control just for those with the moderator flag. So unless you > want everyone able to mute/kick everyone then you can't do it. > > The wiki seems to indicate this as well: > > "Be aware that the caller-controls are applied across the entire > conference. You cannot enter one member of the conference using > caller- > controls ABC and then enter a second member using caller-controls > XYZ." > > http://wiki.freeswitch.org/wiki/Mod_conference > > > I think this might be a limitation of mod_conference. Perhaps one of > the pros can chime in if I'm off-base or there's some nifty way to > accomplish this. > > Cheers, > Rob > > On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > >> Any ideas on the below...has anyone implemented the below: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ujjval Karihaloo >> Sent: Monday, November 02, 2009 12:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Rob: >> >> Once I have the Moderator and Participants logged on, how do I >> invoke the moderator previlidges, LIk esay muting everyone/someone >> or kicking someone out of the Conf and the like? >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 9:34 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> Hm, strange. I haven't seen that before. Can you pastebin your logs >> at debug level? >> >> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: >> >>> It's strange... a tcpdump tells me that there is no DTMF from my >>> provider when using IVR, but when I call into a TN that goes >>> directly into the Conference App, I see DTMF from the provider. >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Rob Forman >>> Sent: Friday, October 30, 2009 7:23 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I've never had any problem with that. Is your logging at debug >>> level >>> so you can see the RECV DTFM in the log/fs_cli? Are you calling >>> from >>> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >>> isn't passing them through. >>> >>> Make sure your logging is turned up then try something simpler, like >>> calling the echo application, and see if DTFM comes through. >>> >>> Rob >>> >>> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >>> >>>> Rob: >>>> >>>> For some reason, I don't see the DTMF appear on the fs_CLI when >>>> using the below configuration....so it basically timesout. >>>> >>>> UK >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Ujjval Karihaloo >>>> Sent: Monday, October 26, 2009 9:21 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>> Moderator >>>> >>>> Thx a lot Rob, reading the wiki your way or using IVR seems >>>> correct.. >>>> =============== >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> ====================== >>>> >>>> I will try it out. >>>> >>>> Again thx a lot for your help. Will keep everyone posted. >>>> >>>> ________________________________________ >>>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org >>>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com] >>>> Sent: Friday, October 23, 2009 12:22 PM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Setting up Conference with >>>> Moderator >>>> >>>> I just re-tested with the pin in my dial plan: >>>> >>>> >>>> >>>> And it doesn't challenge me for the pin. I just drop right in. I >>>> figured this is how it was intended, since the wiki says the pin is >>>> set initially and only challenged in later attempts [by future >>>> callers]: >>>> >>>> "The first time a conference name (confname) is used, it will be >>>> created on demand, and the pin will be set to what ever is >>>> specified >>>> at that time: the pin in the data string if specified, or if not, >>>> the >>>> "pin" setting in the conference profile, and if that is also >>>> unspecified, then there is no pin protection. Any later attempt to >>>> join the conference must specify the same pin number, if one >>>> existed >>>> when it was created. " >>>> >>>> >>>> The wiki also says that the wait-mod might be "used in conjunction >>>> with an IVR where the moderators are authenticated with an extra >>>> pass- >>>> code", which is what I did. I guess that's why I didn't understand >>>> the point of the +pin. >>>> >>>> I'm sure there's a scenario where its used and useful, the wiki >>>> just >>>> doesn't explain it. >>>> >>>> Rob >>>> >>>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>>> >>>>> Well first off you're not defining a pine here... >>>>> >>>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference >>>>> pin >>>>> number] >>>>> >>>>> That might be why its not asking for a pin. >>>>> >>>>> /b >>>>> >>>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From b0ef at esben-stien.name Thu Nov 19 19:16:55 2009 From: b0ef at esben-stien.name (Esben Stien) Date: Fri, 20 Nov 2009 04:16:55 +0100 Subject: [Freeswitch-users] Freeswitch Video Capture and Playback Message-ID: <87k4xlga1k.fsf@quasar.esben-stien.name> I'm using ekiga with mod_fsv, trying to record and play back video. When I dial the record extension, it seems to record something, as the video file gets bigger. Trying then to dial the extension for play back, just hangs up, with freeswitch saying: od_fsv.c:247 File version does not match! There seems to be no information on the FSV format or the mod_fsv module on the wiki. Is this at all supposed to work?. What clients and codecs were successful?. Any pointers as to what I can try?. -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n From anthony.minessale at gmail.com Thu Nov 19 18:49:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Nov 2009 20:49:40 -0600 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: References: Message-ID: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> Your "annoying behaviour" is the exact behavior you should be getting considering what you told FS to do. As soon as you call uuid_bridge you are transferring both legs of the call to bridge to each other. This means your java app must exit so the channels can connect to each other. remember that you hangup hook can be called when the channel is transferred not only when it hangs up. you have to test which is happening based on the input to your callback. On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: > Hi there! > > I've got annoying FS behavior: > There are 2 channels executing the same Java application (application > itself is an IVR). If I try to bridge them with uuid_bridged then both > channels are killed. Here is a log from FS console: > uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 > 7d6c08fc-62bf-4a6c-a9ae-763d607e43de > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 > (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> > CS_HIBERNATE > 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called > 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1005 at 192.168.147.130 [BREAK] > 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 > (sofia/internal/1001 at master.agent.starpoundtech.net) State Change > CS_EXECUTE -> CS_HIBERNATE > 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called > API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 > 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: > +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de > > freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] > switch_core_session.c:933 Send signal > sofia/internal/1001 at master.agent.starpoundtec > 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal > sofia/internal/1005 at 192.168.147.130 [BREAK] > > 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal > sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] > > (FS version is 1.0.4) > > Any thoughts? > > > Artem > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/8c06b56f/attachment-0002.html From ujjval at simplesignal.com Thu Nov 19 19:17:32 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Thu, 19 Nov 2009 19:17:32 -0800 Subject: [Freeswitch-users] upgrading to latest SVN Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> Getting error below..not sure whats wrong..which line number in what file does this refer to? [root at ss_freeswitch log]# freeswitch 2009-11-19 20:15:44.725118 [INFO] switch_event.c:568 Activate Eventing Engine. 2009-11-19 20:15:44.727095 [DEBUG] switch_event.c:556 Create event dispatch thread 0 Cannot Initialize [[error near line 733]: missing >] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/351b99b6/attachment-0002.html From jason at jasonjgw.net Thu Nov 19 19:30:07 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Nov 2009 14:30:07 +1100 Subject: [Freeswitch-users] upgrading to latest SVN In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> Message-ID: <20091120033007.GA23643@jdc.jasonjgw.net> Ujjval Karihaloo wrote: > Getting error below..not sure whats wrong..which line number in what file > does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. From rob4manhere at gmail.com Thu Nov 19 19:41:30 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 19 Nov 2009 21:41:30 -0600 Subject: [Freeswitch-users] APT Utility In-Reply-To: <000301ca6987$c47edbb0$4d7c9310$@com> References: <005a01ca6898$f16d99d0$d448cd70$@com> <000301ca6987$c47edbb0$4d7c9310$@com> Message-ID: <0B55F774-9F77-4B4D-891D-7FD9595E644A@gmail.com> Ubuntu has pretty good package management with apt-get and should work well for a beginner. The recommended OS (though FreeSWITCH runs on a wide variety of platforms) is 64-bit CentOS. You can get it here: http://www.centos.org/ if you'd like, but at this point I think it's fine to just keep digging into whichever flavor of linux you have handy. If you have FreeSWITCH compiled and installed, have you tried just starting it from the command line? cd /usr/local/freeswitch ; ./bin/ freeswitch On Nov 19, 2009, at 8:18 PM, David V. Fansler wrote: > Thanks for your answers Rob and Shelby. I found more info on apt- > get and ran it against all the missing dependences noted. I also > ran through the sequence of commands Shelby suggested. In the end, > running dpkg ?checkbuilddeps I got the following in return > > dpkg-checkbuilddeps: Unnet builddependencies: debhelper (>=5) > > then followed the instructions for Ubuntu to enable freeswitch > nano /etc/default/freeswitch > FREESWITCH_ENABLE=?true? > > And then tried > invoke ?rc.d freeswitch start > but nothing obvious happened. > > I am only using Ubuntu since it came as a free DVD in the Linux Pro > mag that the article about Freeswitch was in. Is there a better > version of Linux to use? > thanks > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Wednesday, November 18, 2009 5:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] APT Utility > > Hi David, > > When using Apt, you would install packages with: > > apt-get install > > Or search for packages with > > apt-cache search > > > If you're not root, you'll need to stick "sudo " in front of those > command. Honestly, you might want to find a better tutorial with > explicit command-by-command instructions. > > Good luck! > Rob > > On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: > > > Greetings ? I am trying to startup a freeSwitch on a P4 running > Ubuntu 9.04 ?Jaunty?. I know very little about Linux. I decided to > try this after reading the article in Linux Pro Magazine. I have > been following the detailed instructions in the wiki for using > Ubuntu Jaunty, however I have run into an unknown ? ?Use your > favorite APT utility to get the needed packages?. > I am good at following direct instructions ? but this statement is > too vague for my minimal minimal ? did I mention minimal - knowledge > of Linux. > > Could someone please give me detailed instructions on how to use APT > utility to get the needed packages ? and what are the needed packages? > Thanks kindly, > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091119/1e8440aa/attachment-0002.html From ujjval at simplesignal.com Thu Nov 19 19:49:06 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Thu, 19 Nov 2009 19:49:06 -0800 Subject: [Freeswitch-users] upgrading to latest SVN In-Reply-To: <20091120033007.GA23643@jdc.jasonjgw.net> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> <20091120033007.GA23643@jdc.jasonjgw.net> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C67@EXMBXCLUS01.citservers.local> I really didn't change anything. I was running 1.0.4 and now built from SVN...I see the oddly placed entry in ivr.conf.xml ...removed it now its error on somewhere here: ERROR is: [root at ss_freeswitch freeswitch]# freeswitch 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event dispatch thread 0 Cannot Initialize [[error near line 2840]: unexpected closing tag ] -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, November 19, 2009 8:30 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN Ujjval Karihaloo wrote: > Getting error below..not sure whats wrong..which line number in what file > does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ujjval at simplesignal.com Thu Nov 19 21:22:39 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Thu, 19 Nov 2009 21:22:39 -0800 Subject: [Freeswitch-users] upgrading to latest SVN In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C67@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> <20091120033007.GA23643@jdc.jasonjgw.net> <3C04B27FC880044F8FCD735D0D952FF7175DAC4C67@EXMBXCLUS01.citservers.local> Message-ID: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C72@EXMBXCLUS01.citservers.local> Does svn update try to merge the config files..Need some help, I think it has added some entries in my config files that is causing tag mismatches.. Please advise how to get back my orig config? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo Sent: Thursday, November 19, 2009 8:49 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN I really didn't change anything. I was running 1.0.4 and now built from SVN...I see the oddly placed entry in ivr.conf.xml ...removed it now its error on somewhere here: ERROR is: [root at ss_freeswitch freeswitch]# freeswitch 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event dispatch thread 0 Cannot Initialize [[error near line 2840]: unexpected closing tag ] -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, November 19, 2009 8:30 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN Ujjval Karihaloo wrote: > Getting error below..not sure whats wrong..which line number in what file > does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mrene_lists at avgs.ca Thu Nov 19 21:30:33 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 19 Nov 2009 21:30:33 -0800 Subject: [Freeswitch-users] upgrading to latest SVN In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C72@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> <20091120033007.GA23643@jdc.jasonjgw.net> <3C04B27FC880044F8FCD735D0D952FF7175DAC4C67@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC4C72@EXMBXCLUS01.citservers.local> Message-ID: Nothing will replace your configs automatically in the whole build system. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 19-Nov-09, at 9:22 PM, Ujjval Karihaloo wrote: > Does svn update try to merge the config files..Need some help, I > think it has added some entries in my config files that is causing > tag mismatches.. > > Please advise how to get back my orig config? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Ujjval Karihaloo > Sent: Thursday, November 19, 2009 8:49 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] upgrading to latest SVN > > I really didn't change anything. > > I was running 1.0.4 and now built from SVN...I see the oddly placed > entry in ivr.conf.xml > > ...removed it now its error on > somewhere here: > > > > > > > > > > > > > > ERROR is: > [root at ss_freeswitch freeswitch]# freeswitch > 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate > Eventing Engine. > 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event > dispatch thread 0 > Cannot Initialize [[error near line 2840]: unexpected closing tag section>] > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Jason White > Sent: Thursday, November 19, 2009 8:30 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] upgrading to latest SVN > > Ujjval Karihaloo wrote: >> Getting error below..not sure whats wrong..which line number in >> what file >> does this refer to? > > freeswitch/log/freeswitch.xml.fsxml > > This will be due to a syntax error somewhere in your configuration. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Nov 19 21:37:10 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 20 Nov 2009 00:37:10 -0500 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: <4B0387F1.7070105@savion.huji.ac.il> References: <4B0266F4.8070602@savion.huji.ac.il> <4B0387F1.7070105@savion.huji.ac.il> Message-ID: <193640CC-3E62-4248-8E80-CE7FE82653C0@jerris.com> check out sofia_contact function. If you use this in combination with binding profiles together so they are one table I think this should work right. Mike On Nov 18, 2009, at 12:36 AM, Eli Hayun wrote: > Brian West wrote: >> >> Why do you need to know the destination profile like that? You get to >> pick that on your own so you should already know that before hand. >> >> >> /b >> >> On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote: >> >> >>> Hi >>> We have more then one profile. To make a call I have to enter : bridge >>> sofia/profile/number at ip >>> The problem is when I use : "${use_profile}" I am getting the caller >>> profile, and I need the destination profile. >>> >>> How do I get this information? >>> >> > Thanks for your answer. > > The problem is when I call to that number that the phone hook to other server, I cannot make the call. > Is there is a variable that can tell me the destination profile? > Lets say the other profile called "ph1" I have to dial > sofia/ph1/xxxxx at host to make the call. Is there other way to do that? From jason at jasonjgw.net Thu Nov 19 21:42:03 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Nov 2009 16:42:03 +1100 Subject: [Freeswitch-users] upgrading to latest SVN In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C72@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF7175DAC4C63@EXMBXCLUS01.citservers.local> <20091120033007.GA23643@jdc.jasonjgw.net> <3C04B27FC880044F8FCD735D0D952FF7175DAC4C67@EXMBXCLUS01.citservers.local> <3C04B27FC880044F8FCD735D0D952FF7175DAC4C72@EXMBXCLUS01.citservers.local> Message-ID: <20091120054203.GA26970@jdc.jasonjgw.net> Ujjval Karihaloo wrote: > Does svn update try to merge the config files..Need some help, I think it > has added some entries in my config files that is causing tag mismatches.. Building and installing FreeSWITCH won't interfere with your configuration. I would suggest using Git or another version control system to keep track of configuration files. I prefer Git. From frank at carmickle.com Thu Nov 19 22:13:47 2009 From: frank at carmickle.com (Frank Carmickle) Date: Fri, 20 Nov 2009 01:13:47 -0500 Subject: [Freeswitch-users] Another Group Question - on VoiceMail In-Reply-To: References: Message-ID: <20091120061347.GB31924@base.carmickle.com> On Thu, Nov 19, Joseph L. Casale wrote: > >Is there any way that each extension in the group can be notified that a > >group Voice Mail is waiting to be picked up so that each phone shows the > >message waiting indication ? > > Wouldn't this be simply accomplished by setting the vicemail as box 100 for > each of the users (such as ext 101....1xx)? Check out the directory parameter MWI-Account. Along with setting the mailbox variable. The variable mailbox sets what voicemail box a person dialing the extension will be dropped in to and MWI_Account, the param, will tell the phone what mailbox to subscribe to. HTH --FC From jason at jasonjgw.net Thu Nov 19 23:01:37 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Nov 2009 18:01:37 +1100 Subject: [Freeswitch-users] RTP issues (possibly nat-related) In-Reply-To: <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> References: <20091120011542.GA20754@jdc.jasonjgw.net> <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> Message-ID: <20091120070137.GA28316@jdc.jasonjgw.net> Brian West wrote: > I think the fix for this is coming to an SVN repo near you... so give > it a few and update. Thanks Brian! I'll watch the svn logs and update when the fix lands. From brian at freeswitch.org Thu Nov 19 23:13:16 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 01:13:16 -0600 Subject: [Freeswitch-users] RTP issues (possibly nat-related) In-Reply-To: <20091120070137.GA28316@jdc.jasonjgw.net> References: <20091120011542.GA20754@jdc.jasonjgw.net> <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> <20091120070137.GA28316@jdc.jasonjgw.net> Message-ID: Update and see if the problem is gone. /b On Nov 20, 2009, at 1:01 AM, Jason White wrote: > > Thanks Brian! > > I'll watch the svn logs and update when the fix lands. From jason at jasonjgw.net Fri Nov 20 00:32:18 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Nov 2009 19:32:18 +1100 Subject: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv In-Reply-To: <3E187F0F-66DC-4282-8643-90662F7863BA@freeswitch.org> References: <87f2f3b90911191301q1083e23fs63fc11722bb60aa5@mail.gmail.com> <3E187F0F-66DC-4282-8643-90662F7863BA@freeswitch.org> Message-ID: <20091120083218.GA17939@jdc.jasonjgw.net> Brian West wrote: > It now works.. update and have fun! Unfortunately it fails to compile under Debian: mod_bv.c:33:24: error: broadvoice.h: No such file or directory The header file exists, so I assume the include path specified to gcc just isn't right. From dfansler at dv-fansler.com Fri Nov 20 03:02:48 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Fri, 20 Nov 2009 06:02:48 -0500 Subject: [Freeswitch-users] APT Utility In-Reply-To: <0B55F774-9F77-4B4D-891D-7FD9595E644A@gmail.com> References: <005a01ca6898$f16d99d0$d448cd70$@com> <000301ca6987$c47edbb0$4d7c9310$@com> <0B55F774-9F77-4B4D-891D-7FD9595E644A@gmail.com> Message-ID: <005301ca69d0$ff7aea80$fe70bf80$@com> Thanks Rob - I guess that it is not installed yet - I get a directory not found trying your suggestion. Taking a look at the freeswitch directory, there is no bin directory. I will keep scratching away at it. David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Thursday, November 19, 2009 10:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] APT Utility Ubuntu has pretty good package management with apt-get and should work well for a beginner. The recommended OS (though FreeSWITCH runs on a wide variety of platforms) is 64-bit CentOS. You can get it here: http://www.centos.org/ if you'd like, but at this point I think it's fine to just keep digging into whichever flavor of linux you have handy. If you have FreeSWITCH compiled and installed, have you tried just starting it from the command line? cd /usr/local/freeswitch ; ./bin/freeswitch On Nov 19, 2009, at 8:18 PM, David V. Fansler wrote: Thanks for your answers Rob and Shelby. I found more info on apt-get and ran it against all the missing dependences noted. I also ran through the sequence of commands Shelby suggested. In the end, running dpkg -checkbuilddeps I got the following in return dpkg-checkbuilddeps: Unnet builddependencies: debhelper (>=5) then followed the instructions for Ubuntu to enable freeswitch nano /etc/default/freeswitch FREESWITCH_ENABLE="true" And then tried invoke -rc.d freeswitch start but nothing obvious happened. I am only using Ubuntu since it came as a free DVD in the Linux Pro mag that the article about Freeswitch was in. Is there a better version of Linux to use? thanks David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman Sent: Wednesday, November 18, 2009 5:53 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] APT Utility Hi David, When using Apt, you would install packages with: apt-get install Or search for packages with apt-cache search If you're not root, you'll need to stick "sudo " in front of those command. Honestly, you might want to find a better tutorial with explicit command-by-command instructions. Good luck! Rob On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: Greetings - I am trying to startup a freeSwitch on a P4 running Ubuntu 9.04 "Jaunty". I know very little about Linux. I decided to try this after reading the article in Linux Pro Magazine. I have been following the detailed instructions in the wiki for using Ubuntu Jaunty, however I have run into an unknown - "Use your favorite APT utility to get the needed packages". I am good at following direct instructions - but this statement is too vague for my minimal minimal - did I mention minimal - knowledge of Linux. Could someone please give me detailed instructions on how to use APT utility to get the needed packages - and what are the needed packages? Thanks kindly, David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/4c572b1f/attachment-0002.html From rob4manhere at gmail.com Fri Nov 20 03:56:09 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Fri, 20 Nov 2009 05:56:09 -0600 Subject: [Freeswitch-users] APT Utility In-Reply-To: <005301ca69d0$ff7aea80$fe70bf80$@com> References: <005a01ca6898$f16d99d0$d448cd70$@com> <000301ca6987$c47edbb0$4d7c9310$@com> <0B55F774-9F77-4B4D-891D-7FD9595E644A@gmail.com> <005301ca69d0$ff7aea80$fe70bf80$@com> Message-ID: <7296F933-0972-47A7-B988-01557C0BDCC5@gmail.com> Hi David, Apt was just for getting your dependencies in order. Now you can go about the business of compiling and installing Freeswitch. You might start with 1.0.4 so you don't have to mess with svn yet. http://wiki.freeswitch.org/wiki/Installation_Guide#FreeSWITCH_1.0.4_.22Phoenix.22_Release Just keep reading through the wiki or google for tutorials. Good luck! Rob On Nov 20, 2009, at 5:02 AM, David V. Fansler wrote: > Thanks Rob ? I guess that it is not installed yet ? I get a > directory not found trying your suggestion. Taking a look at the > freeswitch directory, there is no bin directory. I will keep > scratching away at it. > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Thursday, November 19, 2009 10:42 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] APT Utility > > Ubuntu has pretty good package management with apt-get and should > work well for a beginner. The recommended OS (though FreeSWITCH > runs on a wide variety of platforms) is 64-bit CentOS. You can get > it here: http://www.centos.org/ if you'd like, but at this point I > think it's fine to just keep digging into whichever flavor of linux > you have handy. > > If you have FreeSWITCH compiled and installed, have you tried just > starting it from the command line? cd /usr/local/freeswitch ; ./bin/ > freeswitch > > > On Nov 19, 2009, at 8:18 PM, David V. Fansler wrote: > > > Thanks for your answers Rob and Shelby. I found more info on apt- > get and ran it against all the missing dependences noted. I also > ran through the sequence of commands Shelby suggested. In the end, > running dpkg ?checkbuilddeps I got the following in return > > dpkg-checkbuilddeps: Unnet builddependencies: debhelper (>=5) > > then followed the instructions for Ubuntu to enable freeswitch > nano /etc/default/freeswitch > FREESWITCH_ENABLE=?true? > > And then tried > invoke ?rc.d freeswitch start > but nothing obvious happened. > > I am only using Ubuntu since it came as a free DVD in the Linux Pro > mag that the article about Freeswitch was in. Is there a better > version of Linux to use? > thanks > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Wednesday, November 18, 2009 5:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] APT Utility > > Hi David, > > When using Apt, you would install packages with: > > apt-get install > > Or search for packages with > > apt-cache search > > > If you're not root, you'll need to stick "sudo " in front of those > command. Honestly, you might want to find a better tutorial with > explicit command-by-command instructions. > > Good luck! > Rob > > On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: > > > > Greetings ? I am trying to startup a freeSwitch on a P4 running > Ubuntu 9.04 ?Jaunty?. I know very little about Linux. I decided to > try this after reading the article in Linux Pro Magazine. I have > been following the detailed instructions in the wiki for using > Ubuntu Jaunty, however I have run into an unknown ? ?Use your > favorite APT utility to get the needed packages?. > I am good at following direct instructions ? but this statement is > too vague for my minimal minimal ? did I mention minimal - knowledge > of Linux. > > Could someone please give me detailed instructions on how to use APT > utility to get the needed packages ? and what are the needed packages? > Thanks kindly, > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/454fac6e/attachment-0002.html From siniypin at gmail.com Fri Nov 20 04:30:00 2009 From: siniypin at gmail.com (RobertT) Date: Fri, 20 Nov 2009 15:30:00 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> Message-ID: <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> Well, I start 2 user agents. Each of them successfully registers as 1000 & 1001 extensions via tcp SIP transport. Then I issue a call, say from 1000 to 1001, and watch it being disconnected in several seconds by recieving client due to abovementioned conditions (no completing answer from FS). Why is it happening??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/a8b82cbb/attachment-0002.html From kjv at ken-ton.com Fri Nov 20 05:44:13 2009 From: kjv at ken-ton.com (Karl J. Vesterling) Date: Fri, 20 Nov 2009 08:44:13 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> <868A4E38-D947-4291-BBD7-4F4C9E5B239E@ken-ton.com> Message-ID: <257758F8-77DF-4D05-BA80-B3E115CCD5AB@ken-ton.com> Then why is the 7540 in the Subject of this thread? I hadn't found data on the 7540 either, but hey, for a while the Cisco WAAS device was a Cisco Product and had not any support from Cisco. Matter of fact, I had to get the regional sales director on the phone to argue successfully to the cisco support drone that the WAAS Devices were indeed Cisco devices, and that we were entitled to support, and that Cisco did have a support queue for the WAAS product. (It was quite humorous to listen in as the banter went back and forth. At which point I asked, "Do you now understand the source of my frustration?") (This wasn't a problem before since we had a back door into the WAAS developers, but our back door was on vacation at the time of this support request.) So, you can see my confusion here with regard to 7540 vs 7940... Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Nov 19, 2009, at 1:17 PM, Brian West wrote: > I don't think a 7540 exists. > > /b > > On Nov 19, 2009, at 12:11 PM, Karl J. Vesterling wrote: > >> I haven't provisioned any 7540's... Good luck! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From kjv at ken-ton.com Fri Nov 20 05:45:32 2009 From: kjv at ken-ton.com (Karl J. Vesterling) Date: Fri, 20 Nov 2009 08:45:32 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <4B05D37C.4000607@metik.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> <4B05D37C.4000607@metik.com> Message-ID: <40B86F87-D47E-4164-9ABD-794CA96B7B41@ken-ton.com> You'd think so wouldn't you... Even the DHCP Server with Snow Leopard Server lacks this basic functionality. If someone knows how to enable it, please post destructions on how-to here... Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Nov 19, 2009, at 6:23 PM, Metik wrote: > He should be able to just use "Additional Option" to add option 150 (and > the associated IP address to which the TFTP server is bound). > > Brian West wrote: >> Some Cisco phones need DHCP option 150. >> >> /b >> >> On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: >> >> >>> Metik, >>> >>> thanks a lot for the tip, I will certainly look at it, particularly >>> if it >>> does DHCP too. >>> >>> At the moment, I use my ADSL Router to provide DHCP to the network >>> but I've >>> just discovered that you can't configure options in its DHCP server >>> to point >>> to the TFTP server for the phone. At the moment, I have to have the >>> phone >>> set to a static IP address to be able to configure the TFTP server >>> address >>> which is not as flexible as using DHCP. I had thought about changing >>> over to >>> use Windows Server DHCP services but it sounds like ttpd32 would do >>> the >>> trick. >>> >>> I just need to decide whether I want all of my machines to rely on >>> getting >>> their IP address from another PC - it feels like having DHCP in the >>> router >>> is more robust. >>> >>> Regards >>> Dave >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Fri Nov 20 06:25:44 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 08:25:44 -0600 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> Message-ID: <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> Well depends are you using x-lite 4 beta? you didn't include ANY logs... I know TCP to TCP works fine I use that daily. can you include some debug logs or join #freeswitch on irc.freenode.net? /b On Nov 20, 2009, at 6:30 AM, RobertT wrote: > Well, I start 2 user agents. Each of them successfully registers as > 1000 & 1001 extensions via tcp SIP transport. Then I issue a call, > say from 1000 to 1001, and watch it being disconnected in several > seconds by recieving client due to abovementioned conditions (no > completing answer from FS). Why is it happening??? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Fri Nov 20 06:33:06 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 08:33:06 -0600 Subject: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv In-Reply-To: <20091120083218.GA17939@jdc.jasonjgw.net> References: <87f2f3b90911191301q1083e23fs63fc11722bb60aa5@mail.gmail.com> <3E187F0F-66DC-4282-8643-90662F7863BA@freeswitch.org> <20091120083218.GA17939@jdc.jasonjgw.net> Message-ID: You'll have to rebootstrap and make sure libs/broadvoice builds /b On Nov 20, 2009, at 2:32 AM, Jason White wrote: > Brian West wrote: >> It now works.. update and have fun! > > Unfortunately it fails to compile under Debian: > > mod_bv.c:33:24: error: broadvoice.h: No such file or directory > > The header file exists, so I assume the include path specified to > gcc just > isn't right. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/2b08e373/attachment-0002.html From dave at 3c.co.uk Fri Nov 20 07:51:37 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 20 Nov 2009 08:51:37 -0700 Subject: [Freeswitch-users] Hardware echo cancellation. In-Reply-To: <4B05F78E.3080007@coppice.org> References: <855e4dcf0911181239w1327713dkf49f6273e7d47137@mail.gmail.com> <1258578249.12820.264.camel@localhost.localdomain> <855e4dcf0911181336s4ddd04f0r1be7a9289e7a826@mail.gmail.com> <1258587542.12820.275.camel@localhost.localdomain> <90A332CC-49CE-4763-A4A5-4C20E3C6759E@freeswitch.org> <1258646095.12820.300.camel@localhost.localdomain> <4B056F13.6050106@coppice.org> <1258665321.12582.6.camel@localhost.localdomain> <4B05F78E.3080007@coppice.org> Message-ID: <1258732297.12582.41.camel@localhost.localdomain> On Fri, 2009-11-20 at 09:57 +0800, Steve Underwood wrote: > On 11/20/2009 05:15 AM, David Knell wrote: > > On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote: > > > > > >> The audio path between kernel and user space is not stable with any > >> current PC based telephony system. At some point in the day the odd > >> chunk of data is lost here and there, whether you use asterisk, > >> callweaver, yate or FS, with dahdi or sangoma. This is the key problem > >> for user space echo cancellation. When the path hiccups, the EC goes > >> crazy, and howls. So far kernel space EC has been the only way to keep > >> the path length rock solid. > >> > > Why do you think this is? Getting data from kernel space to user space > > isn't something which it's difficult to do reliably: the disk system > > manages it. Even if data is being lost, buffer overruns can be dealt > > with by using bigger buffers, or timestamping blocks of data on their > > way in so that missing blocks can be detected. > > > Disk isn't audio. Audio is real time, and real time constraints are a > harsh mistress. Big buffers are out of the question, due to latency. Not necessarily. A decent-sized FIFO, mostly run empty, but there to buffer data in the case of the user-side not being able to accept it for a short period wouldn't necessarily add to latency unless it were needed. The user side could then make a decision as to how to deal with the queued data - dump it or handle it - according to its requirements. > Some mitigation could be provided if you can detect where missing chunks > occur and their exact size. Right now, the I/O schemes do not provide > for that, and incorporating support would be tough. You'd need some out > of band indication, like an ioctl or something, which would lead to more > user space/kernel space exchanges, further increasing the problem. I don't think it'd be all that hard. Were I to do this, I'd probably: - define an error return (ESLIP, EDATALOST, something like that) which might be returned by read/write - add an ioctl to enable and disable it - maybe add an ioctl to indicate how much data's been lost Doesn't break existing stuff, doesn't add any overhead under normal conditions, would be handy for better reliability with EC, DTMF, fax, etc. --Dave From jlenk at frontiernet.net Fri Nov 20 07:59:28 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Fri, 20 Nov 2009 07:59:28 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <1258634740580-4031590.post@n2.nabble.com> References: <1258634740580-4031590.post@n2.nabble.com> Message-ID: <1258732768082-4038514.post@n2.nabble.com> That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch-users-list at metik.com Fri Nov 20 08:04:24 2009 From: freeswitch-users-list at metik.com (Metik) Date: Fri, 20 Nov 2009 11:04:24 -0500 Subject: [Freeswitch-users] TFTP Server & Cisco 7540 In-Reply-To: <40B86F87-D47E-4164-9ABD-794CA96B7B41@ken-ton.com> References: <5D261645E0204E1C978DB31982CF7D6C@bp1.ad.bp.com><1258511897776-4023012.post@n2.nabble.com> <4B056BC7.6030009@metik.com> <921128D9-157F-469A-BE3B-55C5C348873E@freeswitch.org> <4B05D37C.4000607@metik.com> <40B86F87-D47E-4164-9ABD-794CA96B7B41@ken-ton.com> Message-ID: <4B06BE08.20703@metik.com> Although I'm not familiar with Snow Leopard's DHCP server implementation, I would assume that to expose that functionality--it is a matter of tweaking what is under the hood. If that is not the case--you could just build and/or install ISC's DHCP server. As far as tftpd32, I have used it in the past and it does support it. It is not by any means feature rich but should suffice given his needs. The other alternative is to install Cygwin-based build of the ISC DHCP server. -metik Karl J. Vesterling wrote: > You'd think so wouldn't you... > > Even the DHCP Server with Snow Leopard Server lacks this basic functionality. > If someone knows how to enable it, please post destructions on how-to here... > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 > > On Nov 19, 2009, at 6:23 PM, Metik wrote: > > >> He should be able to just use "Additional Option" to add option 150 (and >> the associated IP address to which the TFTP server is bound). >> >> Brian West wrote: >> >>> Some Cisco phones need DHCP option 150. >>> >>> /b >>> >>> On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: >>> >>> >>> >>>> Metik, >>>> >>>> thanks a lot for the tip, I will certainly look at it, particularly >>>> if it >>>> does DHCP too. >>>> >>>> At the moment, I use my ADSL Router to provide DHCP to the network >>>> but I've >>>> just discovered that you can't configure options in its DHCP server >>>> to point >>>> to the TFTP server for the phone. At the moment, I have to have the >>>> phone >>>> set to a static IP address to be able to configure the TFTP server >>>> address >>>> which is not as flexible as using DHCP. I had thought about changing >>>> over to >>>> use Windows Server DHCP services but it sounds like ttpd32 would do >>>> the >>>> trick. >>>> >>>> I just need to decide whether I want all of my machines to rely on >>>> getting >>>> their IP address from another PC - it feels like having DHCP in the >>>> router >>>> is more robust. >>>> >>>> Regards >>>> Dave >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From kristian.kielhofner at gmail.com Fri Nov 20 08:41:22 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 20 Nov 2009 11:41:22 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> Message-ID: <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> Finally got a chance to test this, the results are the same. Why am I getting this? Is it because I'm executing ring_ready before I attempt the bridge? Is it because I'm using a socket? On Wed, Nov 11, 2009 at 10:14 PM, Anthony Minessale wrote: > dont execute bridge that way, your bridge itself is the other thing already > broadcasting. > > > api uuid_transfer bridge:sofia/myprofile/foo at bar.com inline > > if you want to do more after the bridge > set the variable park_after_bridge=true to make it go back to idle > > > On Wed, Nov 11, 2009 at 4:33 PM, Kristian Kielhofner > wrote: >> >> Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64. >> >> On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner >> wrote: >> > From the trace: >> > >> ..snip.. >> >> -- >> Kristian Kielhofner >> http://www.astlinux.org >> http://blog.krisk.org >> http://www.star2star.com >> http://www.submityoursip.com >> http://www.voalte.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From info at daccii.it Fri Nov 20 08:46:17 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Fri, 20 Nov 2009 17:46:17 +0100 Subject: [Freeswitch-users] Call doesn't work while registration work for a VOIP provider In-Reply-To: References: <4B059CA7.3040201@daccii.it> Message-ID: <4B06C7D9.9010901@daccii.it> Thank you, this works perfectly! Brian West ha scritto: > I'm going to guess gw+exsorsa is what they don't like. try extensions- > in-contact=true on the gateway config. > > /b -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/82442704/attachment-0002.vcf From brian at freeswitch.org Fri Nov 20 08:49:50 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 10:49:50 -0600 Subject: [Freeswitch-users] Call doesn't work while registration work for a VOIP provider In-Reply-To: <4B06C7D9.9010901@daccii.it> References: <4B059CA7.3040201@daccii.it> <4B06C7D9.9010901@daccii.it> Message-ID: <6ADE7D71-9D61-485B-B829-005540D09610@freeswitch.org> Can you please document that on the wiki for the providers and in the sofia config pages? /b On Nov 20, 2009, at 10:46 AM, Albano Daniele Salvatore - Lavoro wrote: > Thank you, this works perfectly! > > Brian West ha scritto: >> I'm going to guess gw+exsorsa is what they don't like. try >> extensions- in-contact=true on the gateway config. >> /b > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From gauravs74 at yahoo.com Fri Nov 20 01:16:34 2009 From: gauravs74 at yahoo.com (Gaurav Singh) Date: Fri, 20 Nov 2009 01:16:34 -0800 (PST) Subject: [Freeswitch-users] Broadvoice 32 transcoding support? Message-ID: <144359.82983.qm@web113920.mail.gq1.yahoo.com> Hi, Does freeswitch support transcoding between broadvoice (BV32 ) and G711 ? Did anyone try using freeswitch with Xten/counterpath Sip phone configured with broadvoice32? Also, can someone recomend? another free sip phone supporting BV 32? Thanks Gaurav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/23f793ef/attachment-0002.html From igor at 3gnt.net Fri Nov 20 09:09:26 2009 From: igor at 3gnt.net (Igor Neves) Date: Fri, 20 Nov 2009 17:09:26 +0000 Subject: [Freeswitch-users] freeswitch.spec patch Message-ID: <4B06CD46.6050408@3gnt.net> Hi, Attached it's a patch that corrects the problem when doing upgrade to other older version of freeswitch rpm the freeswitch user was being deleted. This patch was made against freeswitch.spec from freeswitch 1.0.4. It should be applied with "patch -p0 < freeswitch.spec". Cheers, -- Igor Neves 3GNTW - Tecnologias de Informa??o, Lda SIP: igor at 3gnt.net JID: igor at 3gnt.net ICQ: 249075444 MSN: igor at 3gnt.net TLM: 00351914503611 PSTN: 00351252377120 -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: freesswitch.patch Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/b9ad9cbc/attachment-0002.pl From william.suffill at gmail.com Fri Nov 20 09:19:39 2009 From: william.suffill at gmail.com (William Suffill) Date: Fri, 20 Nov 2009 12:19:39 -0500 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <4B06CD46.6050408@3gnt.net> References: <4B06CD46.6050408@3gnt.net> Message-ID: <6b65470d0911200919y34546bvc93f3a5c976c1dc7@mail.gmail.com> 1.0.4 is quite old at this point so patches should be against trunk from SVN to make sure they apply against the latest codebase instead of the released version. Also the project prefers that patches be kept in Jira for tracking purposes. http://jira.freeswitch.org/ Thanks for the contribution though and the patch looks small enough that it should be easy to apply to trunk. From itamar at ispbrasil.com.br Fri Nov 20 09:20:47 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Fri, 20 Nov 2009 15:20:47 -0200 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <4B06CD46.6050408@3gnt.net> References: <4B06CD46.6050408@3gnt.net> Message-ID: do you like to help packaging it for fedora, centos and rhel ? On Fri, Nov 20, 2009 at 3:09 PM, Igor Neves wrote: > Hi, > > Attached it's a patch that corrects the problem when doing upgrade to other > older version of freeswitch rpm the freeswitch user was being deleted. > > This patch was made against freeswitch.spec from freeswitch 1.0.4. > It should be applied with "patch -p0 < freeswitch.spec". > > Cheers, > > -- > Igor Neves > 3GNTW - Tecnologias de Informa??o, Lda > > ?SIP: igor at 3gnt.net ? ? JID: igor at 3gnt.net > ?ICQ: 249075444 ? ? ? ? MSN: igor at 3gnt.net > ?TLM: 00351914503611 ? ?PSTN: 00351252377120 > ------------ Itamar Reis Peixoto e-mail/msn/google talk/sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From igor at 3gnt.net Fri Nov 20 09:29:06 2009 From: igor at 3gnt.net (Igor Neves) Date: Fri, 20 Nov 2009 17:29:06 +0000 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: References: <4B06CD46.6050408@3gnt.net> Message-ID: <4B06D1E2.3060008@3gnt.net> Hi, I have it working on CentOS, I can help packaging it, what does it involves more precisely? On 11/20/2009 05:20 PM, Itamar Reis Peixoto wrote: > do you like to help packaging it for fedora, centos and rhel ? > > > > On Fri, Nov 20, 2009 at 3:09 PM, Igor Neves wrote: > >> Hi, >> >> Attached it's a patch that corrects the problem when doing upgrade to other >> older version of freeswitch rpm the freeswitch user was being deleted. >> >> This patch was made against freeswitch.spec from freeswitch 1.0.4. >> It should be applied with "patch -p0< freeswitch.spec". >> >> Cheers, >> >> -- >> Igor Neves >> 3GNTW - Tecnologias de Informa??o, Lda >> >> SIP: igor at 3gnt.net JID: igor at 3gnt.net >> ICQ: 249075444 MSN: igor at 3gnt.net >> TLM: 00351914503611 PSTN: 00351252377120 >> >> > ------------ > > Itamar Reis Peixoto > > e-mail/msn/google talk/sip: itamar at ispbrasil.com.br > skype: itamarjp > icq: 81053601 > +55 11 4063 5033 > +55 34 3221 8599 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Igor Neves 3GNTW - Tecnologias de Informa??o, Lda SIP: igor at 3gnt.net JID: igor at 3gnt.net ICQ: 249075444 MSN: igor at 3gnt.net TLM: 00351914503611 PSTN: 00351252377120 From igor at 3gnt.net Fri Nov 20 09:31:18 2009 From: igor at 3gnt.net (Igor Neves) Date: Fri, 20 Nov 2009 17:31:18 +0000 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <6b65470d0911200919y34546bvc93f3a5c976c1dc7@mail.gmail.com> References: <4B06CD46.6050408@3gnt.net> <6b65470d0911200919y34546bvc93f3a5c976c1dc7@mail.gmail.com> Message-ID: <4B06D266.7020109@3gnt.net> Ok, But how should I proceed? Thanks, On 11/20/2009 05:19 PM, William Suffill wrote: > 1.0.4 is quite old at this point so patches should be against trunk > from SVN to make sure they apply against the latest codebase instead > of the released version. Also the project prefers that patches be kept > in Jira for tracking purposes. > > http://jira.freeswitch.org/ > > Thanks for the contribution though and the patch looks small enough > that it should be easy to apply to trunk. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Igor Neves 3GNTW - Tecnologias de Informa??o, Lda SIP: igor at 3gnt.net JID: igor at 3gnt.net ICQ: 249075444 MSN: igor at 3gnt.net TLM: 00351914503611 PSTN: 00351252377120 From brian at freeswitch.org Fri Nov 20 09:33:12 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 11:33:12 -0600 Subject: [Freeswitch-users] Broadvoice 32 transcoding support? In-Reply-To: <144359.82983.qm@web113920.mail.gq1.yahoo.com> References: <144359.82983.qm@web113920.mail.gq1.yahoo.com> Message-ID: <5C5BA5E8-0AB7-4251-8557-46E39503FEC9@freeswitch.org> Yes it works. You'll need SVN as of last night. Build mod_bv and load it. Works with Aastra and x-lite as far as I can tell... I do have one issue lingering with x-lite but its only when calling 9999 but i'm working on that one. I have no idea why you would want to transcode from BV32 to G711 since its a 16k to 8k resample too... you might as well use BV16 if you are doing that. /b On Nov 20, 2009, at 3:16 AM, Gaurav Singh wrote: > Hi, > > Does freeswitch support transcoding between broadvoice (BV32 ) and > G711 ? > > Did anyone try using freeswitch with Xten/counterpath Sip phone > configured with broadvoice32? Also, can someone recomend another > free sip phone supporting BV 32? > > Thanks > Gaurav > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/406f00d1/attachment-0002.html From brian at freeswitch.org Fri Nov 20 09:34:07 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 11:34:07 -0600 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <4B06D1E2.3060008@3gnt.net> References: <4B06CD46.6050408@3gnt.net> <4B06D1E2.3060008@3gnt.net> Message-ID: Well contribute your patches against SVN Trunk to http://jira.freeswitch.org /b On Nov 20, 2009, at 11:29 AM, Igor Neves wrote: > Hi, > > I have it working on CentOS, I can help packaging it, what does it > involves more precisely? > > > On 11/20/2009 05:20 PM, Itamar Reis Peixoto wrote: >> do you like to help packaging it for fedora, centos and rhel ? >> >> From msc at freeswitch.org Fri Nov 20 09:34:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Nov 2009 09:34:38 -0800 Subject: [Freeswitch-users] Broadvoice 32 transcoding support? In-Reply-To: <144359.82983.qm@web113920.mail.gq1.yahoo.com> References: <144359.82983.qm@web113920.mail.gq1.yahoo.com> Message-ID: <87f2f3b90911200934n20373bc6tf01677ec8d2bb11d@mail.gmail.com> On Fri, Nov 20, 2009 at 1:16 AM, Gaurav Singh wrote: > Hi, > > Does freeswitch support transcoding between broadvoice (BV32 ) and G711 ? > Try latest trunk. There was a new update just added very recently... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/39e52e58/attachment-0002.html From brian at freeswitch.org Fri Nov 20 09:34:47 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 11:34:47 -0600 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <4B06D266.7020109@3gnt.net> References: <4B06CD46.6050408@3gnt.net> <6b65470d0911200919y34546bvc93f3a5c976c1dc7@mail.gmail.com> <4B06D266.7020109@3gnt.net> Message-ID: <925C296B-2582-4A38-9E43-A4FBF9B9224E@freeswitch.org> Hope on IRC and talk to MikeJ in #freeswitch he can direct you better on what to do vs not do since he maintains the builds system in FreeSWITCH. /b On Nov 20, 2009, at 11:31 AM, Igor Neves wrote: > Ok, > > But how should I proceed? > > Thanks, From msc at freeswitch.org Fri Nov 20 09:40:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Nov 2009 09:40:46 -0800 Subject: [Freeswitch-users] APT Utility In-Reply-To: <7296F933-0972-47A7-B988-01557C0BDCC5@gmail.com> References: <005a01ca6898$f16d99d0$d448cd70$@com> <000301ca6987$c47edbb0$4d7c9310$@com> <0B55F774-9F77-4B4D-891D-7FD9595E644A@gmail.com> <005301ca69d0$ff7aea80$fe70bf80$@com> <7296F933-0972-47A7-B988-01557C0BDCC5@gmail.com> Message-ID: <87f2f3b90911200940m12f339dbkf511dc76dee35ec2@mail.gmail.com> On Fri, Nov 20, 2009 at 3:56 AM, Rob Forman wrote: > Hi David, > > Apt was just for getting your dependencies in order. Now you can go about > the business of compiling and installing Freeswitch. You might start with > 1.0.4 so you don't have to mess with svn yet. > > > http://wiki.freeswitch.org/wiki/Installation_Guide#FreeSWITCH_1.0.4_.22Phoenix.22_Release > > Just keep reading through the wiki or google for tutorials. Good luck! > Rob > > Actually, if you want to avoid SVN then get the latest 1.0.5 pre-release version. It will be more stable than 1.0.4. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/1492290a/attachment-0002.html From costa.zikalala at gmail.com Fri Nov 20 09:51:59 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Fri, 20 Nov 2009 19:51:59 +0200 Subject: [Freeswitch-users] phpmod fails to make Message-ID: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> I've been trying to make phpmod without any success. I've even tried to ./configure --with-php and it did't help. I've just upgraded to latest svn and am running FS on FC11. I keep getting this error message: make: php-config: Command not found g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:717:18: error: zend.h: No such file or directory esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory esl_wrap.cpp:719:17: error: php.h: No such file or directory esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or directory esl_wrap.cpp:980:17: error: esl.h: No such file or directory esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? declared void esl_wrap.cpp:793: error: ?zval? was not declared in this scope esl_wrap.cpp:793: error: ?z? was not declared in this scope esl_wrap.cpp:793: error: expected primary-expression before ?void? esl_wrap.cpp:793: error: expected primary-expression before ?*? token esl_wrap.cpp:793: error: ?type? was not declared in this scope esl_wrap.cpp:793: error: expected primary-expression before ?int? make: *** [esl_wrap.o] Error 1 Please help. Thanks Costa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/e992646d/attachment-0002.html From brian at freeswitch.org Fri Nov 20 09:58:36 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 11:58:36 -0600 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> Message-ID: I'm going to guess you did cd libs/esl/php then typed make.. move up one dir first then type make phpmod.. but you seem to be missing all the php dev headers. /b On Nov 20, 2009, at 11:51 AM, Costa Zikalala wrote: > I've been trying to make phpmod without any success. I've even tried > to ./configure --with-php and it did't help. > I've just upgraded to latest svn and am running FS on FC11. > > I keep getting this error message: > > make: php-config: Command not found > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o > esl_wrap.o > esl_wrap.cpp:717:18: error: zend.h: No such file or directory > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory > esl_wrap.cpp:719:17: error: php.h: No such file or directory > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or > directory > esl_wrap.cpp:980:17: error: esl.h: No such file or directory > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope > esl_wrap.cpp:788: error: ISO C++ forbids declaration of > ?ZEND_RSRC_DTOR_FUNC? with no type > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this > scope > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? > declared void > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > esl_wrap.cpp:793: error: ?z? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?void? > esl_wrap.cpp:793: error: expected primary-expression before ?*? token > esl_wrap.cpp:793: error: ?type? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?int? > make: *** [esl_wrap.o] Error 1 > > Please help. > Thanks > Costa From intralanman at freeswitch.org Fri Nov 20 10:05:03 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 20 Nov 2009 13:05:03 -0500 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> Message-ID: try installing php-devel -Ray On Nov 20, 2009, at 12:51 PM, Costa Zikalala wrote: > I've been trying to make phpmod without any success. I've even tried > to ./configure --with-php and it did't help. > I've just upgraded to latest svn and am running FS on FC11. > > I keep getting this error message: > > make: php-config: Command not found > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o > esl_wrap.o > esl_wrap.cpp:717:18: error: zend.h: No such file or directory > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory > esl_wrap.cpp:719:17: error: php.h: No such file or directory > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or > directory > esl_wrap.cpp:980:17: error: esl.h: No such file or directory > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope > esl_wrap.cpp:788: error: ISO C++ forbids declaration of > ?ZEND_RSRC_DTOR_FUNC? with no type > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this > scope > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? > declared void > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > esl_wrap.cpp:793: error: ?z? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?void? > esl_wrap.cpp:793: error: expected primary-expression before ?*? token > esl_wrap.cpp:793: error: ?type? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?int? > make: *** [esl_wrap.o] Error 1 > > Please help. > Thanks > Costa > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Fri Nov 20 10:06:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Nov 2009 10:06:43 -0800 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> Message-ID: <87f2f3b90911201006t15f69ef5pa4cd2a265a9f3c2b@mail.gmail.com> On Fri, Nov 20, 2009 at 9:51 AM, Costa Zikalala wrote: > I've been trying to make phpmod without any success. I've even tried to > ./configure --with-php and it did't help. > I've just upgraded to latest svn and am running FS on FC11. > > I keep getting this error message: > > make: php-config: Command not found > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o > esl_wrap.cpp:717:18: error: zend.h: No such file or directory > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory > esl_wrap.cpp:719:17: error: php.h: No such file or directory > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or directory > esl_wrap.cpp:980:17: error: esl.h: No such file or directory > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope > esl_wrap.cpp:788: error: ISO C++ forbids declaration of > ?ZEND_RSRC_DTOR_FUNC? with no type > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this scope > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? > declared void > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > esl_wrap.cpp:793: error: ?z? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?void? > esl_wrap.cpp:793: error: expected primary-expression before ?*? token > esl_wrap.cpp:793: error: ?type? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?int? > make: *** [esl_wrap.o] Error 1 > > Did you install the php-devel stuff? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/27494a57/attachment-0002.html From costa.zikalala at gmail.com Fri Nov 20 10:50:26 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Fri, 20 Nov 2009 20:50:26 +0200 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> Message-ID: <59daa2cd0911201050x18e60941le605d0a40da9cf48@mail.gmail.com> Thanks for quick responses guys, yes I was doing it under the php directory. I've now also installed php-devel, and am now getting this error: make[1]: Entering directory `/home/Costa/freeswitch-1.0.4/libs/esl/php' g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lcrypt -lcrypt -ledit -lncurses -lresolv -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt -lm -lcrypt -o ESL.so -L. /usr/bin/ld: cannot find -ledit collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make[1]: Leaving directory `/home/Costa/freeswitch-1.0.4/libs/esl/php' make: *** [phpmod] Error 2 2009/11/20 Brian West > I'm going to guess you did cd libs/esl/php then typed make.. move up > one dir first then type make phpmod.. but you seem to be missing all > the php dev headers. > > /b > > On Nov 20, 2009, at 11:51 AM, Costa Zikalala wrote: > > > I've been trying to make phpmod without any success. I've even tried > > to ./configure --with-php and it did't help. > > I've just upgraded to latest svn and am running FS on FC11. > > > > I keep getting this error message: > > > > make: php-config: Command not found > > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o > > esl_wrap.o > > esl_wrap.cpp:717:18: error: zend.h: No such file or directory > > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory > > esl_wrap.cpp:719:17: error: php.h: No such file or directory > > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory > > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or > > directory > > esl_wrap.cpp:980:17: error: esl.h: No such file or directory > > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory > > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope > > esl_wrap.cpp:788: error: ISO C++ forbids declaration of > > ?ZEND_RSRC_DTOR_FUNC? with no type > > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this > > scope > > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token > > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? > > declared void > > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > > esl_wrap.cpp:793: error: ?z? was not declared in this scope > > esl_wrap.cpp:793: error: expected primary-expression before ?void? > > esl_wrap.cpp:793: error: expected primary-expression before ?*? token > > esl_wrap.cpp:793: error: ?type? was not declared in this scope > > esl_wrap.cpp:793: error: expected primary-expression before ?int? > > make: *** [esl_wrap.o] Error 1 > > > > Please help. > > Thanks > > Costa > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/0affb30a/attachment-0002.html From siniypin at gmail.com Fri Nov 20 11:07:22 2009 From: siniypin at gmail.com (RobertT) Date: Fri, 20 Nov 2009 22:07:22 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> Message-ID: <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> No, I don't use Xlite. I use my own .Net wrapper around pjsip ua lib. Foreseeing uncertaincies about it's quality I may say that pjsua reference implementation yields the same results in this scenario. Actually I have no doubt that FS is working nicely with tcp and tls as well because I had it working till some moment. And I don't know what the hell happened. =( In order to check if it is something related to my config I switched it back to default and conducted the same test with (urghhh) no luck as well. So now I wonder what could cause this very-very strange behavior? Some issues with network? But why the UDP works then? All traces (FS SIP, FS console, SIP caller and callee's) are here: http://pastebin.com/m2008de4e Regards, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/6a84b66a/attachment-0002.html From intralanman at freeswitch.org Fri Nov 20 11:14:47 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 20 Nov 2009 14:14:47 -0500 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: <59daa2cd0911201050x18e60941le605d0a40da9cf48@mail.gmail.com> References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> <59daa2cd0911201050x18e60941le605d0a40da9cf48@mail.gmail.com> Message-ID: you need additional libs... editline-devel or something similar. alternatively, you can remove them in the Makefile -Ray On Nov 20, 2009, at 1:50 PM, Costa Zikalala wrote: > Thanks for quick responses guys, yes I was doing it under the php > directory. > I've now also installed php-devel, and am now getting this error: > > make[1]: Entering directory `/home/Costa/freeswitch-1.0.4/libs/esl/ > php' > g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lcrypt -lcrypt - > ledit -lncurses -lresolv -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt - > lm -lcrypt -o ESL.so -L. > /usr/bin/ld: cannot find -ledit > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > make[1]: Leaving directory `/home/Costa/freeswitch-1.0.4/libs/esl/php' > make: *** [phpmod] Error 2 > > > > > 2009/11/20 Brian West > I'm going to guess you did cd libs/esl/php then typed make.. move up > one dir first then type make phpmod.. but you seem to be missing all > the php dev headers. > > /b > > On Nov 20, 2009, at 11:51 AM, Costa Zikalala wrote: > > > I've been trying to make phpmod without any success. I've even tried > > to ./configure --with-php and it did't help. > > I've just upgraded to latest svn and am running FS on FC11. > > > > I keep getting this error message: > > > > make: php-config: Command not found > > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o > > esl_wrap.o > > esl_wrap.cpp:717:18: error: zend.h: No such file or directory > > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory > > esl_wrap.cpp:719:17: error: php.h: No such file or directory > > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory > > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or > > directory > > esl_wrap.cpp:980:17: error: esl.h: No such file or directory > > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory > > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope > > esl_wrap.cpp:788: error: ISO C++ forbids declaration of > > ?ZEND_RSRC_DTOR_FUNC? with no type > > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this > > scope > > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token > > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? > > declared void > > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > > esl_wrap.cpp:793: error: ?z? was not declared in this scope > > esl_wrap.cpp:793: error: expected primary-expression before ?void? > > esl_wrap.cpp:793: error: expected primary-expression before ?*? > token > > esl_wrap.cpp:793: error: ?type? was not declared in this scope > > esl_wrap.cpp:793: error: expected primary-expression before ?int? > > make: *** [esl_wrap.o] Error 1 > > > > Please help. > > Thanks > > Costa > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/e0683bde/attachment-0002.html From stevendt at primrosebank.net Fri Nov 20 12:14:51 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Fri, 20 Nov 2009 20:14:51 -0000 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls Message-ID: <31F6B654EEB247F493DEA83DDF816753@bp1.ad.bp.com> Hi, I have just purchased an ATA (Pluscom SIP VoIP ATA, model VPA-11) to try to use a normal (analogue) cordless phone with FreeSwitch. I have got the ATA setup and talking to FreeSwitch, it has registered the right extension and can receive and pick-up incoming calls. However, I can't dial numbers successfully, I can get the dial tone and dial the numbers, but the target numbers (internal or external) are not recognised by FreeSwitch. I have pasted a dump in the pastebin, but I can't see anything that tells me what the problem might be. Can anyone suggest what the problem might be please ? Regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/b384530c/attachment-0002.html From costa.zikalala at gmail.com Fri Nov 20 13:43:32 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Fri, 20 Nov 2009 23:43:32 +0200 Subject: [Freeswitch-users] phpmod fails to make In-Reply-To: References: <59daa2cd0911200951j3d24575qd4d91afcb11865e8@mail.gmail.com> <59daa2cd0911201050x18e60941le605d0a40da9cf48@mail.gmail.com> Message-ID: <59daa2cd0911201343s15cf1d9dlc843c30e51b03582@mail.gmail.com> Thanks Ray, I installed libedit-devel and it worked like charm. Costa 2009/11/20 Raymond Chandler > you need additional libs... editline-devel or something similar. > alternatively, you can remove them in the Makefile > > -Ray > > > On Nov 20, 2009, at 1:50 PM, Costa Zikalala wrote: > > Thanks for quick responses guys, yes I was doing it under the php > directory. > I've now also installed php-devel, and am now getting this error: > > make[1]: Entering directory `/home/Costa/freeswitch-1.0.4/libs/esl/php' > g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lcrypt -lcrypt -ledit > -lncurses -lresolv -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt -lm -lcrypt -o > ESL.so -L. > /usr/bin/ld: cannot find -ledit > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > make[1]: Leaving directory `/home/Costa/freeswitch-1.0.4/libs/esl/php' > make: *** [phpmod] Error 2 > > > > > 2009/11/20 Brian West > >> I'm going to guess you did cd libs/esl/php then typed make.. move up >> one dir first then type make phpmod.. but you seem to be missing all >> the php dev headers. >> >> /b >> >> On Nov 20, 2009, at 11:51 AM, Costa Zikalala wrote: >> >> > I've been trying to make phpmod without any success. I've even tried >> > to ./configure --with-php and it did't help. >> > I've just upgraded to latest svn and am running FS on FC11. >> > >> > I keep getting this error message: >> > >> > make: php-config: Command not found >> > g++ -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o >> > esl_wrap.o >> > esl_wrap.cpp:717:18: error: zend.h: No such file or directory >> > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >> > esl_wrap.cpp:719:17: error: php.h: No such file or directory >> > esl_wrap.cpp:972:21: error: php_ini.h: No such file or directory >> > esl_wrap.cpp:973:31: error: ext/standard/info.h: No such file or >> > directory >> > esl_wrap.cpp:980:17: error: esl.h: No such file or directory >> > esl_wrap.cpp:981:21: error: esl_oop.h: No such file or directory >> > esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope >> > esl_wrap.cpp:788: error: ISO C++ forbids declaration of >> > ?ZEND_RSRC_DTOR_FUNC? with no type >> > esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this >> > scope >> > esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token >> > esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? >> > declared void >> > esl_wrap.cpp:793: error: ?zval? was not declared in this scope >> > esl_wrap.cpp:793: error: ?z? was not declared in this scope >> > esl_wrap.cpp:793: error: expected primary-expression before ?void? >> > esl_wrap.cpp:793: error: expected primary-expression before ?*? token >> > esl_wrap.cpp:793: error: ?type? was not declared in this scope >> > esl_wrap.cpp:793: error: expected primary-expression before ?int? >> > make: *** [esl_wrap.o] Error 1 >> > >> > Please help. >> > Thanks >> > Costa >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/9f5cbef2/attachment-0002.html From Russell.Mosemann at cune.org Fri Nov 20 14:38:21 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Fri, 20 Nov 2009 22:38:21 -0000 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls In-Reply-To: <31F6B654EEB247F493DEA83DDF816753@bp1.ad.bp.com> Message-ID: <20091120223822.23FEF3E941A@mail.cune.org> Dave Stevenson said: > I have pasted a dump in the pastebin, URL? -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From stevendt at primrosebank.net Fri Nov 20 14:46:03 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Fri, 20 Nov 2009 22:46:03 -0000 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls References: <20091120223822.23FEF3E941A@mail.cune.org> Message-ID: <02EE75E8713B4A1BBA3BBEB03B91AF7D@bp1.ad.bp.com> Ooops - sorry about that ! OK, here you go .... http://pastebin.freeswitch.org/11205 I'd really appreciate some help with this as I'm really struggling. I think that the right tones are being sent as I can transfer a Voice Mail call to the phone and activate the voice prompts correctly. I have entered a blank dialplan in the ATA which should let all numbers be processed. FreeSwitch just won't play ball though ! Regards Dave ----- Original Message ----- From: To: Sent: Friday, November 20, 2009 10:38 PM Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out,but can receive calls > Dave Stevenson said: >> I have pasted a dump in the pastebin, > > URL? > > -- > Russell Mosemann > > > > ________________________________________________________ > Concordia University, Nebraska > See http://www.cune.edu/ for the latest news and events! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From malay.thakershi at continuityhealth.com Fri Nov 20 15:18:44 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Fri, 20 Nov 2009 17:18:44 -0600 Subject: [Freeswitch-users] mod_flite sound profiles Message-ID: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> Hello all, I am not able to play any female sound in mod_flite. I did try setting all 4 voice types one by one but it only says male voice. And the voice quality is not good at all. Am I doing something wrong? mObjMainSession.Answer(); mObjMainSession.sleep(1000, 0); //set tts engine mObjMainSession.SetTtsParameters("flite", "awb"); //Introduction message mObjMainSession.Speak("Hello... Welcome to phone system for assessment"); mObjMainSession.sleep(1000, 0); Also, can someone tell me what is the best way to get TTS going with good quality? Thank you for help. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/c414cb22/attachment-0002.html From james at talent.com.au Fri Nov 20 15:19:59 2009 From: james at talent.com.au (James Budge) Date: Sat, 21 Nov 2009 09:19:59 +1000 Subject: [Freeswitch-users] OS X compile error Message-ID: make[6]: *** [mod_amr.so] Error 1 make[5]: *** [all] Error 1 make[4]: *** [mod_amr-all] Error 1 make[3]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[2]: *** [all-recursive] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 OS X 10.6.2 Xcode 3.2.1 FS Updated to revision 15582 From brian at freeswitch.org Fri Nov 20 15:32:27 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 17:32:27 -0600 Subject: [Freeswitch-users] OS X compile error In-Reply-To: References: Message-ID: You have left out the whole bits that say what exactly failed... look UP farther. btw don't make -j http://wiki.freeswitch.org/wiki/Installation_Guide#64-bit_Mac_OS_X_.28Snow_Leopard.29 /b On Nov 20, 2009, at 5:19 PM, James Budge wrote: > make[6]: *** [mod_amr.so] Error 1 > make[5]: *** [all] Error 1 > make[4]: *** [mod_amr-all] Error 1 > make[3]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + make install + > +----------------------------------------------+ > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all] Error 2 > make: *** [current] Error 2 > > > OS X 10.6.2 > Xcode 3.2.1 > FS Updated to revision 15582 From brian at freeswitch.org Fri Nov 20 15:33:00 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 17:33:00 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> Message-ID: <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> You pay top dollar for it. The free stuff just isn't as good as what you PAY good money for. I don't expect that to change anytime soon. /b On Nov 20, 2009, at 5:18 PM, Malay Thakershi wrote: > Also, can someone tell me what is the best way to get TTS going with > good quality? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/5aba3035/attachment-0002.html From jason at jasonjgw.net Fri Nov 20 16:02:02 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 21 Nov 2009 11:02:02 +1100 Subject: [Freeswitch-users] RTP issues (possibly nat-related) In-Reply-To: References: <20091120011542.GA20754@jdc.jasonjgw.net> <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> <20091120070137.GA28316@jdc.jasonjgw.net> Message-ID: <20091121000202.GA21139@jdc.jasonjgw.net> I can still reproduce this as of rev. 15584. Symptom: 1. I called a test number via my ISP (IPv4, subject to nat). This worked. 2. I placed a second call to the same number 30 seconds later - connected, but no audio received. From mike at jerris.com Fri Nov 20 16:23:02 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 20 Nov 2009 19:23:02 -0500 Subject: [Freeswitch-users] freeswitch.spec patch In-Reply-To: <925C296B-2582-4A38-9E43-A4FBF9B9224E@freeswitch.org> References: <4B06CD46.6050408@3gnt.net> <6b65470d0911200919y34546bvc93f3a5c976c1dc7@mail.gmail.com> <4B06D266.7020109@3gnt.net> <925C296B-2582-4A38-9E43-A4FBF9B9224E@freeswitch.org> Message-ID: <5D94EAF2-B446-4E0B-99D0-C7C4FC39456C@jerris.com> This was merged into trunk. On Nov 20, 2009, at 12:34 PM, Brian West wrote: > Hope on IRC and talk to MikeJ in #freeswitch he can direct you better > on what to do vs not do since he maintains the builds system in > FreeSWITCH. > > /b > > On Nov 20, 2009, at 11:31 AM, Igor Neves wrote: > >> Ok, >> >> But how should I proceed? >> >> Thanks, > From mike at jerris.com Fri Nov 20 16:28:57 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 20 Nov 2009 19:28:57 -0500 Subject: [Freeswitch-users] change event value In-Reply-To: <4B04E766.8070706@savion.huji.ac.il> References: <4B04E766.8070706@savion.huji.ac.il> Message-ID: no. On Nov 19, 2009, at 1:36 AM, Eli Hayun wrote: > Hi > Is there is a way to intercept an event (for example : REGISTER) and > change one of its parameters (for example: the extension number) and > fire up the corrected event? > > I need it to set the speedial of the phone value to be "**xxxxx" but to > make it register as "xxxxx" From brian at freeswitch.org Fri Nov 20 16:48:53 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 18:48:53 -0600 Subject: [Freeswitch-users] RTP issues (possibly nat-related) In-Reply-To: <20091121000202.GA21139@jdc.jasonjgw.net> References: <20091120011542.GA20754@jdc.jasonjgw.net> <8F7A6DAF-8C92-462F-9C75-0BCE1A58A2E5@freeswitch.org> <20091120070137.GA28316@jdc.jasonjgw.net> <20091121000202.GA21139@jdc.jasonjgw.net> Message-ID: <7163EBBF-F043-4C63-88BC-0A1F47F5906E@freeswitch.org> Can you give me some console logs and sip traces... maybe an rtp pcap? Thanks, Brian On Nov 20, 2009, at 6:02 PM, Jason White wrote: > I can still reproduce this as of rev. 15584. > > Symptom: > > 1. I called a test number via my ISP (IPv4, subject to nat). This > worked. > > 2. I placed a second call to the same number 30 seconds later - > connected, but > no audio received. From Prometheus001 at gmx.net Fri Nov 20 16:56:45 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 21 Nov 2009 01:56:45 +0100 Subject: [Freeswitch-users] Problems with Voicemail Message-ID: <4B073ACD.1090708@gmx.net> Hello, i have a couple of problems with voicemail. Voicemails are recorded but not played in any way. 1) when I call my voicemail, I can hear the number of new messages, but I canot not hear the recorded files itself. I hear the following * "You have 1 urgent new message in forder inbox" * "You have 7 new messages in forder inbox" * "New message number 1 Jan 011970 at 1 am" * (message is NOT played) * "You have 1 urgent new message in forder inbox" * "You have 7 new messages in forder inbox" * "press 1 to listen, press 2...." * (I press 1) * "New message number 1 Jan 011970 at 1 am" * (message is NOT played) * "You have 1 urgent new message in forder inbox" * "You have 7 new messages in forder inbox" * ... The voicemail files are stored in the file system as wav files and I can play them manually from the file system - so there is sound inside. 2) Another strange thing is that all recorded calls are announced with a date of 01.Jan.1970 although the databse shows correct values. 3) Alternatively playing it on the web Gui on http://fs.ip:8080/api/voicemail/web doesn't work either. Date is again 01.Jan.1970 and shown length of the file is always 00:00:00, although the database shows the correct number of seconds 4) Just to note that whenever I expect a recorded file to be played I see the following on the console 2009-11-21 00:17:02.511110 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA In my installation Freeswitch is running in a cluster and voicemails are stored in a mysql database. read_epoch is always 0, so file seems that Freeswitch never reads and updates an entry. Grepping mysql however shows a number of queries against the database and also the filenames are correctly read (output of ngrep): select * from voicemail_msgs where username='200' and domain='sip11.mydomain.com' and read_epoch=0 order by read_flags, created_epoch 1258748304.0.200.sip11.mydomain.com$db2801c4-d611-11de-8c58-554df1d6d322.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_b0fbf9e6-d611-11de-8c58-554df1d6d322.wav.15..A_URGENT 1258746833.0.200.sip11.mydomain.com$6e486a2e-d60e-11de-bb97-eb22f15930a0.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_50e727c2-d60e-11de-bb97-eb22f15930a0.wav.7..B_NORMAL 1258748679.0.200.sip11.mydomain.com$bac4dd0c-d612-11de-9618-afbc82bc409a.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_9e7865c4-d612-11de-9618-afbc82bc409a.wav.13..B_NORMAL 1258749095.0.200.sip11.mydomain.com$b2376082-d613-11de-80e8-89d0ee29138d.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_a6caaefc-d613-11de-80e8-89d0ee29138d.wav.6..B_NORMAL 1258749417.0.200.sip11.mydomain.com$726b375c-d614-11de-bb4c-6d51cf20cc23.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_6777907a-d614-11de-bb4c-6d51cf20cc23.wav.5..B_NORMAL 1258750260.0.200.sip11.mydomain.com$68cecedc-d616-11de-b8c8-69b0064d633e.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_5b6afb6c-d616-11de-b8c8-69b0064d633e.wav.9..B_NORMAL 1258753767.0.200.sip11.mydomain.com$93657422-d61e-11de-b8c8-69b0064d633e.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_84c1c588-d61e-11de-b8c8-69b0064d633e.wav.10..B_NORMAL Here's the debug log: EXECUTE sofia/internal/200 at sip1.mydomain.com send_display(VM 200) 2009-11-20 23:16:36.392353 [DEBUG] mod_dptools.c:703 sofia/internal/200 at sip1.mydomain.com receive message [DISPLAY] EXECUTE sofia/internal/200 at sip1.mydomain.com voicemail(check default sip11.mydomain.com 200) 2009-11-20 23:16:36.392353 [DEBUG] mod_voicemail.c:799 [default] rwlock 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-hello.wav] (en:en) 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:36.392353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:37.612349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:37.712349 [DEBUG] switch_channel.c:182 sofia/internal/200 at sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:37.812353 [DEBUG] switch_channel.c:182 sofia/internal/200 at sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:37.942376 [DEBUG] switch_channel.c:182 sofia/internal/200 at sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:38.062368 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:38.612348 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:38.712354 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1] (en:en) 2009-11-20 23:16:38.712354 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:38.712354 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:39.172350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:39.272353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-urgent-new.wav] (en:en) 2009-11-20 23:16:39.272353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:39.272353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:40.052349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:40.152353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-message.wav] (en:en) 2009-11-20 23:16:40.152353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:40.152353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:40.732349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:40.832349 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:16:40.832349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:40.832349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:42.012350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:42.112350 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:42.112350 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:42.112350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:42.112350 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:42.672354 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:42.772362 [DEBUG] switch_ivr_play_say.c:273 Handle say:[7] (en:en) 2009-11-20 23:16:42.772362 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:42.772362 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:43.312349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:43.412353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-new.wav] (en:en) 2009-11-20 23:16:43.412353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:43.412353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:43.752353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:43.872362 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-messages.wav] (en:en) 2009-11-20 23:16:43.872362 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:43.872362 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:44.532350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:44.652368 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:16:44.652368 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:44.652368 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:45.832349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:45.932353 [DEBUG] switch_channel.c:182 sofia/internal/200 at sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:46.062365 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:46.062365 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-new.wav] (en:en) 2009-11-20 23:16:46.062365 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:46.062365 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:46.392349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:46.492349 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-message_number.wav] (en:en) 2009-11-20 23:16:46.492349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:46.492349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:47.312349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:47.412349 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1] (en:en) 2009-11-20 23:16:47.412349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:47.412349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:47.872372 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:47.992349 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:47.992349 [DEBUG] switch_ivr_play_say.c:273 Handle say:[25] (en:en) 2009-11-20 23:16:47.992349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:47.992349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:48.632353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:48.632353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:48.632353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:49.152353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:49.152353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:49.152353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:49.612349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:49.612349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:49.612349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:50.092349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:50.092349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:50.092349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:50.532349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:50.532349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:50.532349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:51.012350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:51.012350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:51.012350 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:51.572349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:51.572349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:51.572349 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:51.852353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:51.852353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:51.852353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:52.312350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:52.312350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:52.312350 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:52.892356 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:52.892356 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:52.892356 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:53.472353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA 2009-11-20 23:16:53.602359 [DEBUG] switch_channel.c:182 sofia/internal/200 at sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:53.722367 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:53.722367 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:53.722367 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:53.722367 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:54.272349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:54.372353 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1] (en:en) 2009-11-20 23:16:54.372353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:54.372353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:54.832441 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:54.943764 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-urgent-new.wav] (en:en) 2009-11-20 23:16:54.943764 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:54.943764 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:55.732350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:55.847095 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-message.wav] (en:en) 2009-11-20 23:16:55.847095 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:55.847095 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:56.432351 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:56.532354 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:16:56.532354 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:56.532354 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:57.712350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:57.812353 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:57.812353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:57.812353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:57.812353 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:58.372350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:58.472356 [DEBUG] switch_ivr_play_say.c:273 Handle say:[7] (en:en) 2009-11-20 23:16:58.472356 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:58.472356 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:59.012350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:59.112350 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-new.wav] (en:en) 2009-11-20 23:16:59.112350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:59.112350 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:59.452351 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:59.572350 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-messages.wav] (en:en) 2009-11-20 23:16:59.572350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:16:59.572350 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:17:00.232350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:17:00.332371 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:17:00.332371 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-20 23:17:00.332371 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/f850a9b4/attachment-0002.html From brian at freeswitch.org Fri Nov 20 17:08:06 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 19:08:06 -0600 Subject: [Freeswitch-users] Problems with Voicemail In-Reply-To: <4B073ACD.1090708@gmx.net> References: <4B073ACD.1090708@gmx.net> Message-ID: <976A0342-4F4B-4035-9201-D56F8625AE12@freeswitch.org> I'm going to venture to guess maybe the file was recorded in a different codec and NOT pcma? /b On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: > 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / > usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA From brian at freeswitch.org Fri Nov 20 17:06:01 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Nov 2009 19:06:01 -0600 Subject: [Freeswitch-users] Problems with Voicemail In-Reply-To: <4B073ACD.1090708@gmx.net> References: <4B073ACD.1090708@gmx.net> Message-ID: This should give you some sort of clue. /b On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: > 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / > usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA From mike at jerris.com Fri Nov 20 17:17:23 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 20 Nov 2009 20:17:23 -0500 Subject: [Freeswitch-users] Media got stuck after attended transfer... In-Reply-To: References: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com> <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com> <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> Message-ID: I think a better approach here is to use spandsp. We already have some groundwork done for this. If you are interested in contributing, please email consulting at freeswitch.org and we can discuss further. Mike On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: > Hi, > > one of my customers is willing to contribute for t38 integration. > > The basic idea is to connect HylaFAX to FS: > t38modem <-> FreeSWITCH <-> Media Gateway with t38 support > All this without media proxy. > > Another idea might be to implement t38 origination/termination with a class 1 modem input/output for use with HylaFAX. > > Do you know how much money we need to collect for t38 support? > How much time is needed for implementing this? > > Thanks, Klaus > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: Friday, October 16, 2009 2:10 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Media got stuck after attended transfer... > > > > On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga wrote: > hi, any clue when can t38 be added? > > > "Eventually." :) Of course, if we could get more to add to the bounty it might grease the wheels of innovation. > > http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch > > Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) > > -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/9b0c7e8c/attachment-0002.html From stevendt at primrosebank.net Fri Nov 20 17:20:47 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Sat, 21 Nov 2009 01:20:47 -0000 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Issue Closed References: <20091120223822.23FEF3E941A@mail.cune.org> <02EE75E8713B4A1BBA3BBEB03B91AF7D@bp1.ad.bp.com> Message-ID: <7449BC3B2C05406DAE264F5A11BAF110@bp1.ad.bp.com> OK, Just to close this one out, I've just spent some time on IRC and Michael Collins very quickly helped me get this sorted. I am using a cheap ATA which has a couple of "issues". I should probably have included it in the original post, but the ATA is a :- Pluscom SIP VoIP ATA - Model VPA-11. Quote from the ATA Manual ....... "If a default dial plan string is not required, the Default Dial Plan String field on the General configuration page (Section 4.1.2) can be left empty, in which case the default dial pattern to accept all dialed digits will be incorporated. The default dial pattern, [0-9*]>#[0-9*].e[0-9*].ft4, is transparent to the user and will not be displayed on the General configuration page" I had already (or so I thought) deleted the default ATA dialplan as I suspected that it was causing problems. As it turns out though, even with an apparently blank dialplan configured, the ATA was inserting some characters after the dialed number which were confusing the number handling in FreeSwitch. Michael quickly spotted the problem from my Pastebin dump and took about 10 seconds to come up with a fix ! Adding the following section to the top of the default FreeSwitch dialplan strips these extra characters off the string that FreeSwitch sees To: Sent: Friday, November 20, 2009 10:46 PM Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out,but can receive calls > Ooops - sorry about that ! > > OK, here you go .... > > http://pastebin.freeswitch.org/11205 > > I'd really appreciate some help with this as I'm really struggling. > > I think that the right tones are being sent as I can transfer a Voice Mail > call to the phone and activate the voice prompts correctly. > > I have entered a blank dialplan in the ATA which should let all numbers be > processed. > > FreeSwitch just won't play ball though ! > > Regards > Dave > > > > ----- Original Message ----- > From: > To: > Sent: Friday, November 20, 2009 10:38 PM > Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial > Out,but can receive calls > > >> Dave Stevenson said: >>> I have pasted a dump in the pastebin, >> >> URL? >> >> -- >> Russell Mosemann >> >> >> >> ________________________________________________________ >> Concordia University, Nebraska >> See http://www.cune.edu/ for the latest news and events! >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Fri Nov 20 19:34:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Nov 2009 21:34:05 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> Message-ID: <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> results cant possibly be the same there is not even any broadcast involved in uuid_transfer ? you need to attach a console trace with debug log up On Fri, Nov 20, 2009 at 10:41 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Finally got a chance to test this, the results are the same. > > Why am I getting this? Is it because I'm executing ring_ready before > I attempt the bridge? Is it because I'm using a socket? > > On Wed, Nov 11, 2009 at 10:14 PM, Anthony Minessale > wrote: > > dont execute bridge that way, your bridge itself is the other thing > already > > broadcasting. > > > > > > api uuid_transfer bridge:sofia/myprofile/foo at bar.cominline > > > > if you want to do more after the bridge > > set the variable park_after_bridge=true to make it go back to idle > > > > > > On Wed, Nov 11, 2009 at 4:33 PM, Kristian Kielhofner > > wrote: > >> > >> Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64. > >> > >> On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner > >> wrote: > >> > From the trace: > >> > > >> ..snip.. > >> > >> -- > >> Kristian Kielhofner > >> http://www.astlinux.org > >> http://blog.krisk.org > >> http://www.star2star.com > >> http://www.submityoursip.com > >> http://www.voalte.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091120/3adcc0e9/attachment-0002.html From thangappan143 at gmail.com Sat Nov 21 01:22:27 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Sat, 21 Nov 2009 14:52:27 +0530 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback Message-ID: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have implemented using Perl language. I did the following steps: => Set the playback_delimiter variable => Set the playback_sleep_val variable => Set the event lock as true => Set the freeswitch ( my own) variable as zero => Wait in the loop until the variable is been set as zero => Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) => Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played successfully). => Wait in the loop until the variable is been set as one. => Set the Event lock as false => Trying to get the DTMF digits ( Have a assurance that all the voice files are played). The problem is, The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* *NOTE*: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. Please help me................? Thanks in advance. -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/4c9e3b93/attachment-0002.html From shiyanov at gmail.com Sat Nov 21 01:58:29 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Sat, 21 Nov 2009 12:58:29 +0300 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> References: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> Message-ID: Anthony, >>As soon as you call uuid_bridge you are transferring both legs of the call to bridge to each other. >>This means your java app must exit so the channels can connect to each other. I didn't know that. Now my java app is exiting upon the onHangup() call so everything has become "ok". Thank you much. I'll add note to the wiki about this issue. Artem On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Your "annoying behaviour" is the exact behavior you should be getting > considering what you told FS to do. > > As soon as you call uuid_bridge you are transferring both legs of the call > to bridge to each other. > This means your java app must exit so the channels can connect to each > other. > > remember that you hangup hook can be called when the channel is transferred > not only when it hangs up. > you have to test which is happening based on the input to your callback. > > > On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: > >> Hi there! >> >> I've got annoying FS behavior: >> There are 2 channels executing the same Java application (application >> itself is an IVR). If I try to bridge them with uuid_bridged then both >> channels are killed. Here is a log from FS console: >> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 >> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> >> CS_HIBERNATE >> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing >> file >> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing >> file >> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal >> sofia/internal/1005 at 192.168.147.130 [BREAK] >> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 >> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change >> CS_EXECUTE -> CS_HIBERNATE >> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: >> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >> >> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] >> switch_core_session.c:933 Send signal >> sofia/internal/1001 at master.agent.starpoundtec >> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal >> sofia/internal/1005 at 192.168.147.130 [BREAK] >> >> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream >> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal >> sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] >> >> (FS version is 1.0.4) >> >> Any thoughts? >> >> >> Artem >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/78ead067/attachment-0002.html From mike at yes.net.ua Sat Nov 21 03:41:06 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Sat, 21 Nov 2009 13:41:06 +0200 Subject: [Freeswitch-users] Using odbc in FS core Message-ID: <1382216794.20091121134106@yes.net.ua> Hello Folks, I'm interesting in completely moving away from sqlite and use postgresql everywhere including core ( switch_core.c ) All other applications can use odbc without issues (sofia, limit, fifo etc), but as I see in core only sqlite3 supported. I correctly set 'core-db-dsn' parameter, but looks like the problem that latest psqlodbc_08_04_0100 don't support multiple statements in one request that is often used in switch_core_sqldb.c: > sql = switch_mprintf( > "update channels set uuid='%q' where uuid='%q' and hostname='%q';" > "update calls set caller_uuid='%q' where caller_uuid='%q' and hostname='%q';" > "update calls set callee_uuid='%q' where callee_uuid='%q' and hostname='%q'", > switch_event_get_header_nil(event, "unique-id"), > ... SKIP ... So, does anyone have any clue how to us postgresql in the FS core? Thanks. -- Mike Tkachuk From mike at yes.net.ua Sat Nov 21 04:02:07 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Sat, 21 Nov 2009 14:02:07 +0200 Subject: [Freeswitch-users] Using odbc in FS core In-Reply-To: <1382216794.20091121134106@yes.net.ua> References: <1382216794.20091121134106@yes.net.ua> Message-ID: <1013085378.20091121140207@yes.net.ua> Hello, Looks like the issue is not in multi statements in one request. Manually creating DB schema helped and everything started up. I will continue testing Also in code I see such construction: > switch_cache_db_execute_sql(dbh, "begin;delete from channels where hostname='';delete from channels where hostname='';commit;", &err); Anyone can explain why to do such delete twice and in transaction? Thanks. Saturday, November 21, 2009 1:41:06 PM, you wrote: MT> Hello Folks, MT> I'm interesting in completely moving away from sqlite and use MT> postgresql everywhere including core ( switch_core.c ) MT> All other applications can use odbc without issues (sofia, limit, MT> fifo etc), but as I see in core only sqlite3 supported. MT> I correctly set 'core-db-dsn' parameter, but looks like the problem MT> that latest psqlodbc_08_04_0100 don't support multiple statements in MT> one request that is often used in switch_core_sqldb.c: >> sql = switch_mprintf( >> "update channels set uuid='%q' where uuid='%q' and hostname='%q';" >> "update calls set caller_uuid='%q' where caller_uuid='%q' and hostname='%q';" >> "update calls set callee_uuid='%q' where callee_uuid='%q' and hostname='%q'", >> switch_event_get_header_nil(event, "unique-id"), >> ... SKIP ... MT> So, does anyone have any clue how to us postgresql in the FS core? MT> Thanks. MT> -- MT> Mike Tkachuk -- Mike Tkachuk From stevendt at primrosebank.net Sat Nov 21 04:10:23 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Sat, 21 Nov 2009 12:10:23 -0000 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Updated References: <20091120223822.23FEF3E941A@mail.cune.org><02EE75E8713B4A1BBA3BBEB03B91AF7D@bp1.ad.bp.com> <7449BC3B2C05406DAE264F5A11BAF110@bp1.ad.bp.com> Message-ID: <0E0B2CB55DD7426C91286784F4BF7353@bp1.ad.bp.com> Sorry for the extended forum thread on this subject - This really IS the last post ! I have now got the ATA to work without the dialplan fix provided by Michael. After I'd implemented the "fix", I had more of an idea of what the problem was and was better able to go through the Polycom VPA-11 setup screens through its web interface to see if there were any options that might have had a bearing on the problem. Under Configuration VoIP Non-Line Config General Parameters VoIP General There is an option to "Append UserId" - the Default Value is Yes. That was where the "extra characters" were coming from. Setting this option to No, makes the ATA behave more as expected, Regards Dave ----- Original Message ----- From: "Dave Stevenson" To: Sent: Saturday, November 21, 2009 1:20 AM Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out,but can receive calls - Issue Closed > OK, > > Just to close this one out, I've just spent some time on IRC and Michael > Collins very quickly helped me get this sorted. > > I am using a cheap ATA which has a couple of "issues". > > I should probably have included it in the original post, but the ATA is a > :- > > Pluscom SIP VoIP ATA - Model VPA-11. > > Quote from the ATA Manual ....... > > "If a default dial plan string is not required, the Default Dial Plan > String > field on the General configuration page (Section 4.1.2) > can be left empty, in which case the default dial pattern to accept all > dialed digits will be incorporated. > The default dial pattern, [0-9*]>#[0-9*].e[0-9*].ft4, is transparent to > the > user and will not be displayed on the General > configuration page" > > I had already (or so I thought) deleted the default ATA dialplan as I > suspected that it was causing problems. > > As it turns out though, even with an apparently blank dialplan configured, > the ATA was inserting some characters after the dialed number which were > confusing the number handling in FreeSwitch. > > Michael quickly spotted the problem from my Pastebin dump and took about > 10 > seconds to come up with a fix ! > > Adding the following section to the top of the default FreeSwitch dialplan > strips these extra characters off the string that FreeSwitch sees > > > > > > > > > Thanks a lot Michael ! > > regards > Dave > From Prometheus001 at gmx.net Sat Nov 21 04:14:17 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 21 Nov 2009 13:14:17 +0100 Subject: [Freeswitch-users] Problems with Voicemail In-Reply-To: <976A0342-4F4B-4035-9201-D56F8625AE12@freeswitch.org> References: <4B073ACD.1090708@gmx.net> <976A0342-4F4B-4035-9201-D56F8625AE12@freeswitch.org> Message-ID: <4B07D999.4040004@gmx.net> I installed all sounds from SVN, but usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA isn't there. I checked another, older installation and couldn't this file either. I think that freeswitch tries to build a sound path for the file to be played, and some parts of the path are missing. I expect it would play a recorded message at that time in /usr/local/freeswitch/storage/voicemail/default/${domain} and the defined format is "wav" not pcma. I also set "storage_dir" explicitely in the voicemail configs,but this also didn't help. Best regards Peter Brian West schrieb: > I'm going to venture to guess maybe the file was recorded in a > different codec and NOT pcma? > > /b > > On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: > > >> 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / >> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From siniypin at gmail.com Sat Nov 21 05:28:17 2009 From: siniypin at gmail.com (RobertT) Date: Sat, 21 Nov 2009 16:28:17 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> Message-ID: <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> Attached is graphical representation of SIP message flow. You can see that for some reason FS doesn't resend to callee an ACK message recieved from caller. Regards, RobertT -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/ab8948d5/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: TCP FS SIP msgs.PNG Type: image/png Size: 14515 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/ab8948d5/attachment-0002.png From brian at freeswitch.org Sat Nov 21 07:46:08 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 21 Nov 2009 09:46:08 -0600 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> Message-ID: <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> Well since we aren't a proxy we wouldn't resend the one we receive... what svn rev and are you using proxy media? /b On Nov 21, 2009, at 7:28 AM, RobertT wrote: > Attached is graphical representation of SIP message flow. You can > see that for some reason FS doesn't resend to callee an ACK message > recieved from caller. > > Regards, RobertT From anthony.minessale at gmail.com Sat Nov 21 08:14:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 21 Nov 2009 10:14:59 -0600 Subject: [Freeswitch-users] Using odbc in FS core In-Reply-To: <1013085378.20091121140207@yes.net.ua> References: <1382216794.20091121134106@yes.net.ua> <1013085378.20091121140207@yes.net.ua> Message-ID: <191c3a030911210814l6e50b883uba61815fcd36afe1@mail.gmail.com> we had the code slightly out of order, you should update to latest trunk for the right version. The test of 2 deletes is to see if your odbc driver will fail when trying to execute 2 statements at once so I can properly fail over to sqlite because transactions are mandatory for a database running core in odbc. On Sat, Nov 21, 2009 at 6:02 AM, Mike Tkachuk wrote: > Hello, > > Looks like the issue is not in multi statements in one request. > Manually creating DB schema helped and everything started up. > I will continue testing > > Also in code I see such construction: > > switch_cache_db_execute_sql(dbh, "begin;delete from channels where > hostname='';delete from channels where hostname='';commit;", &err); > Anyone can explain why to do such delete twice and in transaction? > > Thanks. > > > > Saturday, November 21, 2009 1:41:06 PM, you wrote: > > MT> Hello Folks, > > MT> I'm interesting in completely moving away from sqlite and use > MT> postgresql everywhere including core ( switch_core.c ) > > MT> All other applications can use odbc without issues (sofia, limit, > MT> fifo etc), but as I see in core only sqlite3 supported. > > MT> I correctly set 'core-db-dsn' parameter, but looks like the problem > MT> that latest psqlodbc_08_04_0100 don't support multiple statements in > MT> one request that is often used in switch_core_sqldb.c: > > >> sql = switch_mprintf( > >> "update channels set uuid='%q' where uuid='%q' and hostname='%q';" > >> "update calls set caller_uuid='%q' where caller_uuid='%q' and > hostname='%q';" > >> "update calls set callee_uuid='%q' where callee_uuid='%q' and > hostname='%q'", > >> switch_event_get_header_nil(event, "unique-id"), > >> ... SKIP ... > > MT> So, does anyone have any clue how to us postgresql in the FS core? > > MT> Thanks. > > MT> -- > MT> Mike Tkachuk > > > > -- > Mike Tkachuk > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/f31802d3/attachment-0002.html From anthony.minessale at gmail.com Sat Nov 21 08:34:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 21 Nov 2009 10:34:21 -0600 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> Message-ID: <191c3a030911210834o6c134ec0v8a57df04b946f8cf@mail.gmail.com> cant you use the execute_complete events to tell when your playback is done or var is set? On Sat, Nov 21, 2009 at 3:22 AM, Thangappan.M wrote: > Dear all, > > I am in the process of implementing IVR using event outbound > socket (async mode). > I have implemented using Perl language. > > I did the following steps: > => Set the playback_delimiter variable > => Set the playback_sleep_val variable > => Set the event lock as true > => Set the freeswitch ( my own) variable as zero > => Wait in the loop until the variable is been set as > zero > => Playback the voice files ( Here I combined the voice > files with the delimiter value if more than one voice files are there) > => Set the freeswitch(my own) variable as true ( This is > used to identify whether the voice files are played > successfully). > => Wait in the loop until the variable is been set as > one. > => Set the Event lock as false > > => Trying to get the DTMF digits ( Have a assurance > that all the voice files are played). > > The problem is, > > The above steps are working fine when the voice file count is > lesser than or equal to 10. After the voice files are played only the > variable(my own freeswitch) is set. Based on the variable I am doing further > things. > > But when I tried to give the voice files count of more than 10 > the variable has been set while starting to play back the first voice file > itself . Because of this I am not able to proceed further. > > *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* > > *NOTE*: I also referred mod_file_string documentation. In that they > specified 128 files can be used to play back the voice files using > playback_delimiter option. > > Please help me................? > Thanks in advance. > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/3d7dff3c/attachment-0002.html From abeka at greatiam.com Sat Nov 21 14:15:37 2009 From: abeka at greatiam.com (Sam Abekah-Mensah) Date: Sat, 21 Nov 2009 22:15:37 +0000 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway Message-ID: <4B086689.6080804@greatiam.com> I need help as I cannot receive calls through VOIPUSER. This is a learning setup Attached are my conf files. What is wrong with them ? When I dial from a landline I get a continuous beep. Attached are my gateway and the conf file to transfer. Sopfia Status is my screen message. I can see a FAIL and cannot make head or tail of all that message. Hopefully anyone using voipuser or in fact any of you clever folks can make sense of this. Thanks for your time. -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: sofia status.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/06405ed7/attachment-0002.txt -------------- next part -------------- A non-text attachment was scrubbed... Name: voipuser.xml Type: text/xml Size: 300 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/06405ed7/attachment-0004.xml -------------- next part -------------- A non-text attachment was scrubbed... Name: voipuser_org.xml Type: text/xml Size: 271 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/06405ed7/attachment-0005.xml From siniypin at gmail.com Sat Nov 21 15:23:05 2009 From: siniypin at gmail.com (RobertT) Date: Sun, 22 Nov 2009 02:23:05 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> Message-ID: <2160023e0911211523k7998d048nced3af8fb805e770@mail.gmail.com> Yep, I use proxy media. First it started with 1.0.4 release, then I've updated a week or two ago with the latest svn trunk, not sure what was the rev number. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/124d5003/attachment-0002.html From msc at freeswitch.org Sat Nov 21 16:51:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Sat, 21 Nov 2009 16:51:54 -0800 Subject: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Updated In-Reply-To: <0E0B2CB55DD7426C91286784F4BF7353@bp1.ad.bp.com> References: <20091120223822.23FEF3E941A@mail.cune.org> <02EE75E8713B4A1BBA3BBEB03B91AF7D@bp1.ad.bp.com> <7449BC3B2C05406DAE264F5A11BAF110@bp1.ad.bp.com> <0E0B2CB55DD7426C91286784F4BF7353@bp1.ad.bp.com> Message-ID: <87f2f3b90911211651j6f0d5fd1raab7a77805bcfb56@mail.gmail.com> On Sat, Nov 21, 2009 at 4:10 AM, Dave Stevenson wrote: > Sorry for the extended forum thread on this subject - This really IS the > last post ! > > I have now got the ATA to work without the dialplan fix provided by > Michael. > > After I'd implemented the "fix", I had more of an idea of what the problem > was and was better able to go through the Polycom VPA-11 setup screens > through its web interface to see if there were any options that might have > had a bearing on the problem. > > Under Configuration > VoIP > Non-Line Config > General Parameters > VoIP General > > There is an option to "Append UserId" - the Default Value is Yes. > > That was where the "extra characters" were coming from. > > Setting this option to No, makes the ATA behave more as expected, > > Regards > Dave > > Nice work! I can understand why it was easier to look for this AFTER we did the band-aid solution and not before. :) You next task is to visit http://wiki.freeswitch.org and sign up. Please add the setup details for this device. I already created stub to get you going: http://wiki.freeswitch.org/wiki/Interop_List#Pluscom Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/00cd2e85/attachment-0002.html From dfansler at dv-fansler.com Sat Nov 21 18:57:18 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Sat, 21 Nov 2009 18:57:18 -0800 (PST) Subject: [Freeswitch-users] IP1001 Setup Message-ID: <24395241.1258858638513.JavaMail.root@whwamui-deputy.pas.sa.earthlink.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091121/877e3d61/attachment-0002.html From mcampbellsmith at gmail.com Sat Nov 21 23:35:24 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 22 Nov 2009 18:35:24 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS Message-ID: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> HI All, Has anyone got some recommendations on which ATA to buy that supports TLS and SRTP? Thanks! From yehavi.bourvine at gmail.com Sat Nov 21 23:39:19 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 22 Nov 2009 09:39:19 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: <193640CC-3E62-4248-8E80-CE7FE82653C0@jerris.com> References: <4B0266F4.8070602@savion.huji.ac.il> <4B0387F1.7070105@savion.huji.ac.il> <193640CC-3E62-4248-8E80-CE7FE82653C0@jerris.com> Message-ID: Thanks Mike! However, this doesn't fully solve my problem. When using sofia_contact() indeed it works ok with finding the destination's profile. However, it breaks the BLFs... When calling *sofia/sip_profile/local-user%local-do**main* the BLF works ok. When calling sofia_contact(*sofia/sip_profile/local-user at local-domain*) BLF doesn't work (nothing is sent to the watching phone). Any more clues??? Thanks! __Yehavi: 2009/11/20 Michael Jerris > check out sofia_contact function. If you use this in combination with > binding profiles together so they are one table I think this should work > right. > > Mike > > On Nov 18, 2009, at 12:36 AM, Eli Hayun wrote: > > > Brian West wrote: > >> > >> Why do you need to know the destination profile like that? You get to > >> pick that on your own so you should already know that before hand. > >> > >> > >> /b > >> > >> On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote: > >> > >> > >>> Hi > >>> We have more then one profile. To make a call I have to enter : bridge > >>> sofia/profile/number at ip > >>> The problem is when I use : "${use_profile}" I am getting the caller > >>> profile, and I need the destination profile. > >>> > >>> How do I get this information? > >>> > >> > > Thanks for your answer. > > > > The problem is when I call to that number that the phone hook to other > server, I cannot make the call. > > Is there is a variable that can tell me the destination profile? > > Lets say the other profile called "ph1" I have to dial > > sofia/ph1/xxxxx at host to make the call. Is there other way to do that? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/596028b4/attachment-0002.html From itamar at ispbrasil.com.br Sun Nov 22 00:20:04 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Sun, 22 Nov 2009 06:20:04 -0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> Message-ID: sipura/linksys look in ebay. On Sun, Nov 22, 2009 at 5:35 AM, Mark Campbell-Smith wrote: > HI All, > > Has anyone got some recommendations on which ATA to buy that supports > TLS and SRTP? > > Thanks! -- ------------ Itamar Reis Peixoto e-mail/msn/google talk/sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From mcampbellsmith at gmail.com Sun Nov 22 01:21:34 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 22 Nov 2009 20:21:34 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> Message-ID: <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> Do LInksys devices support TLS and SRTP that FS supports? 3102 at least doesn't according to this post http://osdir.com/ml/telephony.freeswitch.user/2008-08/msg00904.html On Sun, Nov 22, 2009 at 7:20 PM, Itamar Reis Peixoto wrote: > sipura/linksys > > look in ebay. > > > On Sun, Nov 22, 2009 at 5:35 AM, Mark Campbell-Smith > wrote: >> HI All, >> >> Has anyone got some recommendations on which ATA to buy that supports >> TLS and SRTP? >> >> Thanks! > > > > > -- > ------------ > > Itamar Reis Peixoto > > e-mail/msn/google talk/sip: itamar at ispbrasil.com.br > skype: itamarjp > icq: 81053601 > +55 11 4063 5033 > +55 34 3221 8599 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From itamar at ispbrasil.com.br Sun Nov 22 01:41:01 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Sun, 22 Nov 2009 07:41:01 -0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> Message-ID: it's support SRTP On Sun, Nov 22, 2009 at 7:21 AM, Mark Campbell-Smith wrote: > Do LInksys devices support TLS and SRTP that FS supports? ?3102 at > least doesn't according to this post -- ------------ Itamar Reis Peixoto e-mail/msn/google talk/sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From lon at kickasspixels.com Sun Nov 22 03:25:40 2009 From: lon at kickasspixels.com (Lon Baker) Date: Sun, 22 Nov 2009 03:25:40 -0800 Subject: [Freeswitch-users] Clarification about channel variables please. Message-ID: <5d3e0dc60911220325i69f663b0meeff47c551be6999@mail.gmail.com> Are either global or regular channel variable mutable during a call? Or can they only be set before and after? Any clarification would help, since the existing wiki doesn't make it clear. Lon From michal.bielicki at halo2.pl Sun Nov 22 04:06:09 2009 From: michal.bielicki at halo2.pl (Michal Bielicki) Date: Sun, 22 Nov 2009 13:06:09 +0100 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: <4B086689.6080804@greatiam.com> References: <4B086689.6080804@greatiam.com> Message-ID: Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah: > > I need help as I cannot receive calls through VOIPUSER. This is a learning setup Attached are my conf files. What is wrong with them ? When I dial from a landline I get a continuous beep. > > Attached are my gateway and the conf file to transfer. Sopfia Status is my screen message. I can see a FAIL and cannot make head or tail of all that message. Hopefully anyone using voipuser or in fact any of you clever folks can make sense of this. > > Thanks for your time. > > 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy RFC2833 Mode! > 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] > 2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] > 2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec sofia/external/nobody at 213.166.5.133 PCMA/8000 20 ms 160 samples > 2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101 > 2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 (sofia/external/nobody at 213.166.5.133) State Change CS_NEW -> CS_INIT > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send signal sofia/external/nobody at 213.166.5.133 [BREAK] > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 (sofia/external/nobody at 213.166.5.133) Running State Change CS_INIT > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 (sofia/external/nobody at 213.166.5.133) State INIT > 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 sofia/external/nobody at 213.166.5.133 SOFIA INIT > 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 (sofia/external/nobody at 213.166.5.133) State Change CS_INIT -> CS_ROUTING > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send signal sofia/external/nobody at 213.166.5.133 [BREAK] > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 (sofia/external/nobody at 213.166.5.133) State INIT going to sleep > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 (sofia/external/nobody at 213.166.5.133) Running State Change CS_ROUTING > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 (sofia/external/nobody at 213.166.5.133) State ROUTING > 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 sofia/external/nobody at 213.166.5.133 SOFIA ROUTING > 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 sofia/external/nobody at 213.166.5.133 Standard ROUTING > 2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing anonymous->abeka in context public > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->unloop] continue=false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->outside_call] continue=true > Dialplan: sofia/external/nobody at 213.166.5.133 Absolute Condition [outside_call] > Dialplan: sofia/external/nobody at 213.166.5.133 Action set(outside_call=true) > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->call_debug] continue=true > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->public_extensions] continue=false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/ break=on-false > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->public_did] continue=false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->sip at sip.voipuser.org] continue=false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [sip at sip.voipuser.org] destination_number(abeka) =~ /08715042951/ break=on-false > Dialplan: sofia/external/nobody at 213.166.5.133 parsing [public->Inbound-abeka at sip.voipuser.org]] continue=false > Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [Inbound-abeka at sip.voipuser.org]] destination_number(abeka) =~ /[08444846450]/ break=on-false > 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 (sofia/external/nobody at 213.166.5.133) State Change CS_ROUTING -> CS_EXECUTE > 2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send signal sofia/external/nobody at 213.166.5.133 [BREAK] > 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 (sofia/external/nobody at 213.166.5.133) State ROUTING going to sleep > 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 (sofia/external/nobody at 213.166.5.133) Running State Change CS_EXECUTE > 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 (sofia/external/nobody at 213.166.5.133) State EXECUTE > 2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 sofia/external/nobody at 213.166.5.133 SOFIA EXECUTE > 2009-11-21 22:07:15.706658 [DEBUG] switch_core_state_machine.c:151 sofia/external/nobody at 213.166.5.133 Standard EXECUTE > EXECUTE sofia/external/nobody at 213.166.5.133 set(outside_call=true) > 2009-11-21 22:07:15.728613 [DEBUG] mod_dptools.c:748 sofia/external/nobody at 213.166.5.133 SET [outside_call]=[true] > 2009-11-21 22:07:15.728613 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/external/nobody at 213.166.5.133 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-11-21 22:07:15.728613 [DEBUG] switch_channel.c:1683 Send signal sofia/external/nobody at 213.166.5.133 [KILL] > 2009-11-21 22:07:15.728613 [DEBUG] switch_core_session.c:932 Send signal sofia/external/nobody at 213.166.5.133 [BREAK] > 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:491 (sofia/external/nobody at 213.166.5.133) State EXECUTE going to sleep > 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:398 (sofia/external/nobody at 213.166.5.133) Running State Change CS_HANGUP > 2009-11-21 22:07:15.735830 [DEBUG] switch_core_state_machine.c:434 (sofia/external/nobody at 213.166.5.133) State HANGUP > 2009-11-21 22:07:15.735830 [DEBUG] mod_sofia.c:338 Channel sofia/external/nobody at 213.166.5.133 hanging up, cause: NORMAL_CLEARING > 2009-11-21 22:07:15.737680 [DEBUG] mod_sofia.c:417 Responding to INVITE with: 480 > 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:46 sofia/external/nobody at 213.166.5.133 Standard HANGUP, cause: NORMAL_CLEARING > 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:434 (sofia/external/nobody at 213.166.5.133) State HANGUP going to sleep > 2009-11-21 22:07:15.742930 [DEBUG] switch_core_state_machine.c:476 (sofia/external/nobody at 213.166.5.133) State Change CS_HANGUP -> CS_REPORTING > 2009-11-21 22:07:15.742930 [DEBUG] switch_core_session.c:932 Send signal sofia/external/nobody at 213.166.5.133 [BREAK] > 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:398 (sofia/external/nobody at 213.166.5.133) Running State Change CS_REPORTING > 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:612 (sofia/external/nobody at 213.166.5.133) State REPORTING > 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:53 sofia/external/nobody at 213.166.5.133 Standard REPORTING, cause: NORMAL_CLEARING > 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:612 (sofia/external/nobody at 213.166.5.133) State REPORTING going to sleep > 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:411 (sofia/external/nobody at 213.166.5.133) State Change CS_REPORTING -> CS_DESTROY > 2009-11-21 22:07:15.800497 [DEBUG] switch_core_session.c:1068 Session 2 (sofia/external/nobody at 213.166.5.133) Locked, Waiting on external entities > 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/external/nobody at 213.166.5.133) Ended > 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/nobody at 213.166.5.133 [CS_DESTROY] > 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 (sofia/external/nobody at 213.166.5.133) State DESTROY > 2009-11-21 22:07:15.802636 [DEBUG] mod_sofia.c:255 sofia/external/nobody at 213.166.5.133 SOFIA DESTROY > 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:60 sofia/external/nobody at 213.166.5.133 Standard DESTROY > 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 (sofia/external/nobody at 213.166.5.133) State DESTROY going to sleep > > > > > > > > > > > > > > > > > > > > : you seem to have not specified an extension where the call should go to my voipuser.org setup looks like: I am also surprised that your setup works with a from-domain of sip.voipuser.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Michal Bielicki HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa t. +48228753290 | f. +48228753291 michal.bielicki at halokwadrat.pl | w. www.halokwadrat.pl Knowledge & Low Prices. Guaranteed! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/01cc6d3a/attachment-0002.html From tculjaga at gmail.com Sun Nov 22 04:15:24 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sun, 22 Nov 2009 13:15:24 +0100 Subject: [Freeswitch-users] Media got stuck after attended transfer... In-Reply-To: References: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com> <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com> <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> Message-ID: <65d96fc80911220415v70d0bafbvad56c4fcb4576d8b@mail.gmail.com> it is better to enhance mod_fax with t.38 support... we have done sometihng and it is close to be work... T. On Sat, Nov 21, 2009 at 2:17 AM, Michael Jerris wrote: > I think a better approach here is to use spandsp. We already have some > groundwork done for this. If you are interested in contributing, please > email consulting at freeswitch.org and we can discuss further. > > Mike > > On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: > > Hi, > > one of my customers is willing to contribute for t38 integration. > > The basic idea is to connect HylaFAX to FS: > t38modem <-> FreeSWITCH <-> Media Gateway with t38 support > All this without media proxy. > > Another idea might be to implement t38 origination/termination with a class > 1 modem input/output for use with HylaFAX. > > Do you know how much money we need to collect for t38 support? > How much time is needed for implementing this? > > Thanks, Klaus > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, October 16, 2009 2:10 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Media got stuck after attended > transfer... > > > > On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga > wrote: > > hi, any clue when can t38 be added? > > "Eventually." :) Of course, if we could get more to add to the bounty it > might grease the wheels of innovation. > > > http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch > > Of course, I was listening to my A.M radio the other day and they said that > there was this new invention called the Internet that would let people send > documents to each other electronically. Maybe you should look into that. > Next thing you know they'll come up with telephones that people don't have > to plug into the wall and can take with them in the car. ;) > > -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/1b995dfa/attachment-0002.html From Mailings at kh-dev.de Sun Nov 22 05:00:06 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 22 Nov 2009 14:00:06 +0100 Subject: [Freeswitch-users] Media got stuck after attended transfer... In-Reply-To: <65d96fc80911220415v70d0bafbvad56c4fcb4576d8b@mail.gmail.com> References: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com> <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com> <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> <65d96fc80911220415v70d0bafbvad56c4fcb4576d8b@mail.gmail.com> Message-ID: For "only" sending and receiving that's true. But my customer wants 2 things: - Using HylaFAX as fax server, as there are a lot of client apps and other tools - Connecting "real" fax machines using a Linksys/Cisco SPA2102 (as this is certified by their SIP/ISDN gateway vendor) So I could really need t38 handling in FS to don't make things more complicated as they already are... J Proxy mode doesn't work for me because it gives an error when resume-media-on-hold is set. Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Sunday, November 22, 2009 1:15 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Media got stuck after attended transfer... it is better to enhance mod_fax with t.38 support... we have done sometihng and it is close to be work... T. On Sat, Nov 21, 2009 at 2:17 AM, Michael Jerris > wrote: I think a better approach here is to use spandsp. We already have some groundwork done for this. If you are interested in contributing, please email consulting at freeswitch.org and we can discuss further. Mike On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: Hi, one of my customers is willing to contribute for t38 integration. The basic idea is to connect HylaFAX to FS: t38modem <-> FreeSWITCH <-> Media Gateway with t38 support All this without media proxy. Another idea might be to implement t38 origination/termination with a class 1 modem input/output for use with HylaFAX. Do you know how much money we need to collect for t38 support? How much time is needed for implementing this? Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, October 16, 2009 2:10 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Media got stuck after attended transfer... On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga > wrote: hi, any clue when can t38 be added? "Eventually." :) Of course, if we could get more to add to the bounty it might grease the wheels of innovation. http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/70a2ab76/attachment-0002.html From dfansler at dv-fansler.com Sun Nov 22 06:09:26 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Sun, 22 Nov 2009 09:09:26 -0500 (GMT-05:00) Subject: [Freeswitch-users] IP0010 SIP Phone Message-ID: <10222602.1258898966890.JavaMail.root@whwamui-deputy.pas.sa.earthlink.net> After the help of a couple of people from this list, I now have FreeSWITCH running - yeah! I have installed X-Lite on a couple of computers and they dial each other, play music on hold, etc. I have not yet connected to the outside world. I purchased an IP-0010 phone off eBay ($20 including shipping - docs at http://www.vanaccess.com/news/news_images/2007131_73_User%20Manual%20-%20IP0010.pdf) I cannot get this phone to work with the system. It gets an IP address, time/date, and a dial tone. After many tries with the http congifuration tool, I got the phone "configured" with the address of the SIP server, and a SIP User ID. When you dial an extension the FreeSWITCH window shows the following: sofica.c3844 Hanugup sofia/internal/101 at 192.168.1.165 [CS_NEW] [INCOMPATIBLE_DESTINATION] switch_core_session.c1139 Session 20 (sofia/internal/101 at 192.165.1.65) Ended switch_core_session.c1141 Close Channel sofia/internal/1001 at 192.168.1.165 [CS_DESTROY] Has anyone else tried this phone, or does anyone have suggestions I could try. I have looked through the website but have not found anything to help. Thanks, David David V. Fansler S/V Annabelle www.dv-fansler.com dfansler at dv-fansler.com From abeka at greatiam.com Sun Nov 22 09:53:13 2009 From: abeka at greatiam.com (Sam Abekah-Mensah) Date: Sun, 22 Nov 2009 17:53:13 +0000 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: References: <4B086689.6080804@greatiam.com> Message-ID: <4B097A89.2050400@greatiam.com> Hi Michael Thanks I had set it to send incoming calls to extension 1001. This is in the file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory. The contents are : Is there anything wrong with this please ? Thanks Michal Bielicki wrote: > > Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah: > >> >> I need help as I cannot receive calls through VOIPUSER. This is a >> learning setup Attached are my conf files. What is wrong with them ? >> When I dial from a landline I get a continuous beep. >> >> Attached are my gateway and the conf file to transfer. Sopfia Status >> is my screen message. I can see a FAIL and cannot make head or tail >> of all that message. Hopefully anyone using voipuser or in fact any >> of you clever folks can make sense of this. >> >> Thanks for your time. >> >> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy >> RFC2833 Mode! >> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec >> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >> 2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec >> Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] >> 2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec >> sofia/external/nobody at 213.166.5.133 PCMA/8000 20 ms 160 samples >> 2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf >> payload to 101 >> 2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 >> (sofia/external/nobody at 213.166.5.133) State Change CS_NEW -> CS_INIT >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/nobody at 213.166.5.133 [BREAK] >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/nobody at 213.166.5.133) Running State Change CS_INIT >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >> (sofia/external/nobody at 213.166.5.133) State INIT >> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 >> sofia/external/nobody at 213.166.5.133 SOFIA INIT >> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 >> (sofia/external/nobody at 213.166.5.133) State Change CS_INIT -> CS_ROUTING >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/nobody at 213.166.5.133 [BREAK] >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >> (sofia/external/nobody at 213.166.5.133) State INIT going to sleep >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/nobody at 213.166.5.133) Running State Change CS_ROUTING >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 >> (sofia/external/nobody at 213.166.5.133) State ROUTING >> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 >> sofia/external/nobody at 213.166.5.133 SOFIA ROUTING >> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 >> sofia/external/nobody at 213.166.5.133 Standard ROUTING >> 2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing >> anonymous->abeka in context public >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->unloop] continue=false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->outside_call] continue=true >> Dialplan: sofia/external/nobody at 213.166.5.133 Absolute Condition >> [outside_call] >> Dialplan: sofia/external/nobody at 213.166.5.133 Action >> set(outside_call=true) >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->call_debug] continue=true >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->public_extensions] continue=false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >> [public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/ >> break=on-false >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->public_did] continue=false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >> [public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->sip at sip.voipuser.org] continue=false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >> [sip at sip.voipuser.org] destination_number(abeka) =~ /08715042951/ >> break=on-false >> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >> [public->Inbound-abeka at sip.voipuser.org]] continue=false >> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >> [Inbound-abeka at sip.voipuser.org]] destination_number(abeka) =~ >> /[08444846450]/ break=on-false >> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 >> (sofia/external/nobody at 213.166.5.133) State Change CS_ROUTING -> >> CS_EXECUTE >> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/nobody at 213.166.5.133 [BREAK] >> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 >> (sofia/external/nobody at 213.166.5.133) State ROUTING going to sleep >> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/nobody at 213.166.5.133) Running State Change CS_EXECUTE >> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 >> (sofia/external/nobody at 213.166.5.133) State EXECUTE >> 2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 >> sofia/external/nobody at 213.166.5.133 SOFIA EXECUTE >> 2009-11-21 22:07:15.706658 [DEBUG] switch_core_state_machine.c:151 >> sofia/external/nobody at 213.166.5.133 Standard EXECUTE >> EXECUTE sofia/external/nobody at 213.166.5.133 set(outside_call=true) >> 2009-11-21 22:07:15.728613 [DEBUG] mod_dptools.c:748 >> sofia/external/nobody at 213.166.5.133 SET [outside_call]=[true] >> 2009-11-21 22:07:15.728613 [NOTICE] switch_core_state_machine.c:179 >> Hangup sofia/external/nobody at 213.166.5.133 [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-11-21 22:07:15.728613 [DEBUG] switch_channel.c:1683 Send signal >> sofia/external/nobody at 213.166.5.133 [KILL] >> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/nobody at 213.166.5.133 [BREAK] >> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:491 >> (sofia/external/nobody at 213.166.5.133) State EXECUTE going to sleep >> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/nobody at 213.166.5.133) Running State Change CS_HANGUP >> 2009-11-21 22:07:15.735830 [DEBUG] switch_core_state_machine.c:434 >> (sofia/external/nobody at 213.166.5.133) State HANGUP >> 2009-11-21 22:07:15.735830 [DEBUG] mod_sofia.c:338 Channel >> sofia/external/nobody at 213.166.5.133 hanging up, cause: NORMAL_CLEARING >> 2009-11-21 22:07:15.737680 [DEBUG] mod_sofia.c:417 Responding to >> INVITE with: 480 >> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:46 >> sofia/external/nobody at 213.166.5.133 Standard HANGUP, cause: >> NORMAL_CLEARING >> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:434 >> (sofia/external/nobody at 213.166.5.133) State HANGUP going to sleep >> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_state_machine.c:476 >> (sofia/external/nobody at 213.166.5.133) State Change CS_HANGUP -> >> CS_REPORTING >> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/nobody at 213.166.5.133 [BREAK] >> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/nobody at 213.166.5.133) Running State Change CS_REPORTING >> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:612 >> (sofia/external/nobody at 213.166.5.133) State REPORTING >> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:53 >> sofia/external/nobody at 213.166.5.133 Standard REPORTING, cause: >> NORMAL_CLEARING >> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:612 >> (sofia/external/nobody at 213.166.5.133) State REPORTING going to sleep >> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:411 >> (sofia/external/nobody at 213.166.5.133) State Change CS_REPORTING -> >> CS_DESTROY >> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_session.c:1068 Session >> 2 (sofia/external/nobody at 213.166.5.133) Locked, Waiting on external >> entities >> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1086 >> Session 2 (sofia/external/nobody at 213.166.5.133) Ended >> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1088 Close >> Channel sofia/external/nobody at 213.166.5.133 [CS_DESTROY] >> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >> (sofia/external/nobody at 213.166.5.133) State DESTROY >> 2009-11-21 22:07:15.802636 [DEBUG] mod_sofia.c:255 >> sofia/external/nobody at 213.166.5.133 SOFIA DESTROY >> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:60 >> sofia/external/nobody at 213.166.5.133 Standard DESTROY >> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >> (sofia/external/nobody at 213.166.5.133) State DESTROY going to sleep >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> : > > > you seem to have not specified an extension where the call should go to > > my voipuser.org setup looks like: > > > > > > > > > > > > > > > > I am also surprised that your setup works with a from-domain of > sip.voipuser.org > >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > *Michal Bielicki* > HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa > t. +48228753290 | f. +48228753291 > michal.bielicki at halokwadrat.pl > | > w. www.halokwadrat.pl > > > > *Knowledge & Low Prices. Guaranteed!* > From JCasale at activenetwerx.com Sun Nov 22 10:02:18 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Sun, 22 Nov 2009 18:02:18 +0000 Subject: [Freeswitch-users] Call quality problem w/ Snom M3's Message-ID: Not sure where to start with this one, the outgoing leg to our sip provider sounds perfectly fine but with the our M3's the incoming leg is super choppy. Using twinkle on my laptop yields good results in both directions so there must be an issue with just the snoms, their firmware was very old but updating it didn't help. Our sip provider uses ulaw and the snoms and vars.xml both have ulaw set as top pref. Any ideas what to look at next? Thanks, jlc From achaloyan at yahoo.com Sun Nov 22 10:02:48 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Sun, 22 Nov 2009 10:02:48 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <1258732768082-4038514.post@n2.nabble.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> Message-ID: <552708.67071.qm@web111314.mail.gq1.yahoo.com> We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org ________________________________ From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/a8f51f90/attachment-0002.html From mike at jerris.com Sun Nov 22 10:20:17 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 22 Nov 2009 13:20:17 -0500 Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <552708.67071.qm@web111314.mail.gq1.yahoo.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> Message-ID: Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan wrote: > We discussed build integration related issues a few months ago with > Mike and seemed to find a solution which would work for both UniMRCP > and FreeSWITCH source trees. > > Now I've just got a chance to look into this a bit closer trying to > further complete VS2008 build integration in FreeSWITCH. So I've got > it working, the module is not only being built, but also is getting > loaded. Current build integration is not as seamless as I want it to > be, but probably we can start with what we have now and then discuss > and identify what can be done in the future. This concerns not only > build integration but overall integrity. > > So would you be interested in the patch? Where should I upload it? > I thought I had a Jira account, but not sure it exists any more. > > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org > > > From: Jeff Lenk > To: freeswitch-users at lists.freeswitch.org > Sent: Fri, November 20, 2009 7:59:28 PM > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch > & uniMRCP > > > That module is not currently being built for Windows. Also the library > unimrcp needs build integration work with FS to make that happen under > windows. > > > ss1 wrote: > > > > Hi Everyone, > > > > Please help freeswitch experts... !!! > > > > i have been working on freeswitch from last 2 days. i have > downloaded > > freeswitch and unimrcp (server + client) for windows. > > I tested the unimrcp client and server, which is running fine with > the > > command: run synth and run recog. I got both synth.pcm & recog.pcm > files. > > > > But my objective is to call Freeswitch through x-lite, where > freeswitch > > should call unimrcp client and return the PCM files. > > > > I tried it alot, but unable to do it. after lots of reading i > found that i > > do not have mod_unimrcp. i do not know from where to download it > and how > > to merge it into freeswitch. > > > > I would be very thankful if you may help. > > > > Thanks, > > ss > > > > > > -- > View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/f9dfb536/attachment-0002.html From mattdfong at gmail.com Sun Nov 22 10:48:50 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Mon, 23 Nov 2009 01:48:50 +0700 Subject: [Freeswitch-users] Recording with Native File PCMU Message-ID: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> I'm trying to conserve processor power by recording in native file format, PCMU in my case. It works great with the following line session:execute("record", "/tmp/my_recording."..session:getVariable("read_codec")); however it fails to work with session:execute("record_session", "/tmp/my_recording."..session:getVariable("read_codec")); or record = api:execute("sched_api", '+1 none uuid_record '..session:getVariable("uuid")..' start /tmp/my_recording.'..session:getVariable("read_codec")); Why is it that it works with record, but not with record_session or uuid_record? Is there something I'm over looking? In the latter two the consul reports 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File [/tmp/my_recording.PCMU] 8000hz as if it's recording, but /tmp/my_recording.PCMU never shows up. However if I change it to .wav instead of .PCMU it works. Any ideas? --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/6eac0a38/attachment-0002.html From achaloyan at yahoo.com Sun Nov 22 11:03:36 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Sun, 22 Nov 2009 11:03:36 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> Message-ID: <634117.24120.qm@web111311.mail.gq1.yahoo.com> Mike, >Jira is the best, otherwise just mail me the patch and I'll take a look. I've uploaded the patch against svn trunk to http://jira.freeswitch.org/browse/MODUNIMRCP-6 it's made for win32 debug only yet. >Can you take a look at my last patch to the module to make it build please. I see. I've not noticed this change introduces API change, makes no sense to me now. I'll provide more convenient solution soon. Arsen. ________________________________ From: Michael Jerris To: "freeswitch-users at lists.freeswitch.org" Sent: Sun, November 22, 2009 10:20:17 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan wrote: We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. > >Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. > >So would you be interested in the patch? Where should I upload it? >I thought I had a Jira account, but not sure it exists any more. > >-- >Arsen Chaloyan >The author of > UniMRCP >http://www.unimrcp.org > > > > > ________________________________ From: Jeff Lenk >To: freeswitch-users at lists.freeswitch.org >Sent: Fri, November 20, 2009 7:59:28 PM >Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP > > >That module is not currently being built for Windows. Also the library >unimrcp needs build integration work with FS to make that happen under >windows. > > >ss1 wrote: >> >> Hi Everyone, >> >> Please help freeswitch experts... !!! >> >> i have been working on freeswitch from last 2 days. i have downloaded >> freeswitch and unimrcp (server + client) for windows. >> I tested the unimrcp client and server, which is running fine with the >> command: run synth and run recog. I got both synth.pcm & recog.pcm files. >> >> But my objective is to call Freeswitch through x-lite, where freeswitch >> should call unimrcp client and return the PCM files. >> >> I tried it alot, but unable to do it. after lots of reading i found that i >> do not have mod_unimrcp. i do not know from where to download it and how >> to merge it into freeswitch. >> >> I would > be very thankful if you may help. >> >> Thanks, >> ss >> >> > >-- >View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html >Sent from the freeswitch-users mailing list archive at Nabble.com. > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > _______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/1e431b0e/attachment-0002.html From Prometheus001 at gmx.net Sun Nov 22 11:27:20 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 22 Nov 2009 20:27:20 +0100 Subject: [Freeswitch-users] Problems with Voicemail In-Reply-To: <4B07D999.4040004@gmx.net> References: <4B073ACD.1090708@gmx.net> <976A0342-4F4B-4035-9201-D56F8625AE12@freeswitch.org> <4B07D999.4040004@gmx.net> Message-ID: <4B099098.2040408@gmx.net> I now created a file inbox.PCMA and get the following: * inbox.PCMA is played * the recorded voive mail file is not played (FS does not even try to do that) * then I hear o "to listen to the recording press 1" o "to save the recording press 2" o ... Here's the debug output 2009-11-22 20:17:43.701098 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-22 20:17:44.278600 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-22 20:17:44.386776 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA] 8000hz 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-listen_to_recording.wav] (en:en) 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-22 20:17:45.201099 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-22 20:17:46.419933 [DEBUG] switch_ivr_play_say.c:1428 done playing file nGrepping port 3306 I can see that the correct filenames are retrieved from the mysql/odbc database: 1258894746.0.200.sip1.mydomain.com$d11c2a74-d766-11de-997b-bd7aecdc2a16.Gor Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_c57a5e84-d766-11de-997b-bd7aecdc2a16.wav.4..B_NORMAL.....47 1258897120.0.200.sip1.mydomain.com$580dafee-d76c-11de-84d4-a1cd7fa320b3.Gor Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_4d484a7e-d76c-11de-84d4-a1cd7fa320b3.wav.5..B_NORMAL......... Both filenames can be read. Best regards Peter Peter P GMX schrieb: > I installed all sounds from SVN, but > > usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA > > isn't there. I checked another, older installation and couldn't this > file either. > > I think that freeswitch tries to build a sound path for the file to be > played, and some parts of the path are missing. > I expect it would play a recorded message at that time in > /usr/local/freeswitch/storage/voicemail/default/${domain} and the > defined format is "wav" not pcma. > > I also set "storage_dir" explicitely in the voicemail configs,but this > also didn't help. > > Best regards > Peter > > > Brian West schrieb: > >> I'm going to venture to guess maybe the file was recorded in a >> different codec and NOT pcma? >> >> /b >> >> On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: >> >> >> >>> 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / >>> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From achaloyan at yahoo.com Sun Nov 22 11:36:59 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Sun, 22 Nov 2009 11:36:59 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <634117.24120.qm@web111311.mail.gq1.yahoo.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> <634117.24120.qm@web111311.mail.gq1.yahoo.com> Message-ID: <192864.88786.qm@web111315.mail.gq1.yahoo.com> Mike, upgrade UniMRCP to http://code.google.com/p/unimrcp/source/detail?r=1297 and remove that #if from mod_unimrcp. API is backward compatible now src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c =================================================================== --- src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c (revision 15605) +++ src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c (working copy) @@ -3510,11 +3510,7 @@ } /* Set up the media engine that will be shared with all profiles */ -#if UNI_VERSION_AT_LEAST(0,8,0) - media_engine = mpf_engine_create(1, pool); -#else media_engine = mpf_engine_create(pool); -#endif if (media_engine) { mrcp_client_media_engine_register(client, media_engine, "MediaEngine"); } Arsen ________________________________ From: Arsen Chaloyan To: freeswitch-users at lists.freeswitch.org Sent: Sun, November 22, 2009 11:03:36 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Mike, >Jira is the best, otherwise just mail me the patch and I'll take a look. I've uploaded the patch against svn trunk to http://jira.freeswitch.org/browse/MODUNIMRCP-6 it's made for win32 debug only yet. >Can you take a look at my last patch to the module to make it build please. I see. I've not noticed this change introduces API change, makes no sense to me now. I'll provide more convenient solution soon. Arsen. ________________________________ From: Michael Jerris To: "freeswitch-users at lists.freeswitch.org" Sent: Sun, November 22, 2009 10:20:17 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan wrote: We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. > >Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build > integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. > >So would you be interested in the patch? Where should I upload it? >I thought I had a Jira account, but not sure it exists any more. > >-- >Arsen Chaloyan >The author of > UniMRCP >http://www.unimrcp.org > > > > > ________________________________ From: Jeff Lenk >To: freeswitch-users at lists.freeswitch.org >Sent: Fri, > November 20, 2009 7:59:28 PM >Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP > > >That module is not currently being built for Windows. Also the library >unimrcp needs build integration work with FS to make that happen under >windows. > > >ss1 wrote: >> >> Hi Everyone, >> >> Please help freeswitch experts... !!! >> >> i have been working on freeswitch from last 2 days. i have downloaded >> freeswitch and unimrcp (server + client) for windows. >> I tested the unimrcp client and server, which is running fine with the >> command: run synth and run recog. I got both synth.pcm & recog.pcm files. >> >> But my objective is to call Freeswitch through x-lite, where freeswitch >> should call unimrcp client and return the PCM files. >> >> I tried it alot, but unable to do it. after lots of reading i found that i >> do not have mod_unimrcp. i do not know from where to download it and how >> to merge it into freeswitch. >> >> I would > be very thankful if you may help. >> >> Thanks, >> ss >> >> > >-- >View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html >Sent from the freeswitch-users mailing list archive at Nabble.com. > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > _______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/ea36e5f4/attachment-0002.html From timuckun at gmail.com Sun Nov 22 15:43:38 2009 From: timuckun at gmail.com (Tim Uckun) Date: Mon, 23 Nov 2009 12:43:38 +1300 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> References: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> Message-ID: <855e4dcf0911221543o222bef63t1c3340b0a41d57c1@mail.gmail.com> On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman wrote: > Hi Sam, > Take a look at mod_xml_curl. ?Pretty sure it'll do everything you're looking > for. Looking at that diagram it seems like mod_xml_curl makes a call for every SIP connection. That seems like overkill. Is there a way to set it up so that it caches the XML it got for a period of time? From dujinfang at gmail.com Sun Nov 22 15:54:11 2009 From: dujinfang at gmail.com (Seven Du) Date: Mon, 23 Nov 2009 07:54:11 +0800 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> Message-ID: <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> did you try without any .wav or .PCMU? 2009/11/23 Matthew Fong > I'm trying to conserve processor power by recording in native file format, > PCMU in my case. It works great with the following line > > session:execute("record", > "/tmp/my_recording."..session:getVariable("read_codec")); > > however it fails to work with > > session:execute("record_session", > "/tmp/my_recording."..session:getVariable("read_codec")); > or > record = api:execute("sched_api", '+1 none uuid_record > '..session:getVariable("uuid")..' start > /tmp/my_recording.'..session:getVariable("read_codec")); > > Why is it that it works with record, but not with record_session or > uuid_record? Is there something I'm over looking? In the latter two the > consul reports > > 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File > [/tmp/my_recording.PCMU] 8000hz > > as if it's recording, but /tmp/my_recording.PCMU never shows up. However if > I change it to .wav instead of .PCMU it works. Any ideas? > > --matt > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/019664f3/attachment-0002.html From gurveshb at yahoo.com Sun Nov 22 01:15:22 2009 From: gurveshb at yahoo.com (Gurvesh Bhutiani) Date: Sun, 22 Nov 2009 01:15:22 -0800 (PST) Subject: [Freeswitch-users] Broadvoice 32 transcoding support? In-Reply-To: <87f2f3b90911200934n20373bc6tf01677ec8d2bb11d@mail.gmail.com> Message-ID: <58250.88935.qm@web35705.mail.mud.yahoo.com> Yes, the latest trunk works. Thank you! Gaurav --- On Fri, 11/20/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Broadvoice 32 transcoding support? > To: freeswitch-users at lists.freeswitch.org > Date: Friday, November 20, 2009, 9:34 AM > > > On Fri, Nov 20, 2009 at 1:16 AM, > Gaurav Singh > wrote: > > > Hi, > > Does freeswitch support transcoding between broadvoice > (BV32 ) and G711 > ? > Try latest trunk. There was a new update just added very > recently... > -MC > > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kdjakovic at hotmail.com Sun Nov 22 10:46:13 2009 From: kdjakovic at hotmail.com (katarina djakovic) Date: Sun, 22 Nov 2009 19:46:13 +0100 Subject: [Freeswitch-users] Problems with sighup and rotating csv files Message-ID: Hi, I am using the Freeswitch 1.0.4pre7. Great application, but I encountered a problem wich I can not solve since I am very new to it. Two things are happening. 1) The mod_cdr_csv.c (line 122 do_rotate()) does not always respond to sighup signal to rotate the cdr-csv files. Some times it happens and some times it does not. I can not see any pattern in the behaviour. Seems that sometimes functions in the mod_cdr_csv.c catch the signal and some times they do not. 2) Playing with the "kill -HUP fspid" all of a sudden I started getting two freeswitch processes in the process list. One being parent of another. Then, when I send the sighup signal to the parent - the console dies off and the other freeswitch process stays (leaving the comment "Hangup" in the fsconsole). Freeswitch conitnues to work with the remaining process. In case when I send the sighup to the child, it will rotate the log files. However, it always rotates the freeswitch.log, but randomly rotates the cdr-csv files. 3) I have a feeling that above behaviours are somehow connected, but do not understand how. Anyone can help? Any comment or idea will be very very much apreciated. Cheers, Katarina Windows Live: Make it easier for your friends to see what you?re up to on Facebook. _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/c112aef2/attachment-0002.html From jlenk at frontiernet.net Sun Nov 22 20:16:28 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Sun, 22 Nov 2009 20:16:28 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <552708.67071.qm@web111314.mail.gq1.yahoo.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> Message-ID: <1258949788572-4048969.post@n2.nabble.com> Hi Arsen, I would be happy to help with the FS integration if you want - please do put your patch in a Jira. Jeff Date: Sun, 22 Nov 2009 10:09:41 -0800 From: ml-node+4047148-1118239605 at n2.nabble.com To: jlenk at frontiernet.net Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org From: Jeff Lenk <[hidden email]> To: [hidden email] Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org View message @ http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html To unsubscribe from Re: need help !! Problem with freeswitch & uniMRCP, click here. _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/ -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4048969.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091122/40949795/attachment-0002.html From thangappan143 at gmail.com Sun Nov 22 20:34:01 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Mon, 23 Nov 2009 10:04:01 +0530 Subject: [Freeswitch-users] Fwd: Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> Message-ID: <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> I am waiting only for DTMF events. That's why I am setting freeswitch variable for knowing whether the playback has done. My question is "why this freeswitch variable is not setting properly when I play back more than 10 files using playback_delimiter option?". When I play back lesser than ten voice files the variable has been set properly. What could be the reason? ---------- Forwarded message ---------- From: Thangappan.M Date: Sat, Nov 21, 2009 at 2:52 PM Subject: Problem while playing more than 10 voice files using playback To: freeswitch-users Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have implemented using Perl language. I did the following steps: => Set the playback_delimiter variable => Set the playback_sleep_val variable => Set the event lock as true => Set the freeswitch ( my own) variable as zero => Wait in the loop until the variable is been set as zero => Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) => Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played successfully). => Wait in the loop until the variable is been set as one. => Set the Event lock as false => Trying to get the DTMF digits ( Have a assurance that all the voice files are played). The problem is, The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* *NOTE*: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. Please help me................? Thanks in advance. -- Regards, Thangappan.M -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/720ecbd7/attachment-0002.html From qinglan_zeng at hotmail.com Sun Nov 22 20:51:55 2009 From: qinglan_zeng at hotmail.com (=?gb2312?B?tPPE4MjL?=) Date: Mon, 23 Nov 2009 04:51:55 +0000 Subject: [Freeswitch-users] FS compile error under Windows: error LNK2019 In-Reply-To: References: Message-ID: All, I tried to compile FS source code under Windows while there are lots of errors: Error LNK2019, external _imp_sleep at 4 can not be resolved, this function was referred by _tMCRTStartup. Some other more similiar errors detail information attached. Any ideas? Thanks Daniel Zeng From: freeswitch-users-request at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 41, Issue 146 To: freeswitch-users at lists.freeswitch.org Date: Sun, 22 Nov 2009 11:37:32 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --??????-- From: mattdfong at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Mon, 23 Nov 2009 01:48:50 +0700 Subject: [Freeswitch-users] Recording with Native File PCMU I'm trying to conserve processor power by recording in native file format, PCMU in my case. It works great with the following line session:execute("record", "/tmp/my_recording."..session:getVariable("read_codec")); however it fails to work with session:execute("record_session", "/tmp/my_recording."..session:getVariable("read_codec")); or record = api:execute("sched_api", '+1 none uuid_record '..session:getVariable("uuid")..' start /tmp/my_recording.'..session:getVariable("read_codec")); Why is it that it works with record, but not with record_session or uuid_record? Is there something I'm over looking? In the latter two the consul reports 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File [/tmp/my_recording.PCMU] 8000hz as if it's recording, but /tmp/my_recording.PCMU never shows up. However if I change it to .wav instead of .PCMU it works. Any ideas? --matt --??????-- From: achaloyan at yahoo.com To: freeswitch-users at lists.freeswitch.org Date: Sun, 22 Nov 2009 11:03:36 -0800 Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Mike, >Jira is the best, otherwise just mail me the patch and I'll take a look. I've uploaded the patch against svn trunk to http://jira.freeswitch.org/browse/MODUNIMRCP-6 it's made for win32 debug only yet. >Can you take a look at my last patch to the module to make it build please. I see. I've not noticed this change introduces API change, makes no sense to me now. I'll provide more convenient solution soon. Arsen. From: Michael Jerris To: "freeswitch-users at lists.freeswitch.org" Sent: Sun, November 22, 2009 10:20:17 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan wrote: We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --??????-- From: Prometheus001 at gmx.net To: freeswitch-users at lists.freeswitch.org Date: Sun, 22 Nov 2009 20:27:20 +0100 Subject: Re: [Freeswitch-users] Problems with Voicemail I now created a file inbox.PCMA and get the following: * inbox.PCMA is played * the recorded voive mail file is not played (FS does not even try to do that) * then I hear o "to listen to the recording press 1" o "to save the recording press 2" o ... Here's the debug output 2009-11-22 20:17:43.701098 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-22 20:17:44.278600 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-22 20:17:44.386776 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA] 8000hz 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-listen_to_recording.wav] (en:en) 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16 at 8000hz 1 channels 20ms 2009-11-22 20:17:45.201099 [DEBUG] switch_core_io.c:660 sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-22 20:17:46.419933 [DEBUG] switch_ivr_play_say.c:1428 done playing file nGrepping port 3306 I can see that the correct filenames are retrieved from the mysql/odbc database: 1258894746.0.200.sip1.mydomain.com$d11c2a74-d766-11de-997b-bd7aecdc2a16.Gor Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_c57a5e84-d766-11de-997b-bd7aecdc2a16.wav.4..B_NORMAL.....47 1258897120.0.200.sip1.mydomain.com$580dafee-d76c-11de-84d4-a1cd7fa320b3.Gor Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_4d484a7e-d76c-11de-84d4-a1cd7fa320b3.wav.5..B_NORMAL......... Both filenames can be read. Best regards Peter Peter P GMX schrieb: > I installed all sounds from SVN, but > > usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA > > isn't there. I checked another, older installation and couldn't this > file either. > > I think that freeswitch tries to build a sound path for the file to be > played, and some parts of the path are missing. > I expect it would play a recorded message at that time in > /usr/local/freeswitch/storage/voicemail/default/${domain} and the > defined format is "wav" not pcma. > > I also set "storage_dir" explicitely in the voicemail configs,but this > also didn't help. > > Best regards > Peter > > > Brian West schrieb: > >> I'm going to venture to guess maybe the file was recorded in a >> different codec and NOT pcma? >> >> /b >> >> On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: >> >> >> >>> 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / >>> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > --??????-- From: achaloyan at yahoo.com To: freeswitch-users at lists.freeswitch.org Date: Sun, 22 Nov 2009 11:36:59 -0800 Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Mike, upgrade UniMRCP to http://code.google.com/p/unimrcp/source/detail?r=1297 and remove that #if from mod_unimrcp. API is backward compatible now src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c =================================================================== --- src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c (revision 15605) +++ src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c (working copy) @@ -3510,11 +3510,7 @@ } /* Set up the media engine that will be shared with all profiles */ -#if UNI_VERSION_AT_LEAST(0,8,0) - media_engine = mpf_engine_create(1, pool); -#else media_engine = mpf_engine_create(pool); -#endif if (media_engine) { mrcp_client_media_engine_register(client, media_engine, "MediaEngine"); } Arsen From: Arsen Chaloyan To: freeswitch-users at lists.freeswitch.org Sent: Sun, November 22, 2009 11:03:36 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Mike, >Jira is the best, otherwise just mail me the patch and I'll take a look. I've uploaded the patch against svn trunk to http://jira.freeswitch.org/browse/MODUNIMRCP-6 it's made for win32 debug only yet. >Can you take a look at my last patch to the module to make it build please. I see. I've not noticed this change introduces API change, makes no sense to me now. I'll provide more convenient solution soon. Arsen. From: Michael Jerris To: "freeswitch-users at lists.freeswitch.org" Sent: Sun, November 22, 2009 10:20:17 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan wrote: We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ????????????? ????????????????! ????? _________________________________________________________________ MSN????????????????25???????????2010????????? http://kaba.msn.com.cn/?k=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/4fb7f97a/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: compile_error-2.JPG Type: image/pjpeg Size: 115300 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/4fb7f97a/attachment-0002.bin From mike at jerris.com Sun Nov 22 21:12:12 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 00:12:12 -0500 Subject: [Freeswitch-users] FS compile error under Windows: error LNK2019 In-Reply-To: References: Message-ID: On Nov 22, 2009, at 11:51 PM, ??? wrote: > From keith at laaks.com Mon Nov 23 00:00:04 2009 From: keith at laaks.com (Keith Laaks) Date: Mon, 23 Nov 2009 10:00:04 +0200 Subject: [Freeswitch-users] Adding headers to INFO messages for Advice of Charge on SNOM Message-ID: <1258963204.4961.8.camel@keithl-lt> Hi, I have tried maintaining charging information on a SNOM 300's display using 'display' - but found that the phone has some timer, whereby every 60 seconds it wipes out whatever happens to be on the display at that time and replaces is with the dialled number. So not a viable option as it impacts usability. Really annoying when the display was just updated with valuable information for the user and a split second later it gets replaced. [If somebody knows how to disable this behaviour - please do tell...] I see that SNOM supports a number of features for Advice of Charge. >From their Wiki: http://wiki.snom.com/Advice_of_charge_%28AOC%29_in_SIP Example of an SIP-Info Message: ----------------------------------------------------- INFO sip:bla at snom.com SIP/2.0 From: ;tag=5354n3 To: ;tag=33rfh3 CSeq: 23423 INFO Call-ID: 3452tw43dt354dm03 AOC: charging;state=active; charging-info=currency; currency=EUR; amount=2000; multiplier=0.001 Content-Length: 0 ----------------------------------------------------- So the question - Is there some method available today to add these additional 'new' headers to an INFO message I can send out to these phones? If not, I guess it's a matter of looking at enhancing the "case SWITCH_MESSAGE_INDICATE_DISPLAY" section in mod_sofia.c ? Best Regards Keith From shiyanov at gmail.com Mon Nov 23 02:05:12 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Mon, 23 Nov 2009 13:05:12 +0300 Subject: [Freeswitch-users] Clarification about channel variables please. In-Reply-To: <5d3e0dc60911220325i69f663b0meeff47c551be6999@mail.gmail.com> References: <5d3e0dc60911220325i69f663b0meeff47c551be6999@mail.gmail.com> Message-ID: both types of variables are mutable On Sun, Nov 22, 2009 at 2:25 PM, Lon Baker wrote: > Are either global or regular channel variable mutable during a call? > Or can they only be set before and after? > > Any clarification would help, since the existing wiki doesn't make it > clear. > > Lon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/ca5a8b80/attachment-0002.html From info at daccii.it Mon Nov 23 02:36:57 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Mon, 23 Nov 2009 11:36:57 +0100 Subject: [Freeswitch-users] User who answer the bridge in a execute_answer Message-ID: <4B0A65C9.10509@daccii.it> Hi, i'm writing some dialplan parts that get executed on execute_on_answer. In this dialplan that get executed i need to make a directory to handle recordings for record_session and my folder structure is: USER/YEAR/MONTH/HOUR-MINUTE-SECOND-CALLER_NUMBER.wav ------ ------ The call flow is: Call from external -> IVR -> Transfer to Group -> Execute on Answer -> system/bind_meta_app Pratically, i need the number (or better the user) that answered the call: what variable should i check? I tried with sip_from_user, callee_id_number and some other. Thank for your help, Best Regards, Daniele -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/eba245aa/attachment-0002.vcf From lakindia89 at gmail.com Mon Nov 23 03:25:52 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 23 Nov 2009 16:55:52 +0530 Subject: [Freeswitch-users] Callback to the user in ESL Message-ID: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> Hi, I'm using perl ESL to control the call in freeswitch. I'm having the following scenario, but not able to get it right. Dialplan: 1. User A calls to an extention (1000). 2. My ESL program will be running, and it answers the call. 3. Then the program will get a number from the user. 4. It will hangup the call. 5. The program has to call to the number that was given by the user. In the above scenario, I was able to do until the 4th step. After hangup the call, if I say originate it is not working. Any ideas on how to do this in ESL. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/233cc5ba/attachment-0002.html From oscav at hotmail.fr Mon Nov 23 03:45:17 2009 From: oscav at hotmail.fr (Oscav) Date: Mon, 23 Nov 2009 03:45:17 -0800 (PST) Subject: [Freeswitch-users] Execute on Answer with JavaScript Message-ID: <26476532.post@talk.nabble.com> Hi, How can we send the answer to the caller only when the callee answers, in JavaScript?? Many thanks. -- View this message in context: http://old.nabble.com/Execute-on-Answer-with-JavaScript-tp26476532p26476532.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at yes.net.ua Mon Nov 23 03:45:28 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Mon, 23 Nov 2009 13:45:28 +0200 Subject: [Freeswitch-users] Using odbc in FS core In-Reply-To: <191c3a030911210814l6e50b883uba61815fcd36afe1@mail.gmail.com> References: <1382216794.20091121134106@yes.net.ua> <1013085378.20091121140207@yes.net.ua> <191c3a030911210814l6e50b883uba61815fcd36afe1@mail.gmail.com> Message-ID: <1202092411.20091123134528@yes.net.ua> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/8c436c02/attachment-0002.html From piotr_zurek at biprotech.com Mon Nov 23 03:52:57 2009 From: piotr_zurek at biprotech.com (=?UTF-8?B?UGlvdHIgxbt1cmVr?=) Date: Mon, 23 Nov 2009 12:52:57 +0100 Subject: [Freeswitch-users] [This is a repost. I'm not sure if my message was delivered.] How to pick up someone's phone remotely. In-Reply-To: <4AFC005A.4090200@biprotech.com> References: <4AF9803D.9050806@biprotech.com> <4468a6770911100806v2cf1098epf0483ee5948cdebc@mail.gmail.com> <4AFC005A.4090200@biprotech.com> Message-ID: <4B0A7799.6050500@biprotech.com> Hello again. This is a repost. I'm having difficulties communicating with this list (I'm getting reports from the list saying something about "excessive bounces"...), so I'm not sure anybody got this message. I'm trying to mimic behavior of my analogue PBX with FS. I want to be able to answer any incomming/transfered (from IVR or a person) call remotely, and to cancel the possibility of intercepting this call afterwards. Greetings Peter -- Original message -- My problems evolve, because I didn't know all these functions in FS are so much dependent on each other. But I'm learning fast... The scenario I written about before appears to be too much simplified version of what I need to achieve. In fact, below scenario and solution works OK only one time - when someone calls and there's no person on the called extension, and someone manually answers that phone on other extension. Then any other person can't intercept this call. Thats is correct and needed behavior. But if the same person who answered the phone transfers this call - everything goes back to normal and below solution does not work because the call has been answered already and execute_on_answer does not execute ever again during this call/channel. The same happens if there's IVR on the external extension answering calls and then forwarding to extensions - everyone can intercept last call even if it's already answered because IVR answers all call on start (and execute_on_answer doesn't get executed). So I think I need similar solution but working everywhere: on calls and transfers. Is there some variable or some other thing that I could set to block and unblock intercept when needed to get wanted behavior. Any hints? Greetings Peter Piotr Zurek pisze: > Thank You for such an elegant and simple solution that I have not > thought about. > With an exception that I'm using FS 1.0.4 right now and it appears > that something changed in time and following line should use hash > instead of db (when using default 1.0.4 FS config): > . > After a few hours of experimenting everything works as planned. > > Thank You very much. > Peter > > Ognjen Seslija pisze: >> Add the following: >> >> . >> >> after >> >> > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> >> in local extensions default example, or change it globally previously >> than this extension. You can join us on IRC if you can any more >> questions (sekil). >> >> Regards, >> Ognjen >> >> >> >> On Tue, Nov 10, 2009 at 4:01 PM, Piotr ?urek >> > wrote: >> >> Hello. >> >> Thank You developers for Freeswitch. >> I have installed it lately and it's working quite nicely, but I >> have one problem: >> >> I need to mimic behavior of my current analogue PBX installation >> using Freeswitch. >> >> This is the scenario: >> In the office with a few desks (extensions 1000-1010) and only >> one person behind one of desks (whatever extension - in example >> 1000). >> 1. There's incoming call on _one_ of extensions 1001-1010 >> 2. The person on extension 1000 wants to answer this call on his >> phone so dials #37 and this call is redirected to his phone. >> >> That's how it works on my office on analogue PBX system. Anyone >> can answer a call from any other phone as long as it hasn't been >> answered already. >> >> I tried to use the intercept action (with global example in >> default config) but it's not what I need because it intercepts >> the call even if it's already answered. I need to intercept all >> but only unanswered calls. I tried to use Redirect but it does >> not work on other's extensions call's (or does it?). >> >> Please help. >> Peter ?urek >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> -------------- next part -------------- A non-text attachment was scrubbed... Name: piotr_zurek.vcf Type: text/x-vcard Size: 414 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/63c139c3/attachment-0002.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3678 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/63c139c3/attachment-0002.bin From maciej.aniserowicz at gmail.com Mon Nov 23 04:00:39 2009 From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz) Date: Mon, 23 Nov 2009 04:00:39 -0800 (PST) Subject: [Freeswitch-users] Question about rtp-timeout-sec variable Message-ID: <1258977639954-4050650.post@n2.nabble.com> Hello, I have 2 instances of FS: one controlled by my application (making calls with TCP commands, recording sessions, listening to events etc) and one acting as a remote gateway to which all users register. When I leave the default values of rtp-timeout-sec and brutally kill x-lite during conversation, the 'hangup' event with 'media_timeout' cause is obviously sent after the default 5 minutes (and until then, the other leg is still connected to a 'dead' channel). The question is: which FS instance is responsible for terminating the connection after timeout? Only the 'remote' FS instance config seems to work. I thought that the shortest configured value should cause the timeout, but it's not the case. Am I missing something, or is this the correct behavior? Regards, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Question-about-rtp-timeout-sec-variable-tp4050650p4050650.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at yes.net.ua Mon Nov 23 04:11:31 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Mon, 23 Nov 2009 14:11:31 +0200 Subject: [Freeswitch-users] Using odbc in FS core In-Reply-To: <1202092411.20091123134528@yes.net.ua> References: <1382216794.20091121134106@yes.net.ua> <1013085378.20091121140207@yes.net.ua> <191c3a030911210814l6e50b883uba61815fcd36afe1@mail.gmail.com> <1202092411.20091123134528@yes.net.ua> Message-ID: <16025244.20091123141131@yes.net.ua> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/51d67b45/attachment-0002.html From nameer.kazzaz at gmail.com Mon Nov 23 04:23:23 2009 From: nameer.kazzaz at gmail.com (Nameer Kazzaz) Date: Mon, 23 Nov 2009 12:23:23 +0000 Subject: [Freeswitch-users] SIP Digest nonce (stale="true") Message-ID: <4B0A7EBB.8040702@gmail.com> Hi Anthony, I'm having an issue with a gateway after the nonce-ttl expires we are sending stale="true", the cpe some how only likes stale=true without the "". I see on rev 15441 you made a change and marked it out. So my question is who is correct on this is it the CPE or are we sticking with the quoted ("true"). Thanks Nameer Kazzaz From Russell.Mosemann at cune.org Mon Nov 23 05:50:17 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Mon, 23 Nov 2009 13:50:17 -0000 Subject: [Freeswitch-users] [This is a repost. I'm not sure if my message was delivered.] How to pick up someone's phone remotely. In-Reply-To: <4B0A7799.6050500@biprotech.com> Message-ID: <20091123135017.D61CC43C11B@mail.cune.org> Piotr ??urek said: > This is a repost. I'm having difficulties communicating with this list=20 > (I'm getting reports from the list saying something about "excessive=20 > bounces"...), so I'm not sure anybody got this message. 1. http://lists.freeswitch.org/pipermail/freeswitch-users/ 2. Click the link for "Thread" for November 2009 3. Search for your topic -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From achaloyan at yahoo.com Mon Nov 23 06:33:12 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Mon, 23 Nov 2009 06:33:12 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <1258949788572-4048969.post@n2.nabble.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> <1258949788572-4048969.post@n2.nabble.com> Message-ID: <858430.90192.qm@web111301.mail.gq1.yahoo.com> Hi Jeff, Your input would be very helpful, I just wanted to understand where the problem is and contribute the way I can. I see you're the assignee, so please go ahead and let me know if there is anything left I can help with. Arsen. ________________________________ From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Mon, November 23, 2009 8:16:28 AM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Hi Arsen, I would be happy to help with the FS integration if you want - please do put your patch in a Jira. Jeff ________________________________ Date: Sun, 22 Nov 2009 10:09:41 -0800 From: [hidden email] To: [hidden email] Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org ________________________________ From: Jeff Lenk <[hidden email]> To: [hidden email] Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: > > Hi Everyone, > > Please help freeswitch experts... !!! > > i have been working on freeswitch from last 2 days. i have downloaded > freeswitch and unimrcp (server + client) for windows. > I tested the unimrcp client and server, which is running fine with the > command: run synth and run recog. I got both synth.pcm & recog.pcm files. > > But my objective is to call Freeswitch through x-lite, where freeswitch > should call unimrcp client and return the PCM files. > > I tried it alot, but unable to do it. after lots of reading i found that i > do not have mod_unimrcp. i do not know from where to download it and how > to merge it into freeswitch. > > I would be very thankful if you may help. > > Thanks, > ss > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ View message @ http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html To unsubscribe from Re: need help !! Problem with freeswitch & uniMRCP, click here. ________________________________ Hotmail: Trusted email with powerful SPAM protection. Sign up now. ________________________________ View this message in context: RE: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/75fb2718/attachment-0002.html From brian at freeswitch.org Mon Nov 23 06:42:33 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 08:42:33 -0600 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> Message-ID: <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> If you're doing native file you DO NOT put an extension on the file name. /b On Nov 22, 2009, at 5:54 PM, Seven Du wrote: > did you try without any .wav or .PCMU? From abeka at greatiam.com Mon Nov 23 07:22:30 2009 From: abeka at greatiam.com (Otis) Date: Mon, 23 Nov 2009 15:22:30 +0000 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX Message-ID: <4B0AA8B6.2080305@greatiam.com> Hi Folks Is anyone using this on Fedora and is there a binary or installation script anywhere Thanks From brian at freeswitch.org Mon Nov 23 07:30:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 09:30:08 -0600 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <4B0AA8B6.2080305@greatiam.com> References: <4B0AA8B6.2080305@greatiam.com> Message-ID: cd /usr/src wget http://www.freeswitch.org/eg/Makefile make /b On Nov 23, 2009, at 9:22 AM, Otis wrote: > Hi Folks > > Is anyone using this on Fedora and is there a binary or installation > script anywhere > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Nov 23 07:52:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 09:52:03 -0600 Subject: [Freeswitch-users] SIP Digest nonce (stale="true") In-Reply-To: <4B0A7EBB.8040702@gmail.com> References: <4B0A7EBB.8040702@gmail.com> Message-ID: <191c3a030911230752r70b702b1g61694350f56b01e0@mail.gmail.com> The quoted true is the correct way from my research. The commented line was to test a device, a grandstream, they apparently do not accept it with quotes and I was using the unquoted version it to gather evidence to issue a bug report to them. They told me it will be fixed in the next firmware, was this the brand of device you have as well? On Mon, Nov 23, 2009 at 6:23 AM, Nameer Kazzaz wrote: > Hi Anthony, > I'm having an issue with a gateway after the nonce-ttl expires we > are sending stale="true", the cpe some how only likes stale=true without > the "". I see on rev 15441 > < > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_reg.c?r=15441#l687 > > > you made a change and marked it out. So my question is who is correct on > this is it the CPE or are we sticking with the quoted ("true"). > > Thanks > Nameer Kazzaz > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/912e243c/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 23 07:54:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 09:54:50 -0600 Subject: [Freeswitch-users] Fwd: Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> Message-ID: <191c3a030911230754i1d863180q694292e08beb7e44@mail.gmail.com> Maybe it is a race condition, I can't tell you from just such a basic description the code is complicated and I would have to reproduce it myself, but I can tell you one more time for good measure that you should use execute_complete events to tell when a command you tried to execute has finished and not poll the channel for a variable to be set because FreeSWITCH is an asynchronous application in the mode you are describing and you can never be sure of the timing. On Sun, Nov 22, 2009 at 10:34 PM, Thangappan.M wrote: > I am waiting only for DTMF events. That's why I am setting freeswitch > variable for knowing whether the playback has done. > > My question is "why this freeswitch variable is not setting properly when I > play back more than 10 files using playback_delimiter option?". > > When I play back lesser than ten voice files the variable has been set > properly. What could be the reason? > > > > ---------- Forwarded message ---------- > From: Thangappan.M > Date: Sat, Nov 21, 2009 at 2:52 PM > Subject: Problem while playing more than 10 voice files using playback > To: freeswitch-users > > > Dear all, > > I am in the process of implementing IVR using event outbound > socket (async mode). > I have implemented using Perl language. > > I did the following steps: > => Set the playback_delimiter variable > => Set the playback_sleep_val variable > => Set the event lock as true > => Set the freeswitch ( my own) variable as zero > => Wait in the loop until the variable is been set as > zero > => Playback the voice files ( Here I combined the voice > files with the delimiter value if more than one voice files are there) > => Set the freeswitch(my own) variable as true ( This is > used to identify whether the voice files are played > successfully). > => Wait in the loop until the variable is been set as > one. > => Set the Event lock as false > > => Trying to get the DTMF digits ( Have a assurance > that all the voice files are played). > > The problem is, > > The above steps are working fine when the voice file count is > lesser than or equal to 10. After the voice files are played only the > variable(my own freeswitch) is set. Based on the variable I am doing further > things. > > But when I tried to give the voice files count of more than 10 > the variable has been set while starting to play back the first voice file > itself . Because of this I am not able to proceed further. > > *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* > > *NOTE*: I also referred mod_file_string documentation. In that they > specified 128 files can be used to play back the voice files using > playback_delimiter option. > > Please help me................? > Thanks in advance. > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/7139944b/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 23 07:57:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 09:57:47 -0600 Subject: [Freeswitch-users] Problems with sighup and rotating csv files In-Reply-To: References: Message-ID: <191c3a030911230757j4bfd84e4mbf80534cecce5107@mail.gmail.com> did you really mean 1.0.4pre7 ? We are now on pre release of 1.0.5 so I cannot really debug such an old version so you may want to install one of the newer version first before you report an issue and when you do use our issue tracker not this mailing list http://jira.freeswitch.org be sure to answer all the questions carefully when filing the report. On Sun, Nov 22, 2009 at 12:46 PM, katarina djakovic wrote: > Hi, > I am using the Freeswitch 1.0.4pre7. Great application, but I encountered a > problem wich I can not solve since I am very new to it. > Two things are happening. > 1) The mod_cdr_csv.c (line 122 do_rotate()) does not always respond to > sighup signal to rotate the cdr-csv files. Some times it happens and some > times it does not. I can not see any pattern in the behaviour. Seems that > sometimes functions in the mod_cdr_csv.c catch the signal and some times > they do not. > > 2) Playing with the "kill -HUP fspid" all of a sudden I started getting two > freeswitch processes in the process list. One being parent of another. > Then, when I send the sighup signal to the parent - the console dies off > and the other freeswitch process stays (leaving the comment "Hangup" in the > fsconsole). Freeswitch conitnues to work with the remaining process. > In case when I send the sighup to the child, it will rotate the log files. > However, it always rotates the freeswitch.log, but randomly rotates the > cdr-csv files. > > 3) I have a feeling that above behaviours are somehow connected, but do not > understand how. > > Anyone can help? > Any comment or idea will be very very much apreciated. > Cheers, > Katarina > > ------------------------------ > Windows Live: Make it easier for your friends to see what you?re up to on > Facebook. > > ------------------------------ > Windows Live: Friends get your Flickr, Yelp, and Digg updates when they > e-mail you. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/d75a0624/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 23 08:00:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 10:00:11 -0600 Subject: [Freeswitch-users] Media got stuck after attended transfer... In-Reply-To: References: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com> <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com> <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com> <65d96fc80911220415v70d0bafbvad56c4fcb4576d8b@mail.gmail.com> Message-ID: <191c3a030911230800t6926fcb8w37d55d0bd794f185@mail.gmail.com> I think that issue has been fixed in trunk re: proxy-mode and resume-media-on-hold On Sun, Nov 22, 2009 at 7:00 AM, Klaus Hochlehnert wrote: > For ?only? sending and receiving that?s true. > > > > But my customer wants 2 things: > > - Using HylaFAX as fax server, as there are a lot of client apps and other > tools > > - Connecting ?real? fax machines using a Linksys/Cisco SPA2102 (as this is > certified by their SIP/ISDN gateway vendor) > > > > So I could really need t38 handling in FS to don?t make things more > complicated as they already are? J > > Proxy mode doesn?t work for me because it gives an error when > resume-media-on-hold is set. > > > > Klaus > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir > Culjaga > *Sent:* Sunday, November 22, 2009 1:15 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Media got stuck after attended > transfer... > > > > it is better to enhance mod_fax with t.38 support... we have done sometihng > and it is close to be work... > > T. > > On Sat, Nov 21, 2009 at 2:17 AM, Michael Jerris wrote: > > I think a better approach here is to use spandsp. We already have some > groundwork done for this. If you are interested in contributing, please > email consulting at freeswitch.org and we can discuss further. > > > > Mike > > > > On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: > > > > Hi, > > > > one of my customers is willing to contribute for t38 integration. > > > > The basic idea is to connect HylaFAX to FS: > > t38modem <-> FreeSWITCH <-> Media Gateway with t38 support > > All this without media proxy. > > > > Another idea might be to implement t38 origination/termination with a class > 1 modem input/output for use with HylaFAX. > > > > Do you know how much money we need to collect for t38 support? > > How much time is needed for implementing this? > > > > Thanks, Klaus > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, October 16, 2009 2:10 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Media got stuck after attended > transfer... > > > > > > On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga > wrote: > > hi, any clue when can t38 be added? > > > "Eventually." :) Of course, if we could get more to add to the bounty it > might grease the wheels of innovation. > > > http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch > > Of course, I was listening to my A.M radio the other day and they said that > there was this new invention called the Internet that would let people send > documents to each other electronically. Maybe you should look into that. > Next thing you know they'll come up with telephones that people don't have > to plug into the wall and can take with them in the car. ;) > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/420082bc/attachment-0002.html From grimsqueaker13 at gmail.com Mon Nov 23 05:53:03 2009 From: grimsqueaker13 at gmail.com (Daniel Browne) Date: Mon, 23 Nov 2009 15:53:03 +0200 Subject: [Freeswitch-users] mod_ldap Message-ID: I'm new to Freeswitch and I'm looking for some help using mod_ldap to authenticate SIP endpoints to an LDAP database on registration. I have successfully installed the Freeswitch trunk and OpenLDAP 2.4.18 and I have compiled mod_ldap. I have setup a config file for mod_ldap. It is active and I can see its config in the compiled Freeswitch config file. I have turned on SIP traces, sofia logging and console logging in the Freeswitch command line. However, when I register a SIP phone to my Freeswitch server, I see no reference to mod_ldap in any logs. I have not set up the correct schema in ldap yet, so I would expect to see some indication that the module has searched my ldap server and found no useful information. My phone registers correctly and the normal tests (eg. music on hold) work. I am not sure if mod_ldap falls back on the usual xml config files if it fails to find information on the specified ldap server (as mod_xml_curl does), so I can't be sure if it is working or if my phone is simply registering in the normal way. Can anyone give me pointers on where to look next? If I can just get some feedback from the module I should be able to work out what to do. Thanks -- Grimsqueaker "Even a fool, when he holdeth his peace, is counted wise." Proverbs 17:28a -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/ae5e7b70/attachment-0002.html From malay.thakershi at continuityhealth.com Mon Nov 23 08:07:52 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Mon, 23 Nov 2009 10:07:52 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> Message-ID: <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> Ok. I understand that. It would be great if someone can help me figure out: 1. Why mod_flite is not changing to the female voice even though I tried switching all 4 profiles it provides? 2. I would be alright for purchasing Cepstral for its quality. But FS doesn't come with it compiled I guess (it says swift.dll required when I enabled it in the config file). I asked Cepstral support but they say I have to purchase their SDK (no trial available) even though I just need it to compile it with FS. I understand I will be purchase the voices but how can I get Cepstral DLLs without purchasing the SDK. Thank you for help. Malay Thakershi From: Brian West [mailto:brian at freeswitch.org] Sent: Friday, November 20, 2009 5:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles You pay top dollar for it. The free stuff just isn't as good as what you PAY good money for. I don't expect that to change anytime soon. /b On Nov 20, 2009, at 5:18 PM, Malay Thakershi wrote: Also, can someone tell me what is the best way to get TTS going with good quality? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/7b9b73c8/attachment-0002.html From brian at freeswitch.org Mon Nov 23 08:23:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 10:23:27 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> Message-ID: If you're on linux the SDK comes with the voices. /b On Nov 23, 2009, at 10:07 AM, Malay Thakershi wrote: > Ok. I understand that. > > It would be great if someone can help me figure out: > 1. Why mod_flite is not changing to the female voice even > though I tried switching all 4 profiles it provides? > 2. I would be alright for purchasing Cepstral for its quality. > But FS doesn?t come with it compiled I guess (it says swift.dll > required when I enabled it in the config file). I asked Cepstral > support but they say I have to purchase their SDK (no trial > available) even though I just need it to compile it with FS. I > understand I will be purchase the voices but how can I get Cepstral > DLLs without purchasing the SDK. > > Thank you for help. > > > Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/869f3ee4/attachment-0002.html From mike at jerris.com Mon Nov 23 08:28:19 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 11:28:19 -0500 Subject: [Freeswitch-users] FS compile error under Windows: error LNK2019 In-Reply-To: References: Message-ID: It sounds like the platform sdk is set up wrong. This used to be a problem with older versions of express edition. Double check that your compiler works at all with anything else. Mike On Nov 22, 2009, at 11:51 PM, ??? wrote: > All, > > I tried to compile FS source code under Windows while there are lots of errors: > > Error LNK2019, external _imp_sleep at 4 can not be resolved, this function was referred by _tMCRTStartup. > > Some other more similiar errors detail information attached. > > Any ideas? > > Thanks > Daniel Zeng -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/5a410502/attachment-0002.html From nameer.kazzaz at gmail.com Mon Nov 23 08:35:26 2009 From: nameer.kazzaz at gmail.com (Nameer Kazzaz) Date: Mon, 23 Nov 2009 16:35:26 +0000 Subject: [Freeswitch-users] SIP Digest nonce (stale="true") In-Reply-To: <191c3a030911230752r70b702b1g61694350f56b01e0@mail.gmail.com> References: <4B0A7EBB.8040702@gmail.com> <191c3a030911230752r70b702b1g61694350f56b01e0@mail.gmail.com> Message-ID: <4B0AB9CE.5040300@gmail.com> Hey Anthony, Thanks for the quick response. No the device is a OneAccess so they are saying 'no quotes is the standard'. Thanks Nameer Anthony Minessale wrote: > The quoted true is the correct way from my research. The commented > line was to test a device, a grandstream, they apparently do not > accept it with quotes and I was using the unquoted version it to > gather evidence to issue a bug report to them. They told me it will > be fixed in the next firmware, was this the brand of device you have > as well? > > > > On Mon, Nov 23, 2009 at 6:23 AM, Nameer Kazzaz > > wrote: > > Hi Anthony, > I'm having an issue with a gateway after the nonce-ttl expires we > are sending stale="true", the cpe some how only likes stale=true > without > the "". I see on rev 15441 > > you made a change and marked it out. So my question is who is > correct on > this is it the CPE or are we sticking with the quoted ("true"). > > Thanks > Nameer Kazzaz > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Mon Nov 23 08:38:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 10:38:21 -0600 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <855e4dcf0911221543o222bef63t1c3340b0a41d57c1@mail.gmail.com> References: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> <855e4dcf0911221543o222bef63t1c3340b0a41d57c1@mail.gmail.com> Message-ID: <191c3a030911230838l103bc466p7582c1d05730f61a@mail.gmail.com> There is a formula to implement caching but it's very complicated and nobody has had time to work on it. You have to take every single input variable into account when caching because who is calling the extension, why they are calling it when they are calling it all make a difference. Web servers are designed to get thousands of hits per second so typically they can handle delivering custom xml instruction quite well. If you do not require such a dynamic setup, you could generate static files instead. On Sun, Nov 22, 2009 at 5:43 PM, Tim Uckun wrote: > On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman wrote: > > Hi Sam, > > Take a look at mod_xml_curl. Pretty sure it'll do everything you're > looking > > for. > > > Looking at that diagram it seems like mod_xml_curl makes a call for > every SIP connection. That seems like overkill. Is there a way to set > it up so that it caches the XML it got for a period of time? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/1b3337c8/attachment-0002.html From abeka at greatiam.com Mon Nov 23 08:46:07 2009 From: abeka at greatiam.com (Otis) Date: Mon, 23 Nov 2009 16:46:07 +0000 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: <4B097A89.2050400@greatiam.com> References: <4B086689.6080804@greatiam.com> <4B097A89.2050400@greatiam.com> Message-ID: <4B0ABC4F.1010103@greatiam.com> Hello Could anyone point out what I have missed please ? At the moment I configured a gateway voipuser as described here : Any suggestion as to what path I can take will be highly welcome Thanks . Sam Abekah-Mensah wrote: >
Hi Michael > > Thanks > > I had set it to send incoming calls to extension 1001. This is in the > file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory. > The contents are : > > > > > > > > > Is there > anything wrong with this please ? > > Thanks > > > > Michal Bielicki wrote: >> >> Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah: >> >>> >>> I need help as I cannot receive calls through VOIPUSER. This is a >>> learning setup Attached are my conf files. What is wrong with them ? >>> When I dial from a landline I get a continuous beep. >>> >>> Attached are my gateway and the conf file to transfer. Sopfia Status >>> is my screen message. I can see a FAIL and cannot make head or tail >>> of all that message. Hopefully anyone using voipuser or in fact any >>> of you clever folks can make sense of this. >>> >>> Thanks for your time. >>> >>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy >>> RFC2833 Mode! >>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec >>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>> 2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec >>> Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] >>> 2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec >>> sofia/external/nobody at 213.166.5.133 PCMA/8000 20 ms 160 samples >>> 2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf >>> payload to 101 >>> 2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 >>> (sofia/external/nobody at 213.166.5.133) State Change CS_NEW -> CS_INIT >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_INIT >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>> (sofia/external/nobody at 213.166.5.133) State INIT >>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 >>> sofia/external/nobody at 213.166.5.133 SOFIA INIT >>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 >>> (sofia/external/nobody at 213.166.5.133) State Change CS_INIT -> >>> CS_ROUTING >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>> (sofia/external/nobody at 213.166.5.133) State INIT going to sleep >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_ROUTING >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 >>> (sofia/external/nobody at 213.166.5.133) State ROUTING >>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 >>> sofia/external/nobody at 213.166.5.133 SOFIA ROUTING >>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 >>> sofia/external/nobody at 213.166.5.133 Standard ROUTING >>> 2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing >>> anonymous->abeka in context public >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->unloop] continue=false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (PASS) [unloop] >>> ${unroll_loops}(true) =~ /^true$/ break=on-false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [unloop] >>> ${sip_looped_call}() =~ /^true$/ break=on-false >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->outside_call] continue=true >>> Dialplan: sofia/external/nobody at 213.166.5.133 Absolute Condition >>> [outside_call] >>> Dialplan: sofia/external/nobody at 213.166.5.133 Action >>> set(outside_call=true) >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->call_debug] continue=true >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->public_extensions] continue=false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>> [public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/ >>> break=on-false >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->public_did] continue=false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>> [public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->sip at sip.voipuser.org] continue=false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>> [sip at sip.voipuser.org] destination_number(abeka) =~ /08715042951/ >>> break=on-false >>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>> [public->Inbound-abeka at sip.voipuser.org]] continue=false >>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>> [Inbound-abeka at sip.voipuser.org]] destination_number(abeka) =~ >>> /[08444846450]/ break=on-false >>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 >>> (sofia/external/nobody at 213.166.5.133) State Change CS_ROUTING -> >>> CS_EXECUTE >>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 >>> (sofia/external/nobody at 213.166.5.133) State ROUTING going to sleep >>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_EXECUTE >>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 >>> (sofia/external/nobody at 213.166.5.133) State EXECUTE >>> 2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 >>> sofia/external/nobody at 213.166.5.133 SOFIA EXECUTE >>> 2009-11-21 22:07:15.706658 [DEBUG] switch_core_state_machine.c:151 >>> sofia/external/nobody at 213.166.5.133 Standard EXECUTE >>> EXECUTE sofia/external/nobody at 213.166.5.133 set(outside_call=true) >>> 2009-11-21 22:07:15.728613 [DEBUG] mod_dptools.c:748 >>> sofia/external/nobody at 213.166.5.133 SET [outside_call]=[true] >>> 2009-11-21 22:07:15.728613 [NOTICE] switch_core_state_machine.c:179 >>> Hangup sofia/external/nobody at 213.166.5.133 [CS_EXECUTE] >>> [NORMAL_CLEARING] >>> 2009-11-21 22:07:15.728613 [DEBUG] switch_channel.c:1683 Send signal >>> sofia/external/nobody at 213.166.5.133 [KILL] >>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:491 >>> (sofia/external/nobody at 213.166.5.133) State EXECUTE going to sleep >>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_HANGUP >>> 2009-11-21 22:07:15.735830 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/external/nobody at 213.166.5.133) State HANGUP >>> 2009-11-21 22:07:15.735830 [DEBUG] mod_sofia.c:338 Channel >>> sofia/external/nobody at 213.166.5.133 hanging up, cause: NORMAL_CLEARING >>> 2009-11-21 22:07:15.737680 [DEBUG] mod_sofia.c:417 Responding to >>> INVITE with: 480 >>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:46 >>> sofia/external/nobody at 213.166.5.133 Standard HANGUP, cause: >>> NORMAL_CLEARING >>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/external/nobody at 213.166.5.133) State HANGUP going to sleep >>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_state_machine.c:476 >>> (sofia/external/nobody at 213.166.5.133) State Change CS_HANGUP -> >>> CS_REPORTING >>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_REPORTING >>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:612 >>> (sofia/external/nobody at 213.166.5.133) State REPORTING >>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:53 >>> sofia/external/nobody at 213.166.5.133 Standard REPORTING, cause: >>> NORMAL_CLEARING >>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:612 >>> (sofia/external/nobody at 213.166.5.133) State REPORTING going to sleep >>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:411 >>> (sofia/external/nobody at 213.166.5.133) State Change CS_REPORTING -> >>> CS_DESTROY >>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_session.c:1068 >>> Session 2 (sofia/external/nobody at 213.166.5.133) Locked, Waiting on >>> external entities >>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1086 >>> Session 2 (sofia/external/nobody at 213.166.5.133) Ended >>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1088 Close >>> Channel sofia/external/nobody at 213.166.5.133 [CS_DESTROY] >>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>> (sofia/external/nobody at 213.166.5.133) State DESTROY >>> 2009-11-21 22:07:15.802636 [DEBUG] mod_sofia.c:255 >>> sofia/external/nobody at 213.166.5.133 SOFIA DESTROY >>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:60 >>> sofia/external/nobody at 213.166.5.133 Standard DESTROY >>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>> (sofia/external/nobody at 213.166.5.133) State DESTROY going to sleep >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> : >> >> >> you seem to have not specified an extension where the call should go to >> my voipuser.org setup looks like: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> I am also surprised that your setup works with a from-domain of >> sip.voipuser.org >> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> *Michal Bielicki* >> HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa >> t. +48228753290 | f. +48228753291 michal.bielicki at halokwadrat.pl >> | w. >> www.halokwadrat.pl >> >> >> >> *Knowledge & Low Prices. Guaranteed!* >> > > > >
From egable+freeswitch at gmail.com Mon Nov 23 08:48:37 2009 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Mon, 23 Nov 2009 11:48:37 -0500 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: <191c3a030911230838l103bc466p7582c1d05730f61a@mail.gmail.com> References: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> <855e4dcf0911221543o222bef63t1c3340b0a41d57c1@mail.gmail.com> <191c3a030911230838l103bc466p7582c1d05730f61a@mail.gmail.com> Message-ID: Or, you can use something like Smarty to cache your generated XML on your web server and only invalidate those cached results when you change something that will impact them. On Mon, Nov 23, 2009 at 11:38 AM, Anthony Minessale wrote: > There is a formula to implement caching but it's very complicated and nobody > has had time to work on it. > You have to take every single input variable into account when caching > because who is calling the extension, why they are calling it when they are > calling it all make a difference. > > Web servers are designed to get thousands of hits per second so typically > they can handle delivering custom xml instruction quite well. > > If you do not require such a dynamic setup, you could generate static files > instead. > > > On Sun, Nov 22, 2009 at 5:43 PM, Tim Uckun wrote: >> >> On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman wrote: >> > Hi Sam, >> > Take a look at mod_xml_curl. ?Pretty sure it'll do everything you're >> > looking >> > for. >> >> >> Looking at that diagram it seems like mod_xml_curl makes a call for >> every SIP connection. That seems like overkill. ?Is there a way to set >> it up so that it caches the XML it got for a period of time? >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From anthony.minessale at gmail.com Mon Nov 23 08:50:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 10:50:11 -0600 Subject: [Freeswitch-users] SIP Digest nonce (stale="true") In-Reply-To: <4B0AB9CE.5040300@gmail.com> References: <4B0A7EBB.8040702@gmail.com> <191c3a030911230752r70b702b1g61694350f56b01e0@mail.gmail.com> <4B0AB9CE.5040300@gmail.com> Message-ID: <191c3a030911230850g5fdf5558wfb53237f3179b52b@mail.gmail.com> Tell you what, I don't have the patience for it, i'm sure most stuff does it either way and I'm sure nobody insists you have them so I will take them out so I can have some peace. On Mon, Nov 23, 2009 at 10:35 AM, Nameer Kazzaz wrote: > Hey Anthony, > Thanks for the quick response. No the device is a OneAccess so they > are saying 'no quotes is the standard'. > > Thanks > Nameer > > Anthony Minessale wrote: > > The quoted true is the correct way from my research. The commented > > line was to test a device, a grandstream, they apparently do not > > accept it with quotes and I was using the unquoted version it to > > gather evidence to issue a bug report to them. They told me it will > > be fixed in the next firmware, was this the brand of device you have > > as well? > > > > > > > > On Mon, Nov 23, 2009 at 6:23 AM, Nameer Kazzaz > > > wrote: > > > > Hi Anthony, > > I'm having an issue with a gateway after the nonce-ttl expires we > > are sending stale="true", the cpe some how only likes stale=true > > without > > the "". I see on rev 15441 > > < > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_reg.c?r=15441#l687 > > > > you made a change and marked it out. So my question is who is > > correct on > > this is it the CPE or are we sticking with the quoted ("true"). > > > > Thanks > > Nameer Kazzaz > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/6dcbf497/attachment-0002.html From jlenk at frontiernet.net Mon Nov 23 09:01:25 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Mon, 23 Nov 2009 09:01:25 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <858430.90192.qm@web111301.mail.gq1.yahoo.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> <1258949788572-4048969.post@n2.nabble.com> <858430.90192.qm@web111301.mail.gq1.yahoo.com> Message-ID: <1258995685201-4052409.post@n2.nabble.com> Hi Arsen, I have merged your changes in now - thank you. Would you perhaps be able to look at the x64 changes I made to the projects and merge them back into your code to ease the future updating. Thanks Jeff Arsen Chaloyan wrote: > > Hi Jeff, > > > Your input would be very helpful, I just wanted to understand where the > problem is and contribute the way I can. > I see you're the assignee, so please go ahead and let me know if there is > anything left I can help with. > > Arsen. > > > > ________________________________ > From: Jeff Lenk > To: freeswitch-users at lists.freeswitch.org > Sent: Mon, November 23, 2009 8:16:28 AM > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > Hi Arsen, > > I would be happy to help with the FS integration if you want - please do > put your patch in a Jira. > > Jeff > > ________________________________ > Date: Sun, 22 Nov 2009 10:09:41 -0800 > From: [hidden email] > To: [hidden email] > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > > We discussed build integration related issues a few months ago with Mike > and seemed to find a solution which would work for both UniMRCP and > FreeSWITCH source trees. > > Now I've just got a chance to look into this a bit closer trying to > further complete VS2008 build integration in FreeSWITCH. So I've got it > working, the module is not only being built, but also is getting loaded. > Current build integration is not as seamless as I want it to be, but > probably we can start with what we have now and then discuss and identify > what can be done in the future. This concerns not only build integration > but overall integrity. > > So would you be interested in the patch? Where should I upload it? > I thought I had a Jira account, but not sure it exists any more. > > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org > > > > > > ________________________________ > From: Jeff Lenk <[hidden email]> > To: [hidden email] > Sent: Fri, November 20, 2009 7:59:28 PM > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > > That module is not currently being built for Windows. Also the library > unimrcp needs build integration work with FS to make that happen under > windows. > > > ss1 wrote: > >> >> Hi Everyone, >> >> Please help freeswitch experts... !!! >> >> i have been working on freeswitch from last 2 days. i have downloaded >> freeswitch and unimrcp (server + client) for windows. >> I tested the unimrcp client and server, which is running fine with the >> command: run synth and run recog. I got both synth.pcm & recog.pcm files. >> >> But my objective is to call Freeswitch through x-lite, where freeswitch >> should call unimrcp client and return the PCM files. >> >> I tried it alot, but unable to do it. after lots of reading i found that >> i >> do not have mod_unimrcp. i do not know from where to download it and how >> to merge it into freeswitch. >> >> I would be very thankful if you may help. >> >> Thanks, >> ss >> >> -- > View this message in context: > http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > > View message @ > http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html > To unsubscribe from Re: need help !! Problem with freeswitch & uniMRCP, > click here. > > ________________________________ > Hotmail: Trusted email with powerful SPAM protection. Sign up now. > ________________________________ > View this message in context: RE: [Freeswitch-users] need help !! Problem > with freeswitch & uniMRCP > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4052409.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jonas.gauffin at gmail.com Mon Nov 23 09:08:44 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Mon, 23 Nov 2009 18:08:44 +0100 Subject: [Freeswitch-users] NAT problem Message-ID: Hello I got the following setup: Phones -> FreeSwitch -> NAT -> Internet -> Gateway And I'm struggling to get NAT working properly. I'm running freeswitch with the "-nonat" option and have tried different ext-rtp-ip/ext-sip-ip combinations in external/internal profiles. The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly. Currently I have this setup: API CALL [sofia(status profile external)] output: ======================================================== Name external Domain Name N/A Context public Challenge Realm auto_to RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 Ext-SIP-IP 85.89.XX.XX OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false API CALL [sofia(status profile default)] output: ======================================================== Name default Domain Name N/A Alias Of internal Context public Challenge Realm auto_from RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED false STUN-AUTO-DISABLE false Sample phone registration: Call-ID: Xmbw9PyQ5Q6L2MnQ at 192.168.1.121 User: u1000009 at default Contact: "u1000009" Agent: IP PHONE 3 V1.58.004 CFG0 Status: Registered(UDP)(unknown) EXP(2009-11-23 19:26:40) Host: jonas-PC IP: 192.168.1.121 Port: 6094 Auth-User: u1000009 Auth-Realm: default MWI-Account: u1000009 at default Outbound INVITE: send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000: ------------------------------------------------------------------------ INVITE sip:0706930XXX at sipgw2.XXXXX.se SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp Max-Forwards: 69 From: "Kundtj??nst Arne" ;tag=B7pve7F6eeH7c To: > Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 CSeq: 123379614 INVITE Contact: Call-Info: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 293 X-FS-Support: update_display Remote-Party-ID: "Kundtj??nst Arne" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110 s=FreeSWITCH c=IN IP4 192.168.1.110 t=0 0 m=audio 24986 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Many thanks, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/3ca2dcf5/attachment-0002.html From brian at freeswitch.org Mon Nov 23 09:21:47 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 11:21:47 -0600 Subject: [Freeswitch-users] NAT problem In-Reply-To: References: Message-ID: You set the ext-rtp-ip on the profile the phones talk too... but you shouldn't be doing that. /b On Nov 23, 2009, at 11:08 AM, Jonas Gauffin wrote: > Hello > > I got the following setup: Phones -> FreeSwitch -> NAT -> Internet - > > Gateway > > And I'm struggling to get NAT working properly. I'm running > freeswitch with the "-nonat" option and have tried different ext-rtp- > ip/ext-sip-ip combinations in external/internal profiles. > The From header seems to be correct while contact header and SDP > uses local ip? Please help me configure everything correctly. > > Currently I have this setup: From jonas.gauffin at gmail.com Mon Nov 23 09:24:18 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Mon, 23 Nov 2009 18:24:18 +0100 Subject: [Freeswitch-users] NAT problem In-Reply-To: References: Message-ID: Ok. Found the problem. I had started using "sofia/outbound/ XXXXXX at sipgw2.XXXX.se" as bridge destination to try to get outbound_caller_id_name/outbound_caller_id_number working. It works if I use the correct profile name, "sofia/internal/ XXXXXX at sipgw2.XXXX.se" When do FS use outbound_caller_id instead of effective_caller_id? On Mon, Nov 23, 2009 at 6:08 PM, Jonas Gauffin wrote: > Hello > > I got the following setup: Phones -> FreeSwitch -> NAT -> Internet -> > Gateway > > And I'm struggling to get NAT working properly. I'm running freeswitch with > the "-nonat" option and have tried different ext-rtp-ip/ext-sip-ip > combinations in external/internal profiles. > The From header seems to be correct while contact header and SDP uses local > ip? Please help me configure everything correctly. > > Currently I have this setup: > > API CALL [sofia(status profile external)] output: > ======================================================== > Name external > Domain Name N/A > Context public > Challenge Realm auto_to > RTP-IP 192.168.1.110 > Ext-RTP-IP 85.89.XX.XX > SIP-IP 192.168.1.110 > Ext-SIP-IP 85.89.XX.XX > OUTBOUND-PROXY N/A > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > > API CALL [sofia(status profile default)] output: > ======================================================== > Name default > Domain Name N/A > Alias Of internal > Context public > Challenge Realm auto_from > RTP-IP 192.168.1.110 > Ext-RTP-IP 85.89.XX.XX > SIP-IP 192.168.1.110 > OUTBOUND-PROXY N/A > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED false > STUN-AUTO-DISABLE false > > Sample phone registration: > Call-ID: Xmbw9PyQ5Q6L2MnQ at 192.168.1.121 > User: u1000009 at default > Contact: "u1000009" > Agent: IP PHONE 3 V1.58.004 CFG0 > Status: Registered(UDP)(unknown) EXP(2009-11-23 19:26:40) > Host: jonas-PC > IP: 192.168.1.121 > Port: 6094 > Auth-User: u1000009 > Auth-Realm: default > MWI-Account: u1000009 at default > > Outbound INVITE: > send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000: > ------------------------------------------------------------------------ > INVITE sip:0706930XXX at sipgw2.XXXXX.seSIP/2.0 > Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp > Max-Forwards: 69 > From: "Kundtj??nst Arne" ;tag=B7pve7F6eeH7c > To: > > Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 > CSeq: 123379614 INVITE > Contact: > Call-Info: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 293 > X-FS-Support: update_display > Remote-Party-ID: "Kundtj??nst Arne" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110 > s=FreeSWITCH > c=IN IP4 192.168.1.110 > t=0 0 > m=audio 24986 RTP/AVP 0 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > Many thanks, > Jonas > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/686b9f8a/attachment-0002.html From msc at freeswitch.org Mon Nov 23 09:27:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 09:27:01 -0800 Subject: [Freeswitch-users] IP0010 SIP Phone In-Reply-To: <10222602.1258898966890.JavaMail.root@whwamui-deputy.pas.sa.earthlink.net> References: <10222602.1258898966890.JavaMail.root@whwamui-deputy.pas.sa.earthlink.net> Message-ID: <87f2f3b90911230927hc286e28gac6578ce5c8432c2@mail.gmail.com> On Sun, Nov 22, 2009 at 6:09 AM, David V. Fansler wrote: > After the help of a couple of people from this list, I now have FreeSWITCH > running - yeah! I have installed X-Lite on a couple of computers and they > dial each other, play music on hold, etc. I have not yet connected to the > outside world. > > I purchased an IP-0010 phone off eBay ($20 including shipping - docs at > http://www.vanaccess.com/news/news_images/2007131_73_User%20Manual%20-%20IP0010.pdf) > I cannot get this phone to work with the system. It gets an IP address, > time/date, and a dial tone. After many tries with the http congifuration > tool, I got the phone "configured" with the address of the SIP server, and a > SIP User ID. When you dial an extension the FreeSWITCH window shows the > following: > > sofica.c3844 Hanugup sofia/internal/101 at 192.168.1.165 [CS_NEW] > [INCOMPATIBLE_DESTINATION] > switch_core_session.c1139 Session 20 (sofia/internal/101 at 192.165.1.65) > Ended > switch_core_session.c1141 Close Channel sofia/internal/1001 at 192.168.1.165[CS_DESTROY] > > Has anyone else tried this phone, or does anyone have suggestions I could > try. I have looked through the website but have not found anything to help. > > Thanks, > > David > > David, Time to do a little digging. First off, review this wiki page on reporting bugs - it has lots of useful information on how to gather information from your system and report it to the community: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd recommend that you get a debug log and a sip trace and post it to pastebin. Report back the pastebin URL here in this thread and we'll have a look. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/e8e61035/attachment-0002.html From brian at freeswitch.org Mon Nov 23 09:31:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 11:31:02 -0600 Subject: [Freeswitch-users] NAT problem In-Reply-To: References: Message-ID: <56195859-FC9B-46C1-9ABE-88CCC26B881B@freeswitch.org> outbound_caller_id is a made up variable that is used in the defaults that are used in the examples only. /b On Nov 23, 2009, at 11:24 AM, Jonas Gauffin wrote: > Ok. Found the problem. I had started using "sofia/outbound/XXXXXX at sipgw2.XXXX.se > " as bridge destination to try to get outbound_caller_id_name/ > outbound_caller_id_number working. > It works if I use the correct profile name, "sofia/internal/XXXXXX at sipgw2.XXXX.se > " > > When do FS use outbound_caller_id instead of effective_caller_id? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/afed17c5/attachment-0002.html From tleyden at branchcut.com Mon Nov 23 09:47:33 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Mon, 23 Nov 2009 22:17:33 +0430 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <4B0AA8B6.2080305@greatiam.com> References: <4B0AA8B6.2080305@greatiam.com> Message-ID: Yeah a kind user (Innotel) took the time to write up Cent OS installation instructions for wikipbx and posted it to the wiki: http://wikipbx.subwiki.com/forum/t-115012/freeswitch-svn-1-0-2-wikipbx-svn-61-centos-5-1-installation-instructions If you have any problems please post in the forum: http://wikipbx.subwiki.com/forum:start On Mon, Nov 23, 2009 at 7:52 PM, Otis wrote: > Hi Folks > > Is anyone using this on Fedora and is there a binary or installation > script anywhere > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/6756706d/attachment-0002.html From jonas.gauffin at gmail.com Mon Nov 23 09:50:07 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Mon, 23 Nov 2009 18:50:07 +0100 Subject: [Freeswitch-users] NAT problem In-Reply-To: <56195859-FC9B-46C1-9ABE-88CCC26B881B@freeswitch.org> References: <56195859-FC9B-46C1-9ABE-88CCC26B881B@freeswitch.org> Message-ID: Ok. It would be a nice feature if outbound_caller_id was used by freeswitch. I do quite often bridge to both internal and external destinations in the same bridge command (as in "sofia/internal/5530,sofia/internal/ 070123456 at sipgw2.XXXX.se). This forces me to always use complete phone numbers in the caller id since my gateway would reject the call otherwise. It would be really neat if FS could use effective_caller_id (5531) for the internal bridge and outbound_caller_id (+4681235531) for the external bridge. On Mon, Nov 23, 2009 at 6:31 PM, Brian West wrote: > outbound_caller_id is a made up variable that is used in the defaults that > are used in the examples only. > > /b > > On Nov 23, 2009, at 11:24 AM, Jonas Gauffin wrote: > > Ok. Found the problem. I had started using "sofia/outbound/ > XXXXXX at sipgw2.XXXX.se" as bridge destination to try to get > outbound_caller_id_name/outbound_caller_id_number working. > It works if I use the correct profile name, "sofia/internal/ > XXXXXX at sipgw2.XXXX.se" > > When do FS use outbound_caller_id instead of effective_caller_id? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/32c8b040/attachment-0002.html From msc at freeswitch.org Mon Nov 23 09:51:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 09:51:50 -0800 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> Message-ID: <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy wrote: > Hi, > I'm using perl ESL to control the call in freeswitch. > I'm having the following scenario, but not able to get it right. > > Dialplan: > > > > > > > > > 1. User A calls to an extention (1000). > 2. My ESL program will be running, and it answers the call. > 3. Then the program will get a number from the user. > 4. It will hangup the call. > 5. The program has to call to the number that was given by the user. > > In the above scenario, I was able to do until the 4th step. After hangup > the call, if I say originate it is not working. > Any ideas on how to do this in ESL. > > I want to make sure I understand what the script is supposed to be doing. The caller will key in a phone number to your script and your script will collect those digits. The script will then hangup on the caller and originate a completely new call? Perhaps you could use sched_api to schedule a new originate command for a few seconds into the future and then hangup? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/8cc94bfe/attachment-0002.html From brian at freeswitch.org Mon Nov 23 09:52:31 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 11:52:31 -0600 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: References: <4B0AA8B6.2080305@greatiam.com> Message-ID: s/i386/x86_64/ if you are 64bit /b On Nov 23, 2009, at 11:47 AM, Traun Leyden wrote: > > Yeah a kind user (Innotel) took the time to write up Cent OS > installation instructions for wikipbx and posted it to the wiki: > > http://wikipbx.subwiki.com/forum/t-115012/freeswitch-svn-1-0-2-wikipbx-svn-61-centos-5-1-installation-instructions > > If you have any problems please post in the forum: http://wikipbx.subwiki.com/forum:start > > On Mon, Nov 23, 2009 at 7:52 PM, Otis wrote: > Hi Folks > > Is anyone using this on Fedora and is there a binary or installation > script anywhere > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/6384852d/attachment-0002.html From robert.hadley at teotech.com Mon Nov 23 09:53:16 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Mon, 23 Nov 2009 09:53:16 -0800 Subject: [Freeswitch-users] Building in a builddir using --srcdir option but modules still build in srcdir Message-ID: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? Thanks, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/c23bed4e/attachment-0002.html From brian at freeswitch.org Mon Nov 23 09:56:17 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 11:56:17 -0600 Subject: [Freeswitch-users] NAT problem In-Reply-To: References: <56195859-FC9B-46C1-9ABE-88CCC26B881B@freeswitch.org> Message-ID: See default config it lsets you do that. Use the variables to store two versions of the callerid then set it depending on if its outside or inside... its rather easy to do. /b On Nov 23, 2009, at 11:50 AM, Jonas Gauffin wrote: > Ok. It would be a nice feature if outbound_caller_id was used by > freeswitch. > I do quite often bridge to both internal and external destinations > in the same bridge command (as in "sofia/internal/5530,sofia/internal/070123456 at sipgw2.XXXX.se > ). This forces me to always use complete phone numbers in the caller > id since my gateway would reject the call otherwise. > > It would be really neat if FS could use effective_caller_id (5531) > for the internal bridge and outbound_caller_id (+4681235531) for the > external bridge. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/ba55b893/attachment-0002.html From malay.thakershi at continuityhealth.com Mon Nov 23 10:09:02 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Mon, 23 Nov 2009 12:09:02 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> Message-ID: <010701ca6c68$0950f6a0$1bf2e3e0$@thakershi@continuityhealth.com> I am not using Linux. I am using Windows 2008 server. Malay Thakershi From: Brian West [mailto:brian at freeswitch.org] Sent: Monday, November 23, 2009 10:23 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles If you're on linux the SDK comes with the voices. /b On Nov 23, 2009, at 10:07 AM, Malay Thakershi wrote: Ok. I understand that. It would be great if someone can help me figure out: 1. Why mod_flite is not changing to the female voice even though I tried switching all 4 profiles it provides? 2. I would be alright for purchasing Cepstral for its quality. But FS doesn't come with it compiled I guess (it says swift.dll required when I enabled it in the config file). I asked Cepstral support but they say I have to purchase their SDK (no trial available) even though I just need it to compile it with FS. I understand I will be purchase the voices but how can I get Cepstral DLLs without purchasing the SDK. Thank you for help. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/3a588fb3/attachment-0002.html From brian at freeswitch.org Mon Nov 23 10:14:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 12:14:02 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <010701ca6c68$0950f6a0$1bf2e3e0$@thakershi@continuityhealth.com> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> <010701ca6c68$0950f6a0$1bf2e3e0$@thakershi@continuityhealth.com> Message-ID: <57CD54FA-C16E-4F36-9678-6CF1FB2D1A17@freeswitch.org> You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. /b On Nov 23, 2009, at 12:09 PM, Malay Thakershi wrote: > I am not using Linux. I am using Windows 2008 server. > > Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/41ffe00d/attachment-0002.html From william.suffill at gmail.com Mon Nov 23 10:22:16 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 23 Nov 2009 13:22:16 -0500 Subject: [Freeswitch-users] Git Message-ID: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> Just wondering if anyone is keeping an update to date git repo of FreeSwitch? I been using git-svn to keep a copy on my machines but it can be quite time consuming due to the per revision fetching. If there was a repo to clone that would speed up the process considerably. -- W From lon at kickasspixels.com Mon Nov 23 10:36:48 2009 From: lon at kickasspixels.com (Lon Baker) Date: Mon, 23 Nov 2009 10:36:48 -0800 Subject: [Freeswitch-users] Git In-Reply-To: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> References: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> Message-ID: William, Perhaps someone could setup one on github? It's free for open source project. Lon On Nov 23, 2009, at 10:22 AM, William Suffill wrote: > Just wondering if anyone is keeping an update to date git repo of > FreeSwitch? I been using git-svn to keep a copy on my machines but it > can be quite time consuming due to the per revision fetching. If there > was a repo to clone that would speed up the process considerably. > > -- W > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From abeka at greatiam.com Mon Nov 23 10:37:09 2009 From: abeka at greatiam.com (Otis) Date: Mon, 23 Nov 2009 18:37:09 +0000 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: References: <4B0AA8B6.2080305@greatiam.com> Message-ID: <4B0AD655.9070507@greatiam.com> Thanks. I have to get a centos box I guess. Much appreciated Samuel 'Otis' Traun Leyden wrote: > > Yeah a kind user (Innotel) took the time to write up Cent OS > installation instructions for wikipbx and posted it to the wiki: > > http://wikipbx.subwiki.com/forum/t-115012/freeswitch-svn-1-0-2-wikipbx-svn-61-centos-5-1-installation-instructions > > If you have any problems please post in the forum: > http://wikipbx.subwiki.com/forum:start > > On Mon, Nov 23, 2009 at 7:52 PM, Otis > wrote: > > Hi Folks > > Is anyone using this on Fedora and is there a binary or installation > script anywhere > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From malay.thakershi at continuityhealth.com Mon Nov 23 10:41:55 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Mon, 23 Nov 2009 12:41:55 -0600 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <57CD54FA-C16E-4F36-9678-6CF1FB2D1A17@freeswitch.org> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> <010701ca6c68$0950f6a0$1bf2e3e0$@thakershi@continuityhealth.com> <57CD54FA-C16E-4F36-9678-6CF1FB2D1A17@freeswitch.org> Message-ID: <012401ca6c6c$a19673a0$e4c35ae0$@thakershi@continuityhealth.com> Thank you for your responses. I did follow that web link to ask them as instructed but they declined. They asked me where I want to use it. I told them I wanted it to build FreeSwitch so that I can use Cepstral voices (to be purchased from them with it). Their response was they do not provide trial of the SDK. They do not support FreeSwitch. Malay Thakershi From: Brian West [mailto:brian at freeswitch.org] Sent: Monday, November 23, 2009 12:14 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. /b On Nov 23, 2009, at 12:09 PM, Malay Thakershi wrote: I am not using Linux. I am using Windows 2008 server. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/9f6321ab/attachment-0002.html From mike at jerris.com Mon Nov 23 10:43:27 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 13:43:27 -0500 Subject: [Freeswitch-users] Git In-Reply-To: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> References: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> Message-ID: <1336A1DE-ECBA-43EA-968E-7C6BE89A1251@jerris.com> I think this one is kept up to date, but we may re-do this at some point soon, so it may get re-built. http://svn.freeswitch.org/freeswitch.git/ Mike On Nov 23, 2009, at 1:22 PM, William Suffill wrote: > Just wondering if anyone is keeping an update to date git repo of > FreeSwitch? I been using git-svn to keep a copy on my machines but it > can be quite time consuming due to the per revision fetching. If there > was a repo to clone that would speed up the process considerably. From mike at jerris.com Mon Nov 23 10:46:48 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 13:46:48 -0500 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911211523k7998d048nced3af8fb805e770@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <34D3FA11-00E2-4D8A-A5D6-2118F0AEDE2F@freeswitch.org> <2160023e0911122330m55b0128ene07e3b2e8a6553fd@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> <2160023e0911211523k7998d048nced3af8fb805e770@mail.gmail.com> Message-ID: This looks like a nat issue to me, please re-test this against latest svn trunk and if its still not working pastebin a full sip trace and report the link back here. Mike On Nov 21, 2009, at 6:23 PM, RobertT wrote: > Yep, I use proxy media. First it started with 1.0.4 release, then I've updated a week or two ago with the latest svn trunk, not sure what was the rev number. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From william.suffill at gmail.com Mon Nov 23 10:52:06 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 23 Nov 2009 13:52:06 -0500 Subject: [Freeswitch-users] Git In-Reply-To: <1336A1DE-ECBA-43EA-968E-7C6BE89A1251@jerris.com> References: <6b65470d0911231022g29ff49b5j89b1fb390f5fa80f@mail.gmail.com> <1336A1DE-ECBA-43EA-968E-7C6BE89A1251@jerris.com> Message-ID: <6b65470d0911231052q57129ab5n2359c06d327d93d4@mail.gmail.com> I'd rather it be a decision by the community as a whole and authorized. Sure there are ways to have anyone who wants to on their own. Thanks for the insight. -- W From brian at freeswitch.org Mon Nov 23 10:53:30 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 12:53:30 -0600 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: References: <4B0266F4.8070602@savion.huji.ac.il> <4B0387F1.7070105@savion.huji.ac.il> <193640CC-3E62-4248-8E80-CE7FE82653C0@jerris.com> Message-ID: <6293479E-C530-4510-BD4D-592FE3E79D35@freeswitch.org> Because if you dial local-user at local-domain thats not the correct way this will usually trigger a call out and back in on the profile thus moving you one leg away from the actual user. If you're going to do that use sofia_contact and review how the defaults abstract this so you can just call user/xxxx at domain, You need to make sure the presence_id is set like the defaults have it. /b On Nov 22, 2009, at 1:39 AM, Yehavi Bourvine wrote: > Thanks Mike! However, this doesn't fully solve my problem. When > using sofia_contact() indeed it works ok with finding the > destination's profile. However, it breaks the BLFs... > > When calling sofia/sip_profile/local-user%local-domain the BLF works > ok. When calling sofia_contact(sofia/sip_profile/local-user at local- > domain) BLF doesn't work (nothing is sent to the watching phone). > > Any more clues??? > > Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/170c2443/attachment-0002.html From mike at jerris.com Mon Nov 23 10:58:57 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 13:58:57 -0500 Subject: [Freeswitch-users] Adding headers to INFO messages for Advice of Charge on SNOM In-Reply-To: <1258963204.4961.8.camel@keithl-lt> References: <1258963204.4961.8.camel@keithl-lt> Message-ID: <914F7A85-3229-469E-92A2-FD9664FCC03D@jerris.com> Is there any rfc on this or is it something that snom just made up? On Nov 23, 2009, at 3:00 AM, Keith Laaks wrote: > > > > Hi, > > I have tried maintaining charging information on a SNOM 300's display > using 'display' - but found that the phone has some timer, whereby every > 60 seconds it wipes out whatever happens to be on the display at that > time and replaces is with the dialled number. So not a viable option as > it impacts usability. Really annoying when the display was just updated > with valuable information for the user and a split second later it gets > replaced. > [If somebody knows how to disable this behaviour - please do tell...] > > I see that SNOM supports a number of features for Advice of Charge. > >> From their Wiki: > > http://wiki.snom.com/Advice_of_charge_%28AOC%29_in_SIP > Example of an SIP-Info Message: > > ----------------------------------------------------- > INFO sip:bla at snom.com SIP/2.0 > From: ;tag=5354n3 > To: ;tag=33rfh3 > CSeq: 23423 INFO > Call-ID: 3452tw43dt354dm03 > AOC: charging;state=active; > charging-info=currency; > currency=EUR; > amount=2000; > multiplier=0.001 > Content-Length: 0 > ----------------------------------------------------- > > So the question - Is there some method available today to add these additional > 'new' headers to an INFO message I can send out to these phones? > > If not, I guess it's a matter of looking at enhancing the "case SWITCH_MESSAGE_INDICATE_DISPLAY" section in mod_sofia.c ? > From mike at jerris.com Mon Nov 23 11:01:24 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:01:24 -0500 Subject: [Freeswitch-users] User who answer the bridge in a execute_answer In-Reply-To: <4B0A65C9.10509@daccii.it> References: <4B0A65C9.10509@daccii.it> Message-ID: <9133578A-F706-46C2-9653-6C22D6E056CB@jerris.com> Try running the info app there and see if the info is anywhere in that output . Mike On Nov 23, 2009, at 5:36 AM, Albano Daniele Salvatore - Lavoro wrote: > Hi, > > i'm writing some dialplan parts that get executed on execute_on_answer. In this dialplan that get executed i need to make a directory to handle recordings for record_session and my folder structure is: > USER/YEAR/MONTH/HOUR-MINUTE-SECOND-CALLER_NUMBER.wav > > ------ > > > ------ > > The call flow is: > Call from external -> IVR -> Transfer to Group -> Execute on Answer -> system/bind_meta_app > > > Pratically, i need the number (or better the user) that answered the call: what variable should i check? > > I tried with sip_from_user, callee_id_number and some other. > > > Thank for your help, > > Best Regards, > Daniele > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Mon Nov 23 11:02:11 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:02:11 -0500 Subject: [Freeswitch-users] Execute on Answer with JavaScript In-Reply-To: <26476532.post@talk.nabble.com> References: <26476532.post@talk.nabble.com> Message-ID: This is done automatically when you bridge 2 sessions together. Mike On Nov 23, 2009, at 6:45 AM, Oscav wrote: > How can we send the answer to the caller only when the callee answers, in > JavaScript?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/28a00622/attachment-0002.html From mike at jerris.com Mon Nov 23 11:03:34 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:03:34 -0500 Subject: [Freeswitch-users] Question about rtp-timeout-sec variable In-Reply-To: <1258977639954-4050650.post@n2.nabble.com> References: <1258977639954-4050650.post@n2.nabble.com> Message-ID: Take a look at a pcap of the traffic, I suspect the other side still has media flowing. On Nov 23, 2009, at 7:00 AM, Maciej Aniserowicz wrote: > > Hello, > I have 2 instances of FS: one controlled by my application (making calls > with TCP commands, recording sessions, listening to events etc) and one > acting as a remote gateway to which all users register. When I leave the > default values of rtp-timeout-sec and brutally kill x-lite during > conversation, the 'hangup' event with 'media_timeout' cause is obviously > sent after the default 5 minutes (and until then, the other leg is still > connected to a 'dead' channel). > The question is: which FS instance is responsible for terminating the > connection after timeout? Only the 'remote' FS instance config seems to > work. I thought that the shortest configured value should cause the timeout, > but it's not the case. Am I missing something, or is this the correct > behavior? From mike at jerris.com Mon Nov 23 11:05:06 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:05:06 -0500 Subject: [Freeswitch-users] Using odbc in FS core In-Reply-To: <1202092411.20091123134528@yes.net.ua> References: <1382216794.20091121134106@yes.net.ua> <1013085378.20091121140207@yes.net.ua> <191c3a030911210814l6e50b883uba61815fcd36afe1@mail.gmail.com> <1202092411.20091123134528@yes.net.ua> Message-ID: Yes please On Nov 23, 2009, at 6:45 AM, Mike Tkachuk wrote: > Hello Anthony, > > Is clear, thanks, I'll test and will let you know. > Should I add 'core-db-dsn' parameter description to Wiki? Maybe we need to add this parameter also to sample conf files? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/0ff063e4/attachment-0002.html From mike at jerris.com Mon Nov 23 11:16:18 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:16:18 -0500 Subject: [Freeswitch-users] Building in a builddir using --srcdir option but modules still build in srcdir In-Reply-To: References: Message-ID: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com> The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: > I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/b1038920/attachment-0002.html From christian.loeschenkohl at xpirio.com Mon Nov 23 11:17:53 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 23 Nov 2009 20:17:53 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing Message-ID: <4B0ADFE1.4070506@xpirio.com> hi our freeswitch server has to talk to a sonus ip-switch when we want to setup a call we do get a "100 Trying" and then a "180 Ringing" within the "180 Ringing" we get a sdp with "a=sendonly" then our freeswitch quits with a CANCEL message. i simply don't get why our freeswitch aborts the session - i think it would work if no "a=sendonly" would be present in the sdp. my technical contact doesn't want to switch 180 to 183 on the sonus side - this would also work (i think). in fact he says that 180 ringing is vaild, he isn't that wrong in this case. our freeswitch works in proxy mode, we do use trunk 15396 see a ngrep trace under http://pastebin.freeswitch.org/11235 92.63.208.36 - freeswitch 38.105.229.100 - sonus br -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From mike at jerris.com Mon Nov 23 11:22:17 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 14:22:17 -0500 Subject: [Freeswitch-users] mod_flite sound profiles In-Reply-To: <012401ca6c6c$a19673a0$e4c35ae0$@thakershi@continuityhealth.com> References: <008301ca6a37$ce104a00$6a30de00$@thakershi@continuityhealth.com> <1AB27F16-3096-49ED-B812-F37D8DADD96C@freeswitch.org> <00c901ca6c57$1c2df950$5489ebf0$@thakershi@continuityhealth.com> <010701ca6c68$0950f6a0$1bf2e3e0$@thakershi@continuityhealth.com> <57CD54FA-C16E-4F36-9678-6CF1FB2D1A17@freeswitch.org> <012401ca6c6c$a19673a0$e4c35ae0$@thakershi@continuityhealth.com> Message-ID: <80AF7620-0980-4B75-A9B0-F046356DF591@jerris.com> Sounds like they don't want your business that much. You can try using mrcp with them , not sure if they have that released on their side or not. I think the build integration for mrcp client just went into the windows build earlier today. To be honest we used to have a pretty good relationship with them but we have had basically no response at all to any technical problems we have had with them in quite some time, so maybe they have decided to move on and not work with open source any more. It would appear so from their actions at least. Mike On Nov 23, 2009, at 1:41 PM, Malay Thakershi wrote: > Thank you for your responses. > > I did follow that web link to ask them as instructed but they declined. They asked me where I want to use it. > > I told them I wanted it to build FreeSwitch so that I can use Cepstral voices (to be purchased from them with it). Their response was they do not provide trial of the SDK. They do not support FreeSwitch. > > Malay Thakershi > > From: Brian West [mailto:brian at freeswitch.org] > Sent: Monday, November 23, 2009 12:14 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_flite sound profiles > > You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. > > /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/a4153509/attachment-0002.html From pjintheusa at gmail.com Mon Nov 23 11:24:18 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 23 Nov 2009 14:24:18 -0500 Subject: [Freeswitch-users] Simplest of Conference Setup questions Message-ID: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> Hi there, I have created a simple conference that works great. The only problem is, when a participant press # it exits the call. So when a user enters a conference with a PIN, and by habit they enter 12345 followed by pound, it puts them in and then straight out. So I edited conference.conf.xml so: and even assigned # to another function: and the same occurs. Pressing # exits the conference. What am I missing here? tia - phil Conf Setup: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/4ad7a208/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 23 11:35:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 13:35:54 -0600 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> Message-ID: <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> issue console loglevel debug from the cli then try again and see if there is any hint On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones wrote: > Hi there, > > I have created a simple conference that works great. The only problem is, > when a participant press # it exits the call. So when a user enters a > conference with a PIN, and by habit they enter 12345 followed by pound, it > puts them in and then straight out. > > So I edited conference.conf.xml so: > > > > and even assigned # to another function: > > > > and the same occurs. Pressing # exits the conference. > > What am I missing here? > > tia - phil > > > > Conf Setup: > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/c6e78263/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 23 11:40:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 13:40:22 -0600 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> Message-ID: <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> or open a new outbound connection at the end of your script so you can send your originate command. Since the channel hanging up will close your existing connection since it's only an outbound single session socket. On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins wrote: > > > On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> Hi, >> I'm using perl ESL to control the call in freeswitch. >> I'm having the following scenario, but not able to get it right. >> >> Dialplan: >> >> >> >> >> >> >> >> >> 1. User A calls to an extention (1000). >> 2. My ESL program will be running, and it answers the call. >> 3. Then the program will get a number from the user. >> 4. It will hangup the call. >> 5. The program has to call to the number that was given by the user. >> >> In the above scenario, I was able to do until the 4th step. After hangup >> the call, if I say originate it is not working. >> Any ideas on how to do this in ESL. >> >> > I want to make sure I understand what the script is supposed to be doing. > The caller will key in a phone number to your script and your script will > collect those digits. The script will then hangup on the caller and > originate a completely new call? Perhaps you could use sched_api to schedule > a new originate command for a few seconds into the future and then hangup? > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/70c790ad/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 23 11:45:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 13:45:34 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <4B0ADFE1.4070506@xpirio.com> References: <4B0ADFE1.4070506@xpirio.com> Message-ID: <191c3a030911231145j384e5bbat7208633895b3b7af@mail.gmail.com> you need to provide a FS console trace of your problem from your FS source dir (build root) cd libs/esl make perlmod cd perl perl logger.pl -pb christian reproduce then hit ctl-c and tell me the url it posted to. 2009/11/23 Christian L?schenkohl > hi > > our freeswitch server has to talk to a sonus ip-switch > when we want to setup a call we do get a "100 Trying" and then a "180 > Ringing" > within the "180 Ringing" we get a sdp with "a=sendonly" then our freeswitch > quits with a CANCEL message. > i simply don't get why our freeswitch aborts the session - i think it would > work > if no "a=sendonly" would be present in the sdp. > > my technical contact doesn't want to switch 180 to 183 on the sonus side - > this would > also work (i think). in fact he says that 180 ringing is vaild, he isn't > that wrong in > this case. > > our freeswitch works in proxy mode, we do use trunk 15396 > see a ngrep trace under http://pastebin.freeswitch.org/11235 > > 92.63.208.36 - freeswitch > 38.105.229.100 - sonus > > br > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/5a0fe41b/attachment-0002.html From brian at freeswitch.org Mon Nov 23 11:48:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 13:48:18 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <4B0ADFE1.4070506@xpirio.com> References: <4B0ADFE1.4070506@xpirio.com> Message-ID: <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> Well its also G729 so I suspect you don't have G729 /b On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > hi > > our freeswitch server has to talk to a sonus ip-switch > when we want to setup a call we do get a "100 Trying" and then a > "180 Ringing" > within the "180 Ringing" we get a sdp with "a=sendonly" then our > freeswitch > quits with a CANCEL message. > i simply don't get why our freeswitch aborts the session - i think > it would work > if no "a=sendonly" would be present in the sdp. > > my technical contact doesn't want to switch 180 to 183 on the sonus > side - this would > also work (i think). in fact he says that 180 ringing is vaild, he > isn't that wrong in > this case. > > our freeswitch works in proxy mode, we do use trunk 15396 > see a ngrep trace under http://pastebin.freeswitch.org/11235 > > 92.63.208.36 - freeswitch > 38.105.229.100 - sonus > > br > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From robert.hadley at teotech.com Mon Nov 23 11:54:19 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Mon, 23 Nov 2009 11:54:19 -0800 Subject: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir In-Reply-To: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com> References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com> Message-ID: <8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com> Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test -d "$(switch_srcdir)/src/mod/$$confmoddir" ; then \ moddir = "$(switch_srcdir)/src/mod/$$confmoddir" ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test -f "$$moddir/Makefile" ; then \ <-- Yep, this will be true cd $$moddir && . && $(MAKE) I'm not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/4d01c933/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 23 11:57:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 13:57:43 -0600 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: <4B0266F4.8070602@savion.huji.ac.il> References: <4B0266F4.8070602@savion.huji.ac.il> Message-ID: <191c3a030911231157p44612c5dm3f0ee1e7b37e9cd3@mail.gmail.com> Let's just do this: r15629 or higher look for sip_profile_name On Tue, Nov 17, 2009 at 3:03 AM, Eli Hayun wrote: > Hi > We have more then one profile. To make a call I have to enter : bridge > sofia/profile/number at ip > The problem is when I use : "${use_profile}" I am getting the caller > profile, and I need the destination profile. > > How do I get this information? > > Thanks > > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/e961d695/attachment-0002.html From msc at freeswitch.org Mon Nov 23 12:12:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 12:12:21 -0800 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> Message-ID: <87f2f3b90911231212x2467e0f3r44824e52f86773ea@mail.gmail.com> On Mon, Nov 23, 2009 at 11:24 AM, Phillip Jones wrote: > Hi there, > > I have created a simple conference that works great. The only problem is, > when a participant press # it exits the call. So when a user enters a > conference with a PIN, and by habit they enter 12345 followed by pound, it > puts them in and then straight out. > > So I edited conference.conf.xml so: > > > > and even assigned # to another function: > > > > and the same occurs. Pressing # exits the conference. > > What am I missing here? > > tia - phil > > > Phil, I recommend that you create a custom profile and a custom caller control group. Just copy the defaults and rename them to something meaningful. In conference.conf.xml you can add a new call control group like this: Then make a copy of the default profile changing the profile name and the caller-controls parameter: Give that a whirl and report back. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/1bd3bc70/attachment-0002.html From pjintheusa at gmail.com Mon Nov 23 12:17:33 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 23 Nov 2009 15:17:33 -0500 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> Message-ID: <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> Thanks for replying. Well in the log I see: 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving conference, cause: NONE which make sense because just above I see: 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default caller control action 'hangup' bound to '#'. The question I have - is how do I change that default caller control action if it is not in conference.conf.xml ?? ... ** On Mon, Nov 23, 2009 at 2:35 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > issue > > console loglevel debug > from the cli > > then try again and see if there is any hint > > > On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones wrote: > >> Hi there, >> >> I have created a simple conference that works great. The only problem is, >> when a participant press # it exits the call. So when a user enters a >> conference with a PIN, and by habit they enter 12345 followed by pound, it >> puts them in and then straight out. >> >> So I edited conference.conf.xml so: >> >> >> >> and even assigned # to another function: >> >> >> >> and the same occurs. Pressing # exits the conference. >> >> What am I missing here? >> >> tia - phil >> >> >> >> Conf Setup: >> >> >> >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/a35891b1/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 23 12:25:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 14:25:35 -0600 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> Message-ID: <191c3a030911231225i457a4329j3ec578d9f594db03@mail.gmail.com> see what happens if you set hangup to some other key or the word "event" On Mon, Nov 23, 2009 at 2:17 PM, Phillip Jones wrote: > Thanks for replying. > > Well in the log I see: > > 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 > 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving > conference, cause: NONE > > which make sense because just above I see: > > 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default > caller control action 'hangup' bound to '#'. > > The question I have - is how do I change that default caller control action > if it is not in conference.conf.xml ?? > > > > ... > > ** > > > On Mon, Nov 23, 2009 at 2:35 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> issue >> >> console loglevel debug >> from the cli >> >> then try again and see if there is any hint >> >> >> On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones wrote: >> >>> Hi there, >>> >>> I have created a simple conference that works great. The only problem is, >>> when a participant press # it exits the call. So when a user enters a >>> conference with a PIN, and by habit they enter 12345 followed by pound, it >>> puts them in and then straight out. >>> >>> So I edited conference.conf.xml so: >>> >>> >>> >>> and even assigned # to another function: >>> >>> >>> >>> and the same occurs. Pressing # exits the conference. >>> >>> What am I missing here? >>> >>> tia - phil >>> >>> >>> >>> Conf Setup: >>> >>> >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/37969033/attachment-0002.html From msc at freeswitch.org Mon Nov 23 12:27:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 12:27:38 -0800 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> Message-ID: <87f2f3b90911231227s51f2a2f4r28ab93c77eb9ac61@mail.gmail.com> On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones wrote: > Thanks for replying. > > Well in the log I see: > > 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 > 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving > conference, cause: NONE > > which make sense because just above I see: > > 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default > caller control action 'hangup' bound to '#'. > > The question I have - is how do I change that default caller control action > if it is not in conference.conf.xml ?? > > > > ... > > ** > I believe that this is because the caller-controls param is commented out in the default profile config. I prefer not to mess w/ the default configs which is why I recommended the custom configs in my previous email... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/5ba7a184/attachment-0002.html From Mailings at kh-dev.de Mon Nov 23 12:29:51 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Mon, 23 Nov 2009 21:29:51 +0100 Subject: [Freeswitch-users] FS dies after some minutes Message-ID: Hi, I did a new installation with the trunk from Saturday (21. Nov.) and it always dies with a core after 5-10 minutes. It happened several times. After that I did a new installation of 1.0.4 and this runs without problems on the same host. I'm using Ubuntu 8.04 Server with all patches. Anyone else experiencing this problem? Thanks, Klaus Here's the bt: #0 0x00007f7aa2eb22fc in sofia_reg_nonce_callback (pArg=0x40f3ca50, argc=, argv=0x7f7a9c006758, columnNames=) at ../../../../src/include/switch_utils.h:78 #1 0x00007f7aa91f4a12 in sqlite3_exec (db=0x7f7a9c00a6a0, zSql=0x7f7a9c006cd0 "select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8'", xCallback=0x7f7aa2eb22d0 , pArg=0x40f3ca50, pzErrMsg=0x40f3c680) at ./src/legacy.c:95 #2 0x00007f7aa917b98d in switch_core_db_exec (db=0x7f7a9c00a6a0, sql=0x7f7a9c006cd0 "select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8'", callback=0x7f7aa2eb22d0 , data=0x40f3ca50, errmsg=0x40f3c6e8) at src/switch_core_db.c:93 #3 0x00007f7aa2e985b1 in sofia_glue_execute_sql_callback (profile=0x72e940, mutex=0x0, sql=0x7f7a9c006cd0 "select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8'", callback=0x7f7aa2eb22d0 , pdata=0x40f3ca50) at sofia_glue.c:4297 #4 0x00007f7aa2ead8ec in sofia_reg_parse_auth (profile=0x72e940, authorization=0x7f7a9c078ad0, sip=0x7f7a9c0695d8, regstr=0x7f7aa2fe7137 "REGISTER", np=0x40f3d940 "b7ed6efa-d801-11de-a716-67cbb4a551f8", nplen=128, ip=0x40f3d840 "10.134.38.59", v_event=0x40f3d930, exptime=3600, regtype=REG_REGISTER, to_user=0x7f7a9c0dd18e "29", auth_params=0x40f3cd60, reg_count=0x40f3cd58) at sofia_reg.c:1704 #5 0x00007f7aa2eb004a in sofia_reg_handle_register (nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sip=0x7f7a9c0695d8, regtype=REG_REGISTER, key=0x40f3d940 "b7ed6efa-d801-11de-a716-67cbb4a551f8", keylen=0, v_event=0x40f3d930, is_nat=0x0) at sofia_reg.c:888 #6 0x00007f7aa2eb2f1c in sofia_reg_handle_sip_i_register (nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sofia_private=, sip=0x7f7a9c0695d8, tags=) at sofia_reg.c:1362 #7 0x00007f7aa2e9371c in sofia_event_callback (event=, status=100, phrase=0x7f7a9c071700 "Trying", nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sofia_private=0x0, sip=0x7f7a9c0695d8, tags=0x7f7a9c0716f0) at sofia.c:672 #8 0x00007f7aa2f1119e in nua_application_event (dummy=0x0, sumsg=0x40f3dd10, ee=0x7f7a9c0716c8) at nua_stack.c:393 #9 0x00007f7aa2f7e2c1 in su_base_port_execute_msgs (queue=0x0) at su_base_port.c:280 #10 0x00007f7aa2f7e039 in su_base_port_getmsgs (self=0x721320) at su_base_port.c:202 #11 0x00007f7aa2f7e5dc in su_base_port_step (self=0x721320, tout=0) at su_base_port.c:473 #12 0x00007f7aa2f7b68e in su_port_step (self=0x721320, tout=1000) at su_port.h:340 #13 0x00007f7aa2f7b656 in su_root_step (self=0x723320, tout=1000) at su_root.c:858 #14 0x00007f7aa2e8d40a in sofia_profile_thread_run (thread=, obj=) at sofia.c:1194 #15 0x00007f7aa88b63f7 in start_thread () from /lib/libpthread.so.0 #16 0x00007f7aa7e20b4d in clone () from /lib/libc.so.6 #17 0x0000000000000000 in ?? () -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/ca7d653d/attachment-0002.html From pjintheusa at gmail.com Mon Nov 23 12:32:39 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 23 Nov 2009 15:32:39 -0500 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <87f2f3b90911231212x2467e0f3r44824e52f86773ea@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <87f2f3b90911231212x2467e0f3r44824e52f86773ea@mail.gmail.com> Message-ID: <367751820911231232mea2e90dlf75590ca3f0ae839@mail.gmail.com> Michael that for the reply. I created a new group with # unbound and referenced it from the default profile: And that worked fine. Strangely though, changing the default group and referencing that from the default profile does not. Do you want me to test this on the latest trunk or is this as expected? Phil On Mon, Nov 23, 2009 at 3:12 PM, Michael Collins wrote: > > > On Mon, Nov 23, 2009 at 11:24 AM, Phillip Jones wrote: > >> Hi there, >> >> I have created a simple conference that works great. The only problem is, >> when a participant press # it exits the call. So when a user enters a >> conference with a PIN, and by habit they enter 12345 followed by pound, it >> puts them in and then straight out. >> >> So I edited conference.conf.xml so: >> >> >> >> and even assigned # to another function: >> >> >> >> and the same occurs. Pressing # exits the conference. >> >> What am I missing here? >> >> tia - phil >> >> >> > Phil, > > I recommend that you create a custom profile and a custom caller control > group. Just copy the defaults and rename them to something meaningful. In > conference.conf.xml you can add a new call control group like this: > > > digits="0"/> > > digits="*"/> > > digits="9"/> > > digits="8"/> > > digits="7"/> > > digits="3"/> > > digits="2"/> > > digits="1"/> > > digits="6"/> > > digits="5"/> > > digits="4"/> > > > > > Then make a copy of the default profile changing the profile name and the > caller-controls parameter: > > > > > > > > > > > Give that a whirl and report back. :) > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/5c8d6e2e/attachment-0002.html From brian at freeswitch.org Mon Nov 23 12:34:36 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 14:34:36 -0600 Subject: [Freeswitch-users] FS dies after some minutes In-Reply-To: References: Message-ID: Update to SVN Trunk. /b On Nov 23, 2009, at 2:29 PM, Klaus Hochlehnert wrote: > Hi, > > I did a new installation with the trunk from Saturday (21. Nov.) and > it always dies with a core after 5-10 minutes. > It happened several times. > After that I did a new installation of 1.0.4 and this runs without > problems on the same host. > I?m using Ubuntu 8.04 Server with all patches. > > Anyone else experiencing this problem? > > Thanks, Klaus > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/968d050b/attachment-0002.html From pjintheusa at gmail.com Mon Nov 23 12:42:50 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 23 Nov 2009 15:42:50 -0500 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <87f2f3b90911231227s51f2a2f4r28ab93c77eb9ac61@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> <87f2f3b90911231227s51f2a2f4r28ab93c77eb9ac61@mail.gmail.com> Message-ID: <367751820911231242o5d329480x5523a24696c6fa56@mail.gmail.com> Anthony - setting or does not make a difference, even when the default profile has un-commented. Looks to me like that default group is ignored even when specifically referred to? As Michael says though, creating a specific group: and adding in the default profile works a charm. I am good - but let me know if you want me to try anything else. Phil On Mon, Nov 23, 2009 at 3:27 PM, Michael Collins wrote: > > > On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones wrote: > >> Thanks for replying. >> >> Well in the log I see: >> >> 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 >> 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving >> conference, cause: NONE >> >> which make sense because just above I see: >> >> 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default >> caller control action 'hangup' bound to '#'. >> >> The question I have - is how do I change that default caller control >> action if it is not in conference.conf.xml ?? >> >> >> >> ... >> >> ** >> > > I believe that this is because the caller-controls param is commented out > in the default profile config. I prefer not to mess w/ the default configs > which is why I recommended the custom configs in my previous email... > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/1ec9d80b/attachment-0002.html From christian.loeschenkohl at xpirio.com Mon Nov 23 12:54:21 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 23 Nov 2009 21:54:21 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <191c3a030911231145j384e5bbat7208633895b3b7af@mail.gmail.com> References: <4B0ADFE1.4070506@xpirio.com> <191c3a030911231145j384e5bbat7208633895b3b7af@mail.gmail.com> Message-ID: <4B0AF67D.6040707@xpirio.com> thank you for your answer the relevant part of the log is 2009-11-23 21:46:49.625130 [NOTICE] sofia.c:3693 Pre-Answer sofia/interconnect/24785214448370068 at 38.105.229.100! 2009-11-23 21:46:49.625130 [INFO] sofia.c:3706 Sending early media 2009-11-23 21:46:49.625130 [ERR] sofia_glue.c:2029 No audio codec available 2009-11-23 21:46:49.625130 [NOTICE] switch_channel.c:2048 Hangup sofia/interconnect/nobody at 81.94.55.100 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] it's the same with g729 and alaw (refering to brian) in my opinion the ringing here should be generated near end and no audio codec has to be used here (180 ringing) br On 2009-11-23 20:45, Anthony Minessale wrote: > you need to provide a FS console trace of your problem > > from your FS source dir (build root) > > cd libs/esl > make perlmod > cd perl > perl logger.pl -pb christian > > reproduce > > > then hit ctl-c and tell me the url it posted to. > > > > 2009/11/23 Christian L?schenkohl > > > hi > > our freeswitch server has to talk to a sonus ip-switch > when we want to setup a call we do get a "100 Trying" and then a > "180 Ringing" > within the "180 Ringing" we get a sdp with "a=sendonly" then our > freeswitch > quits with a CANCEL message. > i simply don't get why our freeswitch aborts the session - i think > it would work > if no "a=sendonly" would be present in the sdp. > > my technical contact doesn't want to switch 180 to 183 on the sonus > side - this would > also work (i think). in fact he says that 180 ringing is vaild, he > isn't that wrong in > this case. > > our freeswitch works in proxy mode, we do use trunk 15396 > see a ngrep trace under http://pastebin.freeswitch.org/11235 > > 92.63.208.36 - freeswitch > 38.105.229.100 - sonus > > br > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From christian.loeschenkohl at xpirio.com Mon Nov 23 12:56:15 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 23 Nov 2009 21:56:15 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> Message-ID: <4B0AF6EF.8070507@xpirio.com> thany ou for your answer we use g729 on all our other connections in passthrough mode and it also doesn't work with alaw. so i don't think it's related to this. br On 2009-11-23 20:48, Brian West wrote: > Well its also G729 so I suspect you don't have G729 > > /b > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > >> hi >> >> our freeswitch server has to talk to a sonus ip-switch >> when we want to setup a call we do get a "100 Trying" and then a >> "180 Ringing" >> within the "180 Ringing" we get a sdp with "a=sendonly" then our >> freeswitch >> quits with a CANCEL message. >> i simply don't get why our freeswitch aborts the session - i think >> it would work >> if no "a=sendonly" would be present in the sdp. >> >> my technical contact doesn't want to switch 180 to 183 on the sonus >> side - this would >> also work (i think). in fact he says that 180 ringing is vaild, he >> isn't that wrong in >> this case. >> >> our freeswitch works in proxy mode, we do use trunk 15396 >> see a ngrep trace under http://pastebin.freeswitch.org/11235 >> >> 92.63.208.36 - freeswitch >> 38.105.229.100 - sonus >> >> br >> >> -- >> Ing. Christian L?schenkohl >> Technische Leitung, Forschung& Entwicklung VoIP >> >> xpirio >> Telekommunikation& Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenkohl at xpirio.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From anthony.minessale at gmail.com Mon Nov 23 13:07:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 15:07:30 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <4B0AF6EF.8070507@xpirio.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> Message-ID: <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> do you have the ringback variable set on the channel? if so it will cause 180 to attempt to play inband ringback indication I have nothing left to say because I asked for the whole log with the siptrace enables not just 5 lines of it. If you still want help, give me the log to examine and I will tell you what your problem is. 2009/11/23 Christian L?schenkohl > thany ou for your answer > > we use g729 on all our other connections in passthrough mode and it also > doesn't work with alaw. > so i don't think it's related to this. > > br > > > On 2009-11-23 20:48, Brian West wrote: > > Well its also G729 so I suspect you don't have G729 > > > > /b > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > >> hi > >> > >> our freeswitch server has to talk to a sonus ip-switch > >> when we want to setup a call we do get a "100 Trying" and then a > >> "180 Ringing" > >> within the "180 Ringing" we get a sdp with "a=sendonly" then our > >> freeswitch > >> quits with a CANCEL message. > >> i simply don't get why our freeswitch aborts the session - i think > >> it would work > >> if no "a=sendonly" would be present in the sdp. > >> > >> my technical contact doesn't want to switch 180 to 183 on the sonus > >> side - this would > >> also work (i think). in fact he says that 180 ringing is vaild, he > >> isn't that wrong in > >> this case. > >> > >> our freeswitch works in proxy mode, we do use trunk 15396 > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > >> > >> 92.63.208.36 - freeswitch > >> 38.105.229.100 - sonus > >> > >> br > >> > >> -- > >> Ing. Christian L?schenkohl > >> Technische Leitung, Forschung& Entwicklung VoIP > >> > >> xpirio > >> Telekommunikation& Service GmbH > >> Lakeside B04 > >> 9020 Klagenfurt > >> Austria > >> > >> T +43 (0) 5 77 11 - 1000 > >> F +43 (0) 5 77 11 - 1002 > >> E christian.loeschenkohl at xpirio.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/12a10e08/attachment-0002.html From rupa at rupa.com Mon Nov 23 13:08:55 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 23 Nov 2009 15:08:55 -0600 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231242o5d329480x5523a24696c6fa56@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> <87f2f3b90911231227s51f2a2f4r28ab93c77eb9ac61@mail.gmail.com> <367751820911231242o5d329480x5523a24696c6fa56@mail.gmail.com> Message-ID: The behavior of not being able to change the default caller controls are documented on the wiki: http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E *Reserved Group Names* - none - Use this name to prevent installing caller-controls for callers of a conference. - default - Use this name to utilize the hard-coded set of controls built-in to mod_conference. Do NOT name a custom set of conference-controls "default" as they will be overridden with the hard-coded set. The behavior of the "default" group is defined below: On Mon, Nov 23, 2009 at 2:42 PM, Phillip Jones wrote: > Anthony - setting > > > > or > > > > does not make a difference, even when the default profile has > > > > un-commented. > > > Looks to me like that default group is ignored even when specifically > referred to? > > As Michael says though, creating a specific group: > > > > and adding > > in the default profile > works a charm. > > I am good - but let me know if you want me to try anything else. > > Phil > > > > On Mon, Nov 23, 2009 at 3:27 PM, Michael Collins wrote: > >> >> >> On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones wrote: >> >>> Thanks for replying. >>> >>> Well in the log I see: >>> >>> 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 >>> 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving >>> conference, cause: NONE >>> >>> which make sense because just above I see: >>> >>> 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing >>> default caller control action 'hangup' bound to '#'. >>> >>> The question I have - is how do I change that default caller control >>> action if it is not in conference.conf.xml ?? >>> >>> >>> >>> ... >>> >>> ** >>> >> >> I believe that this is because the caller-controls param is commented out >> in the default profile config. I prefer not to mess w/ the default configs >> which is why I recommended the custom configs in my previous email... >> >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/f49bd641/attachment-0002.html From mike at jerris.com Mon Nov 23 13:09:05 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 16:09:05 -0500 Subject: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir In-Reply-To: <8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com> References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com> <8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com> Message-ID: <99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com> In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: > Thanks Mike. > > modmake.rules is created in the $(switch_builddir)/build. > > What I see as the problem is in src/mod/Makefile.am > > There is a statement line 12 that points moddir to the source > if test ?d ?$(switch_srcdir)/src/mod/$$confmoddir? ; then \ > moddir = ?$(switch_srcdir)/src/mod/$$confmoddir? ; > > And then the statements starting around line 22 that cd to moddir (in src) and fire off make > if test ?f ?$$moddir/Makefile? ; then \ ? Yep, this will be true > cd $$moddir && ? && $(MAKE) > > I?m not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). > > Does anybody have a better idea? > > Thanks, > Robert > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Monday, November 23, 2009 11:16 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir > > The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. > > Mike > > On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: > > > I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/73d28cb1/attachment-0002.html From mike at jerris.com Mon Nov 23 13:14:27 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 16:14:27 -0500 Subject: [Freeswitch-users] Simplest of Conference Setup questions In-Reply-To: <367751820911231242o5d329480x5523a24696c6fa56@mail.gmail.com> References: <367751820911231124l2e5830e9i1b92beb626376a8c@mail.gmail.com> <191c3a030911231135j37a6c0ben5dd60604f88a86d6@mail.gmail.com> <367751820911231217v3fcf009o2ec5ec9c4c507d2f@mail.gmail.com> <87f2f3b90911231227s51f2a2f4r28ab93c77eb9ac61@mail.gmail.com> <367751820911231242o5d329480x5523a24696c6fa56@mail.gmail.com> Message-ID: <0E108DAC-8A58-41D3-A194-F092AB4FBF87@jerris.com> Default controls are hard coded. If you want to change them you must use a name other than default. Mike On Nov 23, 2009, at 3:42 PM, Phillip Jones wrote: > Anthony - setting > > > > or > > > > does not make a difference, even when the default profile has > > > > un-commented. > > > Looks to me like that default group is ignored even when specifically referred to? > > As Michael says though, creating a specific group: > > > > and adding > > in the default profile works a charm. > > I am good - but let me know if you want me to try anything else. > > Phil > From robert.hadley at teotech.com Mon Nov 23 13:19:08 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Mon, 23 Nov 2009 13:19:08 -0800 Subject: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In-Reply-To: <99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com> References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com><8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com> <99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com> Message-ID: <6987A58F102E4AAE8A5F64581653ED78@greyhawk.tonecommander.com> In typical automake builds the configure step takes the Makefile.am from the srcdir and generates the Makefile in the builddir. Most src/mod subdirs are not using automake and/or configure. They just have a simple Makefile in with the source. Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 1:09 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test -d "$(switch_srcdir)/src/mod/$$confmoddir" ; then \ moddir = "$(switch_srcdir)/src/mod/$$confmoddir" ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test -f "$$moddir/Makefile" ; then \ <-- Yep, this will be true cd $$moddir && . && $(MAKE) I'm not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/a3fe6090/attachment-0002.html From christian.loeschenkohl at xpirio.com Mon Nov 23 13:36:30 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 23 Nov 2009 22:36:30 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> Message-ID: <4B0B005E.4080202@xpirio.com> sorry about wasting your time (wasn't my intent) the log is at http://pastebin.freeswitch.org/11240 i called 5214448370068 (also other calls are in the log) they now have changed 180 to 183 on the sonus, but makes no difference here br On 2009-11-23 22:07, Anthony Minessale wrote: > do you have the ringback variable set on the channel? > if so it will cause 180 to attempt to play inband ringback indication > > I have nothing left to say because I asked for the whole log with the > siptrace enables not just 5 lines of it. > If you still want help, give me the log to examine and I will tell you > what your problem is. > > > > 2009/11/23 Christian L?schenkohl > > > thany ou for your answer > > we use g729 on all our other connections in passthrough mode and it > also doesn't work with alaw. > so i don't think it's related to this. > > br > > > On 2009-11-23 20:48, Brian West wrote: > > Well its also G729 so I suspect you don't have G729 > > > > /b > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > >> hi > >> > >> our freeswitch server has to talk to a sonus ip-switch > >> when we want to setup a call we do get a "100 Trying" and then a > >> "180 Ringing" > >> within the "180 Ringing" we get a sdp with "a=sendonly" then our > >> freeswitch > >> quits with a CANCEL message. > >> i simply don't get why our freeswitch aborts the session - i think > >> it would work > >> if no "a=sendonly" would be present in the sdp. > >> > >> my technical contact doesn't want to switch 180 to 183 on the sonus > >> side - this would > >> also work (i think). in fact he says that 180 ringing is vaild, he > >> isn't that wrong in > >> this case. > >> > >> our freeswitch works in proxy mode, we do use trunk 15396 > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > >> > >> 92.63.208.36 - freeswitch > >> 38.105.229.100 - sonus > >> > >> br > >> > >> -- > >> Ing. Christian L?schenkohl > >> Technische Leitung, Forschung& Entwicklung VoIP > >> > >> xpirio > >> Telekommunikation& Service GmbH > >> Lakeside B04 > >> 9020 Klagenfurt > >> Austria > >> > >> T +43 (0) 5 77 11 - 1000 > >> F +43 (0) 5 77 11 - 1002 > >> E christian.loeschenkohl at xpirio.com > > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From mike at jerris.com Mon Nov 23 13:42:24 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 16:42:24 -0500 Subject: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In-Reply-To: <6987A58F102E4AAE8A5F64581653ED78@greyhawk.tonecommander.com> References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com><8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com> <99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com> <6987A58F102E4AAE8A5F64581653ED78@greyhawk.tonecommander.com> Message-ID: I'll work on this, can you open me up a bug on http://jira.freeswitch.org in regards to this please. Mike On Nov 23, 2009, at 4:19 PM, Robert Hadley wrote: > In typical automake builds the configure step takes the Makefile.am from the srcdir and generates the Makefile in the builddir. > > Most src/mod subdirs are not using automake and/or configure. They just have a simple Makefile in with the source. > > Robert > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Monday, November 23, 2009 1:09 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir > > In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? > > Mike > > On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: > > > Thanks Mike. > > modmake.rules is created in the $(switch_builddir)/build. > > What I see as the problem is in src/mod/Makefile.am > > There is a statement line 12 that points moddir to the source > if test ?d ?$(switch_srcdir)/src/mod/$$confmoddir? ; then \ > moddir = ?$(switch_srcdir)/src/mod/$$confmoddir? ; > > And then the statements starting around line 22 that cd to moddir (in src) and fire off make > if test ?f ?$$moddir/Makefile? ; then \ ? Yep, this will be true > cd $$moddir && ? && $(MAKE) > > I?m not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). > > Does anybody have a better idea? > > Thanks, > Robert > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Monday, November 23, 2009 11:16 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir > > The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. > > Mike > > On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: > > > > I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/3704f258/attachment-0002.html From rob4manhere at gmail.com Mon Nov 23 13:53:56 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Mon, 23 Nov 2009 15:53:56 -0600 Subject: [Freeswitch-users] Memory leak with mod_local_stream Message-ID: Hey guys, Having a problem with mod_local_stream. I recently did a "make current" from 15334 to the latest trunk (15630). After restarting, there now appears to be a memory leak. On a test system (CentOS 5.4, 64-bit) with no calls or registrations, Freeswitch gradually consumes all of the host memory (rate of about 200K/second), then swaps out, eventually rendering the system useless. I isolated it to mod_local_stream. If I unload mod_local_stream, the memory use stops climbing. If I re-load mod_local_stream, it starts again. I would submit the logs except they aren't any besides it starting. The system is just sitting there idle. Even valgrind didn't show much (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- check=full --leak-resolution=high --show-reachable=yes .libs/ freeswitch -vg Questions: * has anyone else seen this? * what is the best way I can assist troubleshooting this? I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could that have anything to do with it? Rob From jaybinks at gmail.com Mon Nov 23 14:12:07 2009 From: jaybinks at gmail.com (Jay Binks) Date: Tue, 24 Nov 2009 08:12:07 +1000 Subject: [Freeswitch-users] Memory leak with mod_local_stream In-Reply-To: References: Message-ID: <13396C7D-C89A-4ABC-A63F-EE5A3F8DBC50@gmail.com> if you suspect 15431 to have caused this, then revert to 15430 and see if the problem exists. if you can narrow do the bug to a specific svn revision, then you greatly assist in the resolution of the issue. apart from that im not much help sorry. maybe someone else can lab it up and see if they get the same result. ( Im on a train now, so not so easy :P ) J On 24/11/2009, at 7:53 AM, Rob Forman wrote: > Hey guys, > > Having a problem with mod_local_stream. > > I recently did a "make current" from 15334 to the latest trunk > (15630). After restarting, there now appears to be a memory leak. On > a test system (CentOS 5.4, 64-bit) with no calls or registrations, > Freeswitch gradually consumes all of the host memory (rate of about > 200K/second), then swaps out, eventually rendering the system useless. > > I isolated it to mod_local_stream. If I unload mod_local_stream, the > memory use stops climbing. If I re-load mod_local_stream, it starts > again. > > > I would submit the logs except they aren't any besides it starting. > The system is just sitting there idle. Even valgrind didn't show much > (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I > ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- > check=full --leak-resolution=high --show-reachable=yes .libs/ > freeswitch -vg > > Questions: > * has anyone else seen this? > * what is the best way I can assist troubleshooting this? > > I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could > that have anything to do with it? > > Rob > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Mon Nov 23 14:15:23 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Nov 2009 17:15:23 -0500 Subject: [Freeswitch-users] Memory leak with mod_local_stream In-Reply-To: References: Message-ID: <3BF17413-EC48-4691-8C99-61CC4661E2CA@jerris.com> That rev should have fixed that memory leak, could you test mod_local_stream.c from rev 15430 (http://fisheye.freeswitch.org/browse/~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/mod_local_stream.c) with your current fs version to confirm this is the cause please? Mike On Nov 23, 2009, at 4:53 PM, Rob Forman wrote: > Hey guys, > > Having a problem with mod_local_stream. > > I recently did a "make current" from 15334 to the latest trunk > (15630). After restarting, there now appears to be a memory leak. On > a test system (CentOS 5.4, 64-bit) with no calls or registrations, > Freeswitch gradually consumes all of the host memory (rate of about > 200K/second), then swaps out, eventually rendering the system useless. > > I isolated it to mod_local_stream. If I unload mod_local_stream, the > memory use stops climbing. If I re-load mod_local_stream, it starts > again. > > > I would submit the logs except they aren't any besides it starting. > The system is just sitting there idle. Even valgrind didn't show much > (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I > ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- > check=full --leak-resolution=high --show-reachable=yes .libs/ > freeswitch -vg > > Questions: > * has anyone else seen this? > * what is the best way I can assist troubleshooting this? > > I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could > that have anything to do with it? > > Rob > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From robert.hadley at teotech.com Mon Nov 23 14:21:08 2009 From: robert.hadley at teotech.com (Robert Hadley) Date: Mon, 23 Nov 2009 14:21:08 -0800 Subject: [Freeswitch-users] Building in a builddir using--srcdiroptionbut modules still build in srcdir In-Reply-To: References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com><8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com><99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com><6987A58F102E4AAE8A5F64581653ED78@greyhawk.tonecommander.com> Message-ID: Thanks Mike, How is the easy way to give you the changes I found so far? There are around 10 changes in 30 files (all configure.gnu files need a fix). Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 1:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using--srcdiroptionbut modules still build in srcdir I'll work on this, can you open me up a bug on http://jira.freeswitch.org in regards to this please. Mike On Nov 23, 2009, at 4:19 PM, Robert Hadley wrote: In typical automake builds the configure step takes the Makefile.am from the srcdir and generates the Makefile in the builddir. Most src/mod subdirs are not using automake and/or configure. They just have a simple Makefile in with the source. Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 1:09 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test -d "$(switch_srcdir)/src/mod/$$confmoddir" ; then \ moddir = "$(switch_srcdir)/src/mod/$$confmoddir" ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test -f "$$moddir/Makefile" ; then \ <-- Yep, this will be true cd $$moddir && . && $(MAKE) I'm not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/a3783727/attachment-0002.html From brian at freeswitch.org Mon Nov 23 14:29:14 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 16:29:14 -0600 Subject: [Freeswitch-users] Building in a builddir using--srcdiroptionbut modules still build in srcdir In-Reply-To: References: <83B586B0-70CC-400C-B134-43354709FAC7@jerris.com><8CF1F19F41B6491788AAB34FE3F00466@greyhawk.tonecommander.com><99A38894-0844-4B01-98A4-E91FAA7CA0DF@jerris.com><6987A58F102E4AAE8A5F64581653ED78@greyhawk.tonecommander.com> Message-ID: <51270770-CA9E-4081-B28C-E11113EA4A04@freeswitch.org> go to the src root and type: svn diff > patch.diff then open a jira and attach patch.diff /b On Nov 23, 2009, at 4:21 PM, Robert Hadley wrote: > Thanks Mike, > > How is the easy way to give you the changes I found so far? There > are around 10 changes in 30 files (all configure.gnu files need a > fix). > > Robert > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/f4cadce8/attachment-0002.html From pjintheusa at gmail.com Mon Nov 23 14:34:27 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 23 Nov 2009 17:34:27 -0500 Subject: [Freeswitch-users] register timeout / cisco 7960 Message-ID: <367751820911231434j36b9846dk46d058ddb77c634@mail.gmail.com> hi there, I have set up some cisco 7960 up with fs. They work fine - but the only way I can keep them registered is to set the "timer_register_expires" in the Cisco cfg file to something really short like 10s. Does anyone know the default register timeout for fs? And where I might change this in fs? Thanks! Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/72fd7e4b/attachment-0002.html From lists at redbonez.net Mon Nov 23 14:49:03 2009 From: lists at redbonez.net (Adam Ford) Date: Mon, 23 Nov 2009 15:49:03 -0700 Subject: [Freeswitch-users] FIFO Orgination_caller_id Message-ID: <005701ca6c8f$28eaa570$7abff050$@net> Is there any way to set the origination_caller_id for a FIFO outbound call to an on-hook agent? I can't find anything in the wiki about a FIFO or member variable to set this. It seems to be set to 'Queue' by default, and appears to be hardcoded in the module source. It would be nice to be able to change per FIFO queue. That way agents that handle multiple companies can more easily see which queue is calling and answer accordingly. It is not a big deal, since it does automatically set the origination_caller_id_number to 'fifo+'. However, depending on the phone, the caller ID number is not always readily shown, and must be looked for. Thanks to anyone who has some insight on this, -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/9d905f1e/attachment-0002.html From siniypin at gmail.com Mon Nov 23 14:51:17 2009 From: siniypin at gmail.com (RobertT) Date: Tue, 24 Nov 2009 01:51:17 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <2160023e0911180507k7321dfa7t6104f0cad6e67f9@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> <2160023e0911211523k7998d048nced3af8fb805e770@mail.gmail.com> Message-ID: <2160023e0911231451v59c072adr126584534b1e4f76@mail.gmail.com> OK, this is what I've got. First, I've updated FreeSwitch from trunk to version 15630 and deployed it to my server. Performed a tets and again no magic happened. The link to SIP trace is below. Then I've installed 1.0.4 version to another server (virtual hosting), and performed tha same. And everything went OK. This server's log is below as well. Not working - http://pastebin.com/m2e97985d Working - http://pastebin.com/m3c1e6bfe Also in both cases there is a strange detail - clients' SIP ports are configured to be 5060 and 5061, but what can be seen in trace differs from these values whereas stun resolution shows that there is no NAT (clients connect with ADSL modem). Regards, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/d4d4d38e/attachment-0002.html From rob4manhere at gmail.com Mon Nov 23 15:04:28 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Mon, 23 Nov 2009 17:04:28 -0600 Subject: [Freeswitch-users] Memory leak with mod_local_stream In-Reply-To: <3BF17413-EC48-4691-8C99-61CC4661E2CA@jerris.com> References: <3BF17413-EC48-4691-8C99-61CC4661E2CA@jerris.com> Message-ID: I tried mod_local_stream.c from rev 15430, did a make clean && make all && make install-- but it didn't fix it so it wasn't that patch. I'll make current and try valgrind again unless someone has other ideas. Rob On Nov 23, 2009, at 4:15 PM, Michael Jerris wrote: > That rev should have fixed that memory leak, could you test > mod_local_stream.c from rev 15430 (http://fisheye.freeswitch.org/browse/ > ~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/ > mod_local_stream.c) with your current fs version to confirm this is > the cause please? > > Mike > > > On Nov 23, 2009, at 4:53 PM, Rob Forman wrote: > >> Hey guys, >> >> Having a problem with mod_local_stream. >> >> I recently did a "make current" from 15334 to the latest trunk >> (15630). After restarting, there now appears to be a memory leak. >> On >> a test system (CentOS 5.4, 64-bit) with no calls or registrations, >> Freeswitch gradually consumes all of the host memory (rate of about >> 200K/second), then swaps out, eventually rendering the system >> useless. >> >> I isolated it to mod_local_stream. If I unload mod_local_stream, the >> memory use stops climbing. If I re-load mod_local_stream, it starts >> again. >> >> >> I would submit the logs except they aren't any besides it starting. >> The system is just sitting there idle. Even valgrind didn't show >> much >> (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I >> ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- >> check=full --leak-resolution=high --show-reachable=yes .libs/ >> freeswitch -vg >> >> Questions: >> * has anyone else seen this? >> * what is the best way I can assist troubleshooting this? >> >> I saw a patch to mod_local_stream (rev 15431) a few weeks back. >> Could >> that have anything to do with it? >> >> Rob >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From siniypin at gmail.com Mon Nov 23 15:09:21 2009 From: siniypin at gmail.com (RobertT) Date: Tue, 24 Nov 2009 02:09:21 +0300 Subject: [Freeswitch-users] tcp call misses sip message In-Reply-To: <2160023e0911231451v59c072adr126584534b1e4f76@mail.gmail.com> References: <2160023e0911121427j7df55ae4j6cb0db0993dfccaa@mail.gmail.com> <69D98134-416F-4957-AF63-96E9E7B5DD20@freeswitch.org> <2160023e0911200430h893c50fsdd269db7af7981c5@mail.gmail.com> <8C9B5614-F7B9-4CBF-B406-6DAA2E3D0568@freeswitch.org> <2160023e0911201107x41d84a39r9674ab53939b2242@mail.gmail.com> <2160023e0911210528q5b6c9b37y54a3858ec3a9e138@mail.gmail.com> <69B01CDC-3F11-4937-9F01-4C56E8ED6101@freeswitch.org> <2160023e0911211523k7998d048nced3af8fb805e770@mail.gmail.com> <2160023e0911231451v59c072adr126584534b1e4f76@mail.gmail.com> Message-ID: <2160023e0911231509i6f3ebc2er30d1a24c8d25d1d3@mail.gmail.com> You know what, guys? I've just made it working be opening ALL tcp trafic in and out from server by adding two match-all ip filters into local security policy. I can't say I like this solution... Why did this problem appeared with policy matching exact (sofia profiles) ports? Regards, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/6fcc18ce/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 23 16:13:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 18:13:11 -0600 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <005701ca6c8f$28eaa570$7abff050$@net> References: <005701ca6c8f$28eaa570$7abff050$@net> Message-ID: <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> if you add {origination_caller_id_name=foo,origination_caller_id_number=123} before the static entries for the on hook agent it will prevail over the default one. If you are using 1.0.4, this feature is only available in trunk or one of the 1.0.5 pre releases. On Mon, Nov 23, 2009 at 4:49 PM, Adam Ford wrote: > Is there any way to set the origination_caller_id for a FIFO outbound > call to an on-hook agent? I can?t find anything in the wiki about a FIFO or > member variable to set this. It seems to be set to ?Queue? by default, and > appears to be hardcoded in the module source. It would be nice to be able > to change per FIFO queue. That way agents that handle multiple companies > can more easily see which queue is calling and answer accordingly. > > > > It is not a big deal, since it does automatically set the > origination_caller_id_number to ?fifo+?. However, depending on > the phone, the caller ID number is not always readily shown, and must be > looked for. > > > > Thanks to anyone who has some insight on this, > > -Adam > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/d986f313/attachment-0002.html From msc at freeswitch.org Mon Nov 23 16:16:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 16:16:03 -0800 Subject: [Freeswitch-users] conference digits and conference control In-Reply-To: <200910151944464068246@gmail.com> References: <200910151944464068246@gmail.com> Message-ID: <87f2f3b90911231616i2ebf92ccs485d66f29f00c4f5@mail.gmail.com> On Thu, Oct 15, 2009 at 3:44 AM, god.nirvana wrote: > hi all > how can i get the digits when users in the conference?? > and,in conference.conf.xml > the "action" will set another > value?e.g:transfer? > thanks > > I'm not sure I understand your question, but the wiki covers actions on keystrokes. If you need the user to dial other digits after the caller control then route the call to an extension that asks the user for input, like with play_and_get_digits, and handle the call accordingly. As far as the question about about setting another value - can you expound upon that a bit? I'm not sure what you're trying to accomplish. -MC P.S. - http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/94a4d0c3/attachment-0002.html From abeka at greatiam.com Mon Nov 23 16:18:29 2009 From: abeka at greatiam.com (Otis) Date: Tue, 24 Nov 2009 00:18:29 +0000 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: <4B0ABC4F.1010103@greatiam.com> References: <4B086689.6080804@greatiam.com> <4B097A89.2050400@greatiam.com> <4B0ABC4F.1010103@greatiam.com> Message-ID: <4B0B2655.4010900@greatiam.com> Has anyone got any suggestion how I can set up a gateway to receive incoming call on extension 1001 please. Any generic conf file will do. my username with my gateway is s=say " qwerty" and password "ytrewq" I have used the intruction from the link below without success. Thanks. Otis wrote: > Hello > > Could anyone point out what I have missed please ? > At the moment I configured a gateway voipuser as described here > : > Any suggestion as to what path I can take will be highly welcome > > Thanks > . > > > > > Sam Abekah-Mensah wrote: >>
Hi Michael >> >> Thanks >> >> I had set it to send incoming calls to extension 1001. This is in the >> file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory. >> The contents are : >> >> >> >> >> >> >> >> >> Is there >> anything wrong with this please ? >> >> Thanks >> >> >> >> Michal Bielicki wrote: >>> >>> Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah: >>> >>>> >>>> I need help as I cannot receive calls through VOIPUSER. This is a >>>> learning setup Attached are my conf files. What is wrong with them >>>> ? When I dial from a landline I get a continuous beep. >>>> >>>> Attached are my gateway and the conf file to transfer. Sopfia >>>> Status is my screen message. I can see a FAIL and cannot make head >>>> or tail of all that message. Hopefully anyone using voipuser or in >>>> fact any of you clever folks can make sense of this. >>>> >>>> Thanks for your time. >>>> >>>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy >>>> RFC2833 Mode! >>>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec >>>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>> 2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec >>>> Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] >>>> 2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec >>>> sofia/external/nobody at 213.166.5.133 PCMA/8000 20 ms 160 samples >>>> 2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf >>>> payload to 101 >>>> 2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 >>>> (sofia/external/nobody at 213.166.5.133) State Change CS_NEW -> CS_INIT >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_INIT >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>>> (sofia/external/nobody at 213.166.5.133) State INIT >>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 >>>> sofia/external/nobody at 213.166.5.133 SOFIA INIT >>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 >>>> (sofia/external/nobody at 213.166.5.133) State Change CS_INIT -> >>>> CS_ROUTING >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>>> (sofia/external/nobody at 213.166.5.133) State INIT going to sleep >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_ROUTING >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 >>>> (sofia/external/nobody at 213.166.5.133) State ROUTING >>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 >>>> sofia/external/nobody at 213.166.5.133 SOFIA ROUTING >>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 >>>> sofia/external/nobody at 213.166.5.133 Standard ROUTING >>>> 2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing >>>> anonymous->abeka in context public >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->unloop] continue=false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (PASS) [unloop] >>>> ${unroll_loops}(true) =~ /^true$/ break=on-false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) [unloop] >>>> ${sip_looped_call}() =~ /^true$/ break=on-false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->outside_call] continue=true >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Absolute Condition >>>> [outside_call] >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Action >>>> set(outside_call=true) >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->call_debug] continue=true >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->public_extensions] continue=false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>> [public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/ >>>> break=on-false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->public_did] continue=false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>> [public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->sip at sip.voipuser.org] continue=false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>> [sip at sip.voipuser.org] destination_number(abeka) =~ /08715042951/ >>>> break=on-false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>> [public->Inbound-abeka at sip.voipuser.org]] continue=false >>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>> [Inbound-abeka at sip.voipuser.org]] destination_number(abeka) =~ >>>> /[08444846450]/ break=on-false >>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 >>>> (sofia/external/nobody at 213.166.5.133) State Change CS_ROUTING -> >>>> CS_EXECUTE >>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send >>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 >>>> (sofia/external/nobody at 213.166.5.133) State ROUTING going to sleep >>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 >>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_EXECUTE >>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 >>>> (sofia/external/nobody at 213.166.5.133) State EXECUTE >>>> 2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 >>>> sofia/external/nobody at 213.166.5.133 SOFIA EXECUTE >>>> 2009-11-21 22:07:15.706658 [DEBUG] switch_core_state_machine.c:151 >>>> sofia/external/nobody at 213.166.5.133 Standard EXECUTE >>>> EXECUTE sofia/external/nobody at 213.166.5.133 set(outside_call=true) >>>> 2009-11-21 22:07:15.728613 [DEBUG] mod_dptools.c:748 >>>> sofia/external/nobody at 213.166.5.133 SET [outside_call]=[true] >>>> 2009-11-21 22:07:15.728613 [NOTICE] switch_core_state_machine.c:179 >>>> Hangup sofia/external/nobody at 213.166.5.133 [CS_EXECUTE] >>>> [NORMAL_CLEARING] >>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_channel.c:1683 Send >>>> signal sofia/external/nobody at 213.166.5.133 [KILL] >>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_session.c:932 Send >>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:491 >>>> (sofia/external/nobody at 213.166.5.133) State EXECUTE going to sleep >>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:398 >>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_HANGUP >>>> 2009-11-21 22:07:15.735830 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/external/nobody at 213.166.5.133) State HANGUP >>>> 2009-11-21 22:07:15.735830 [DEBUG] mod_sofia.c:338 Channel >>>> sofia/external/nobody at 213.166.5.133 hanging up, cause: NORMAL_CLEARING >>>> 2009-11-21 22:07:15.737680 [DEBUG] mod_sofia.c:417 Responding to >>>> INVITE with: 480 >>>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:46 >>>> sofia/external/nobody at 213.166.5.133 Standard HANGUP, cause: >>>> NORMAL_CLEARING >>>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/external/nobody at 213.166.5.133) State HANGUP going to sleep >>>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_state_machine.c:476 >>>> (sofia/external/nobody at 213.166.5.133) State Change CS_HANGUP -> >>>> CS_REPORTING >>>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_session.c:932 Send >>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:398 >>>> (sofia/external/nobody at 213.166.5.133) Running State Change >>>> CS_REPORTING >>>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:612 >>>> (sofia/external/nobody at 213.166.5.133) State REPORTING >>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:53 >>>> sofia/external/nobody at 213.166.5.133 Standard REPORTING, cause: >>>> NORMAL_CLEARING >>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:612 >>>> (sofia/external/nobody at 213.166.5.133) State REPORTING going to sleep >>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:411 >>>> (sofia/external/nobody at 213.166.5.133) State Change CS_REPORTING -> >>>> CS_DESTROY >>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_session.c:1068 >>>> Session 2 (sofia/external/nobody at 213.166.5.133) Locked, Waiting on >>>> external entities >>>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1086 >>>> Session 2 (sofia/external/nobody at 213.166.5.133) Ended >>>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1088 >>>> Close Channel sofia/external/nobody at 213.166.5.133 [CS_DESTROY] >>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>>> (sofia/external/nobody at 213.166.5.133) State DESTROY >>>> 2009-11-21 22:07:15.802636 [DEBUG] mod_sofia.c:255 >>>> sofia/external/nobody at 213.166.5.133 SOFIA DESTROY >>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:60 >>>> sofia/external/nobody at 213.166.5.133 Standard DESTROY >>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>>> (sofia/external/nobody at 213.166.5.133) State DESTROY going to sleep >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> : >>> >>> >>> you seem to have not specified an extension where the call should go to >>> my voipuser.org setup looks like: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I am also surprised that your setup works with a from-domain of >>> sip.voipuser.org >>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>> >>> *Michal Bielicki* >>> HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa >>> t. +48228753290 | f. +48228753291 michal.bielicki at halokwadrat.pl >>> | w. >>> www.halokwadrat.pl >>> >>> >>> >>> *Knowledge & Low Prices. Guaranteed!* >>> >> >> >> >>
> > From john_platts at hotmail.com Mon Nov 23 16:19:59 2009 From: john_platts at hotmail.com (John Platts) Date: Mon, 23 Nov 2009 18:19:59 -0600 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH Message-ID: I actually checked out the latest version of FreeSWITCH in the SVN repository. I have the following configured in /usr/local/freeswitch/conf/dialplan/default.xml: ??? ??????? ??????????? ??????????? ??????????? ??????????? ??????? ??? I have the following configured in /usr/local/freeswitch/conf/vars.xml: ? ? Here is the SIP trace for the failing call: Nov 23 17:55:05.245 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:+19725357722 at ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24 Record-Route: v: SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236 record-route: f: ;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a t: i: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585 CSeq: 1 INVITE Max-Forwards: 16 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208 m: c: application/SDP l: 210 P-Asserted-Identity: Privacy: none v=0 o=- 641026559 641026559 IN IP4 199.173.111.147 s=- c=IN IP4 199.173.111.147 t=0 0 m=audio 33344 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 17:55:05.257 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208 From: ;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a To: Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 17:55:05.257 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357722 at 168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: ;tag=105BD148-201C To: Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 Supported: timer,resource-priority,replaces Min-SE:? 1800 Cisco-Guid: 1961129755-3619819998-2727664095-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259020505 Contact: Expires: 180 Allow-Events: telephone-event Max-Forwards: 15 P-Asserted-Identity: Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 5041 5861 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.147 t=0 0 m=audio 33344 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.147 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 17:55:05.261 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: ;tag=105BD148-201C To: Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Timestamp: 1259020505 0.000345 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Content-Length: 0 Nov 23 17:55:05.309 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 P-Asserted-Identity: "19725357722" Nov 23 17:55:05.309 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208 From: ;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a To: ;tag=105BD180-BD7 Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: Record-Route: , Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 202 P-Asserted-Identity: "19725357722" v=0 o=- 211627 211627 IN IP4 192.168.1.4 s=- c=IN IP4 173.57.44.212 t=0 0 m=audio 0 RTP/AVP 96 101 a=rtpmap:96 G729a/8000 a=fmtp:96 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357722 at 168.75.202.212:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659B6A From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:19725357722 at 168.75.202.212:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 P-Asserted-Identity: Timestamp: 1259020508 CSeq: 102 BYE Reason: Q.850;cause=65 Content-Length: 0 Nov 23 17:55:08.401 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 CSeq: 102 BYE Timestamp: 1259020508 0.000093 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Content-Length: 0 Nov 23 17:55:08.401 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D From: ;tag=105BD148-201C To: ;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306 at 168.75.202.246 CSeq: 102 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 The SIP call actually fails. If I remove the following from /usr/local/freeswitch/conf/dialplan/default.xml: ??? ??????? ??????????? ??????????? ??????????? ??????????? ??????? ??? And I change this line in /usr/local/freeswitch/conf/vars.xml from to And I change this line in /usr/local/freeswitch/conf/vars.xml from to both inbound and outbound calls succeed. Here is a SIP trace of a successful call after I apply the above changes: Nov 23 18:16:51.844 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:+19725357722 at ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb Record-Route: v: SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205 record-route: f: ;tag=dc7-13c4-3da425-183eff65-3da425 t: i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Max-Forwards: 16 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 m: c: application/SDP l: 210 P-Asserted-Identity: Privacy: none v=0 o=- 654094598 654094598 IN IP4 199.173.111.138 s=- c=IN IP4 199.173.111.138 t=0 0 m=audio 31456 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 18:16:51.852 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 From: ;tag=dc7-13c4-3da425-183eff65-3da425 To: Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 18:16:51.856 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357722 at 168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: ;tag=106FC130-1578 To: Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 Supported: timer,resource-priority,replaces Min-SE:? 1800 Cisco-Guid: 2142226702-3620016606-2730940895-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259021811 Contact: Expires: 180 Allow-Events: telephone-event Max-Forwards: 15 P-Asserted-Identity: Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 9668 3852 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.138 t=0 0 m=audio 31456 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.138 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 18:16:51.856 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: ;tag=106FC130-1578 To: Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Timestamp: 1259021811 0.000356 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Content-Length: 0 Nov 23 18:16:51.908 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: ;tag=106FC130-1578 To: ;tag=mXNXUN859rKBa Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 P-Asserted-Identity: "19725357722" Nov 23 18:16:51.908 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 From: ;tag=dc7-13c4-3da425-183eff65-3da425 To: ;tag=106FC168-50D Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: Record-Route: , Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 18:16:54.408 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: ;tag=106FC130-1578 To: ;tag=mXNXUN859rKBa Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 251 P-Asserted-Identity: "19725357722" v=0 o=FreeSWITCH 1259003870 1259003871 IN IP4 168.75.202.212 s=FreeSWITCH c=IN IP4 168.75.202.212 t=0 0 m=audio 17544 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 Nov 23 18:16:54.412 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357722 at 168.75.202.212:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AE109A From: ;tag=106FC130-1578 To: ;tag=mXNXUN859rKBa Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 Nov 23 18:16:54.412 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 From: ;tag=dc7-13c4-3da425-183eff65-3da425 To: ;tag=106FC168-50D Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: Record-Route: , Supported: replaces Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 253 v=0 o=CiscoSystemsSIP-GW-UserAgent 7353 3710 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 168.75.202.212 t=0 0 m=audio 17544 RTP/AVP 0 101 c=IN IP4 168.75.202.212 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Nov 23 18:16:54.492 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: ACK sip:19725357722 at 168.75.202.246:5060 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKb17fac77c446113b9154e16639d30287.6be1820d v: SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK99af095cadf31f4291c6a809ef6a6e03.7c44d9fc;received=65.217.40.205 f: ;tag=dc7-13c4-3da425-183eff65-3da425 t: ;tag=106FC168-50D i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 ACK user-agent: CS2000_NGSS/9.0 Max-Forwards: 68 k: 100rel,replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da428-f0c93cc6-61239372;received=199.173.101.208 m: l: 0 Nov 23 18:17:00.636 CST: %FAN-3-FAN_FAILED: Fan 1 had a rotation error reported. Nov 23 18:17:06.748 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: BYE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKmNaaUQK5vrpXQ Max-Forwards: 70 From: ;tag=mXNXUN859rKBa To: ;tag=106FC130-1578 Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 CSeq: 123392377 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Nov 23 18:17:06.748 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKmNaaUQK5vrpXQ From: ;tag=mXNXUN859rKBa To: ;tag=106FC130-1578 Date: Tue, 24 Nov 2009 00:17:06 GMT Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306 at 168.75.202.246 Server: Cisco-SIPGateway/IOS-12.x CSeq: 123392377 BYE Reason: Q.850;cause=16 Content-Length: 0 Nov 23 18:17:06.752 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:199.173.101.208:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AF4FA From: ;tag=106FC168-50D To: ;tag=dc7-13c4-3da425-183eff65-3da425 Date: Tue, 24 Nov 2009 00:16:54 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Route: , Timestamp: 1259021826 CSeq: 101 BYE Reason: Q.850;cause=16 Content-Length: 0 Nov 23 18:17:06.824 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK f: ;tag=106FC168-50D t: ;tag=dc7-13c4-3da425-183eff65-3da425 i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 101 BYE server: CS2000_NGSS/9.0 allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP 168.75.202.246:5060;received=168.75.202.246;branch=z9hG4bK65AF4FA l: 0 I compared the SIP messaging from the failed call to the SIP messaging from the good call. Both calls are inbound calls. Here is the session description for the failed inbound call: v=0 o=- 211627 211627 IN IP4 192.168.1.4 s=- c=IN IP4 173.57.44.212 t=0 0 m=audio 0 RTP/AVP 96 101 a=rtpmap:96 G729a/8000 a=fmtp:96 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Here is the session description for the good outbound call: v=0 o=FreeSWITCH 1259003870 1259003871 IN IP4 168.75.202.212 s=FreeSWITCH c=IN IP4 168.75.202.212 t=0 0 m=audio 17544 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 Here are the differences between the session descriptors of the failed call and the good call: - The c= line has the correct IP address for the failed call, which was using media bypass - The c= line has the correct IP address for the good call, because the media is being processed by FreeSWITCH in the good call - The m= line does not have the correct RTP port in the failed call - The m= line has the correct RTP port in the good call I noticed that the SDP media descriptor is incorrect in the failed call. Has this problem been fixed? I am running revision 15586 from the FreeSWITCH SVN trunk. _________________________________________________________________ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/ From brian at freeswitch.org Mon Nov 23 16:25:44 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 18:25:44 -0600 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH In-Reply-To: References: Message-ID: <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org> What rev exactly? /b On Nov 23, 2009, at 6:19 PM, John Platts wrote: > > I actually checked out the latest version of FreeSWITCH in the SVN > repository. > > I have the following configured in /usr/local/freeswitch/conf/ > dialplan/default.xml: > > > > > > > > From msc at freeswitch.org Mon Nov 23 16:28:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 16:28:08 -0800 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: <4B0B2655.4010900@greatiam.com> References: <4B086689.6080804@greatiam.com> <4B097A89.2050400@greatiam.com> <4B0ABC4F.1010103@greatiam.com> <4B0B2655.4010900@greatiam.com> Message-ID: <87f2f3b90911231628t7e44986ar10451f26a86e7df6@mail.gmail.com> On Mon, Nov 23, 2009 at 4:18 PM, Otis wrote: > Has anyone got any suggestion how I can set up a gateway to receive > incoming call on extension 1001 please. > > Any generic conf file will do. my username with my gateway is s=say " > qwerty" and password "ytrewq" > > I have used the intruction from the link below without success. > > Thanks. > > Get a debug log of an incoming call. Most likely it is hitting the public context and you don't have it properly routed. Look in conf/dialplan/public.xml for an example of how to route to a specific extension. Another example is in conf/dialplan/public/00_inbound_did.xml. The trick is to know what to match on in your condition tag. In a pinch you could route all incoming calls to the info app and then make a test call to see what's coming down the line. Just be sure to turn it off when you're done testing!! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/15d909d0/attachment-0002.html From lists at redbonez.net Mon Nov 23 16:43:42 2009 From: lists at redbonez.net (Adam Ford) Date: Mon, 23 Nov 2009 17:43:42 -0700 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> Message-ID: <009201ca6c9f$2d18cb80$874a6280$@net> I actually tried that, as a guess, based on the configuration output of fifo list. However I am running a tarball release of 1.0.4, which would explain why it did not work for me. I appreciate the feedback, and will make a note to implement this when I update my installation. Are the svn-trunk updates pretty solid? I have not attempted an update yet, as it is a production system. Thanks, -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, November 23, 2009 5:13 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FIFO Orgination_caller_id if you add {origination_caller_id_name=foo,origination_caller_id_number=123} before the static entries for the on hook agent it will prevail over the default one. If you are using 1.0.4, this feature is only available in trunk or one of the 1.0.5 pre releases. On Mon, Nov 23, 2009 at 4:49 PM, Adam Ford wrote: Is there any way to set the origination_caller_id for a FIFO outbound call to an on-hook agent? I can't find anything in the wiki about a FIFO or member variable to set this. It seems to be set to 'Queue' by default, and appears to be hardcoded in the module source. It would be nice to be able to change per FIFO queue. That way agents that handle multiple companies can more easily see which queue is calling and answer accordingly. It is not a big deal, since it does automatically set the origination_caller_id_number to 'fifo+'. However, depending on the phone, the caller ID number is not always readily shown, and must be looked for. Thanks to anyone who has some insight on this, -Adam _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/2fe06353/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 23 16:48:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Nov 2009 18:48:51 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <4B0B005E.4080202@xpirio.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> <4B0B005E.4080202@xpirio.com> Message-ID: <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> You forgot to set freeswitch to debug loglevel You need both of the following: console loglevel debug sofia profile internal siptrace on 2009/11/23 Christian L?schenkohl > sorry about wasting your time (wasn't my intent) > > the log is at http://pastebin.freeswitch.org/11240 > i called 5214448370068 (also other calls are in the log) > > they now have changed 180 to 183 on the sonus, but makes no difference here > > br > > On 2009-11-23 22:07, Anthony Minessale wrote: > > do you have the ringback variable set on the channel? > > if so it will cause 180 to attempt to play inband ringback indication > > > > I have nothing left to say because I asked for the whole log with the > > siptrace enables not just 5 lines of it. > > If you still want help, give me the log to examine and I will tell you > > what your problem is. > > > > > > > > 2009/11/23 Christian L?schenkohl > > > > > > thany ou for your answer > > > > we use g729 on all our other connections in passthrough mode and it > > also doesn't work with alaw. > > so i don't think it's related to this. > > > > br > > > > > > On 2009-11-23 20:48, Brian West wrote: > > > Well its also G729 so I suspect you don't have G729 > > > > > > /b > > > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > > > >> hi > > >> > > >> our freeswitch server has to talk to a sonus ip-switch > > >> when we want to setup a call we do get a "100 Trying" and then a > > >> "180 Ringing" > > >> within the "180 Ringing" we get a sdp with "a=sendonly" then our > > >> freeswitch > > >> quits with a CANCEL message. > > >> i simply don't get why our freeswitch aborts the session - i > think > > >> it would work > > >> if no "a=sendonly" would be present in the sdp. > > >> > > >> my technical contact doesn't want to switch 180 to 183 on the > sonus > > >> side - this would > > >> also work (i think). in fact he says that 180 ringing is vaild, > he > > >> isn't that wrong in > > >> this case. > > >> > > >> our freeswitch works in proxy mode, we do use trunk 15396 > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > > >> > > >> 92.63.208.36 - freeswitch > > >> 38.105.229.100 - sonus > > >> > > >> br > > >> > > >> -- > > >> Ing. Christian L?schenkohl > > >> Technische Leitung, Forschung& Entwicklung VoIP > > >> > > >> xpirio > > >> Telekommunikation& Service GmbH > > >> Lakeside B04 > > >> 9020 Klagenfurt > > >> Austria > > >> > > >> T +43 (0) 5 77 11 - 1000 > > >> F +43 (0) 5 77 11 - 1002 > > >> E christian.loeschenkohl at xpirio.com > > > > >> > > >> _______________________________________________ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch- > > >> users > > >> http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung & Entwicklung VoIP > > > > xpirio > > Telekommunikation & Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/5d1a239e/attachment-0002.html From dujinfang at gmail.com Mon Nov 23 16:58:39 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Nov 2009 08:58:39 +0800 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> Message-ID: <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> And because it's static string for on-hook members, it's hard to set dynamically. For now, I'm using a callback way - whenever the sip client answered the call, it fetch the real connected number from a http server. That's not ideal because not only it add the complexity but also the callee have no idea what the number is before answer. The problem for on-hook agent is that it call the agent first, and then get one customer from the fifo queue, so it is not possible to let the agent know the real caller-id before answer. Ideas? 2009/11/24 Anthony Minessale > if you add > {origination_caller_id_name=foo,origination_caller_id_number=123} before the > static entries for the on hook agent it will prevail over the default one. > > If you are using 1.0.4, this feature is only available in trunk or one of > the 1.0.5 pre releases. > > > On Mon, Nov 23, 2009 at 4:49 PM, Adam Ford wrote: > >> Is there any way to set the origination_caller_id for a FIFO outbound >> call to an on-hook agent? I can?t find anything in the wiki about a FIFO or >> member variable to set this. It seems to be set to ?Queue? by default, and >> appears to be hardcoded in the module source. It would be nice to be able >> to change per FIFO queue. That way agents that handle multiple companies >> can more easily see which queue is calling and answer accordingly. >> >> >> >> It is not a big deal, since it does automatically set the >> origination_caller_id_number to ?fifo+?. However, depending on >> the phone, the caller ID number is not always readily shown, and must be >> looked for. >> >> >> >> Thanks to anyone who has some insight on this, >> >> -Adam >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/ccb99647/attachment-0002.html From msc at freeswitch.org Mon Nov 23 17:01:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 17:01:51 -0800 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <009201ca6c9f$2d18cb80$874a6280$@net> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <009201ca6c9f$2d18cb80$874a6280$@net> Message-ID: <87f2f3b90911231701n7dd58b6fhe354e04890ada239@mail.gmail.com> On Mon, Nov 23, 2009 at 4:43 PM, Adam Ford wrote: > I actually tried that, as a guess, based on the configuration output of > fifo list. However I am running a tarball release of 1.0.4, which would > explain why it did not work for me. > > > > I appreciate the feedback, and will make a note to implement this when I > update my installation. Are the svn-trunk updates pretty solid? I have not > attempted an update yet, as it is a production system. > > > Trunk has been very solid with a few minor exceptions. Best bet is to back up everything and do the upgrade during down time. If you have a test system that you can use as a sandbox that would be even better... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/950fef76/attachment-0002.html From msc at freeswitch.org Mon Nov 23 17:07:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 17:07:45 -0800 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> Message-ID: <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> On Mon, Nov 23, 2009 at 4:58 PM, Seven Du wrote: > And because it's static string for on-hook members, it's hard to set > dynamically. For now, I'm using a callback way - whenever the sip client > answered the call, it fetch the real connected number from a http server. > That's not ideal because not only it add the complexity but also the callee > have no idea what the number is before answer. > > The problem for on-hook agent is that it call the agent first, and then get > one customer from the fifo queue, so it is not possible to let the agent > know the real caller-id before answer. Ideas? > > Tony and Brian were discussing this today. They bring up a really good point: do you want to risk having calls remain on hold as they bounce around looking for an agent? This can happen if you pre-determine which caller goes to which agent and the agent doesn't answer. I do understand why this feature matters to many people - it's how old school ACD systems work. However, mod_fifo is more efficient. It's hard to justify decreasing call routing efficiency in order to display the caller's info to the on-hook agent prior to answering. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/0ca1c6bf/attachment-0002.html From brian at freeswitch.org Mon Nov 23 17:12:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 19:12:20 -0600 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> Message-ID: You do realize that the whole concept is OLD skewl. You should be popping this info via external resources when the agent is bridged to the caller and the info is there before they are done saying "thanks for calling spacely sprockets, this is George how may I help you .... " /b On Nov 23, 2009, at 7:07 PM, Michael Collins wrote: > Tony and Brian were discussing this today. They bring up a really > good point: do you want to risk having calls remain on hold as they > bounce around looking for an agent? This can happen if you pre- > determine which caller goes to which agent and the agent doesn't > answer. I do understand why this feature matters to many people - > it's how old school ACD systems work. However, mod_fifo is more > efficient. It's hard to justify decreasing call routing efficiency > in order to display the caller's info to the on-hook agent prior to > answering. > > -MC From msc at freeswitch.org Mon Nov 23 17:18:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Nov 2009 17:18:08 -0800 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> Message-ID: <87f2f3b90911231718q46c51982j36067f1627dd759c@mail.gmail.com> On Mon, Nov 23, 2009 at 5:12 PM, Brian West wrote: > You do realize that the whole concept is OLD skewl. You should be > popping this info via external resources when the agent is bridged to > the caller and the info is there before they are done saying "thanks > for calling spacely sprockets, this is George how may I help you .... " > > /b > > Agreed! Screen pop should be easy in the 21st Century. If it's not then you've got MUCH bigger problems than caller ID being delivered to your FIFO agents... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/d04f1ea1/attachment-0002.html From dujinfang at gmail.com Mon Nov 23 17:24:05 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Nov 2009 09:24:05 +0800 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> Message-ID: <23f91030911231724nc7dd28cs6a8c1ae738b3aae8@mail.gmail.com> Yes, that's what we are doing. 2009/11/24 Brian West > You do realize that the whole concept is OLD skewl. You should be > popping this info via external resources when the agent is bridged to > the caller and the info is there before they are done saying "thanks > for calling spacely sprockets, this is George how may I help you .... " > > /b > > > On Nov 23, 2009, at 7:07 PM, Michael Collins wrote: > > > Tony and Brian were discussing this today. They bring up a really > > good point: do you want to risk having calls remain on hold as they > > bounce around looking for an agent? This can happen if you pre- > > determine which caller goes to which agent and the agent doesn't > > answer. I do understand why this feature matters to many people - > > it's how old school ACD systems work. However, mod_fifo is more > > efficient. It's hard to justify decreasing call routing efficiency > > in order to display the caller's info to the on-hook agent prior to > > answering. > > > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b7cac2a2/attachment-0002.html From lists at redbonez.net Mon Nov 23 17:26:47 2009 From: lists at redbonez.net (Adam Ford) Date: Mon, 23 Nov 2009 18:26:47 -0700 Subject: [Freeswitch-users] Business/holiday hours routing Message-ID: <00be01ca6ca5$31f64ff0$95e2efd0$@net> Is there a standard module for FreeSWITCH out there that people use for routing calls based on business hours and a holiday schedule? Or is everyone just creating their own in the XML dialplan?(which seems pretty simple) I can't seem to find anything on the wiki, but might just be searching for the wrong thing. I am relatively new at FreeSWITCH and would rather follow what the community has decided is the best practice, instead of trying to reinvent the wheel myself. Thanks for any input, -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/0e7b0f5c/attachment-0002.html From rob4manhere at gmail.com Mon Nov 23 17:30:03 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Mon, 23 Nov 2009 19:30:03 -0600 Subject: [Freeswitch-users] Memory leak with mod_local_stream In-Reply-To: <3BF17413-EC48-4691-8C99-61CC4661E2CA@jerris.com> References: <3BF17413-EC48-4691-8C99-61CC4661E2CA@jerris.com> Message-ID: <009439B2-B025-4C26-8407-A212A762A7F9@gmail.com> Ignore this. I'm an idiot. Rob :) On Nov 23, 2009, at 4:15 PM, Michael Jerris wrote: > That rev should have fixed that memory leak, could you test > mod_local_stream.c from rev 15430 (http://fisheye.freeswitch.org/browse/ > ~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/ > mod_local_stream.c) with your current fs version to confirm this is > the cause please? > > Mike > > > On Nov 23, 2009, at 4:53 PM, Rob Forman wrote: > >> Hey guys, >> >> Having a problem with mod_local_stream. >> >> I recently did a "make current" from 15334 to the latest trunk >> (15630). After restarting, there now appears to be a memory leak. >> On >> a test system (CentOS 5.4, 64-bit) with no calls or registrations, >> Freeswitch gradually consumes all of the host memory (rate of about >> 200K/second), then swaps out, eventually rendering the system >> useless. >> >> I isolated it to mod_local_stream. If I unload mod_local_stream, the >> memory use stops climbing. If I re-load mod_local_stream, it starts >> again. >> >> >> I would submit the logs except they aren't any besides it starting. >> The system is just sitting there idle. Even valgrind didn't show >> much >> (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I >> ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- >> check=full --leak-resolution=high --show-reachable=yes .libs/ >> freeswitch -vg >> >> Questions: >> * has anyone else seen this? >> * what is the best way I can assist troubleshooting this? >> >> I saw a patch to mod_local_stream (rev 15431) a few weeks back. >> Could >> that have anything to do with it? >> >> Rob >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Mon Nov 23 17:43:13 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Nov 2009 09:43:13 +0800 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <00be01ca6ca5$31f64ff0$95e2efd0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> Message-ID: <23f91030911231743m35928483vfcac709fce1cae4e@mail.gmail.com> XML has basic conditioning, but lua rocks. -- Time condition for sales 1 --session:setAutoHangup(false) function do_transfer(extn) --print(extn) session:transfer(extn, "XML", "sales") end now = os.date("%H:%M") w = tonumber(os.date("%w")) if w >= 1 and w <=5 then if ( now >= "09:00" and now < "20:30" ) then do_transfer("sales_fifo_1") elseif ( now >= "20:30" and now < "22:30" ) then do_transfer("sales_fifo_2") else do_transfer("sales_fifo_cellphone") end else if ( now >= "10:00" and now < "19:00" ) then do_transfer("sales_fifo_1") elseif (now >= "20:00" and now < "22:30" ) then do_transfer("sales_fifo_2") else do_transfer("sales_fifo_cellphone") end end 2009/11/24 Adam Ford > Is there a standard module for FreeSWITCH out there that people use for > routing calls based on business hours and a holiday schedule? Or is everyone > just creating their own in the XML dialplan?(which seems pretty simple) > > > > I can?t seem to find anything on the wiki, but might just be searching for > the wrong thing. I am relatively new at FreeSWITCH and would rather follow > what the community has decided is the best practice, instead of trying to > reinvent the wheel myself. > > > > Thanks for any input, > > -Adam > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/98e25335/attachment-0002.html From brian at freeswitch.org Mon Nov 23 17:44:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 19:44:40 -0600 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <00be01ca6ca5$31f64ff0$95e2efd0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> Message-ID: <825B66B6-EE5B-4CA0-8CEF-3CF6A0E7789C@freeswitch.org> Please see default.xml dialplan at the top in SVN. /b On Nov 23, 2009, at 7:26 PM, Adam Ford wrote: > Is there a standard module for FreeSWITCH out there that people use > for routing calls based on business hours and a holiday schedule? Or > is everyone just creating their own in the XML dialplan?(which seems > pretty simple) > > I can?t seem to find anything on the wiki, but might just be > searching for the wrong thing. I am relatively new at FreeSWITCH > and would rather follow what the community has decided is the best > practice, instead of trying to reinvent the wheel myself. > > Thanks for any input, > -Adam > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/a73e6e4b/attachment-0002.html From brian at freeswitch.org Mon Nov 23 17:45:46 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 19:45:46 -0600 Subject: [Freeswitch-users] FIFO Orgination_caller_id In-Reply-To: <23f91030911231724nc7dd28cs6a8c1ae738b3aae8@mail.gmail.com> References: <005701ca6c8f$28eaa570$7abff050$@net> <191c3a030911231613r7207574bode8b53cd4b929d11@mail.gmail.com> <23f91030911231658g608aacb4pd5c32d89aa46b255@mail.gmail.com> <87f2f3b90911231707m1939a49bkc944364781b71a4@mail.gmail.com> <23f91030911231724nc7dd28cs6a8c1ae738b3aae8@mail.gmail.com> Message-ID: Its the proper way to do it. And do you know the second you're bridged... if you have a phone that isn't daft... snom or polycom... the display will update to the person your bridged to. /b On Nov 23, 2009, at 7:24 PM, Seven Du wrote: > Yes, that's what we are doing. > > 2009/11/24 Brian West > You do realize that the whole concept is OLD skewl. You should be > popping this info via external resources when the agent is bridged to > the caller and the info is there before they are done saying "thanks > for calling spacely sprockets, this is George how may I help > you .... " > > /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/1106afa6/attachment-0002.html From andrew at hijacked.us Mon Nov 23 17:48:08 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 23 Nov 2009 20:48:08 -0500 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <00be01ca6ca5$31f64ff0$95e2efd0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> Message-ID: <20091124014808.GB3298@hijacked.us> On Mon, Nov 23, 2009 at 06:26:47PM -0700, Adam Ford wrote: > Is there a standard module for FreeSWITCH out there that people use for > routing calls based on business hours and a holiday schedule? Or is everyone > just creating their own in the XML dialplan?(which seems pretty simple) > > > > I can't seem to find anything on the wiki, but might just be searching for > the wrong thing. I am relatively new at FreeSWITCH and would rather follow > what the community has decided is the best practice, instead of trying to > reinvent the wheel myself. > I assume you've seen this: http://wiki.freeswitch.org/wiki/Time_of_Day_Routing I have a patch that'll let you specify the nth day of the nth week via wday="3,4" for the 4th tuesday in the month. This willl let you do vacations like thanksgiving, MLK day, etc as well. Andrew From lists at redbonez.net Mon Nov 23 18:13:48 2009 From: lists at redbonez.net (Adam Ford) Date: Mon, 23 Nov 2009 19:13:48 -0700 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <20091124014808.GB3298@hijacked.us> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> Message-ID: <00e101ca6cab$c3525240$49f6f6c0$@net> Yes, that wiki page is what I was referring to when I said it seems simple enough to do it in the XML dialplan. So other than the one lua suggestion, it seems the majority say XML is the way to go eh? -Adam Andrew - that does sound like a useful patch, is it in svn or unpublished? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Thompson Sent: Monday, November 23, 2009 6:48 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Business/holiday hours routing On Mon, Nov 23, 2009 at 06:26:47PM -0700, Adam Ford wrote: > Is there a standard module for FreeSWITCH out there that people use for > routing calls based on business hours and a holiday schedule? Or is everyone > just creating their own in the XML dialplan?(which seems pretty simple) > > > > I can't seem to find anything on the wiki, but might just be searching for > the wrong thing. I am relatively new at FreeSWITCH and would rather follow > what the community has decided is the best practice, instead of trying to > reinvent the wheel myself. > I assume you've seen this: http://wiki.freeswitch.org/wiki/Time_of_Day_Routing I have a patch that'll let you specify the nth day of the nth week via wday="3,4" for the 4th tuesday in the month. This willl let you do vacations like thanksgiving, MLK day, etc as well. Andrew _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From james at talent.com.au Mon Nov 23 18:16:34 2009 From: james at talent.com.au (James Budge) Date: Tue, 24 Nov 2009 12:16:34 +1000 Subject: [Freeswitch-users] os x compile failure Message-ID: The compile fails after this. i686-apple-darwin10-gcc-4.2.1: -bundle not allowed with -dynamiclib From brian at freeswitch.org Mon Nov 23 18:17:46 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 20:17:46 -0600 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <00e101ca6cab$c3525240$49f6f6c0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> <00e101ca6cab$c3525240$49f6f6c0$@net> Message-ID: <21CB5F92-98DE-4622-ADC5-013462A93BD2@freeswitch.org> He's working on it for SVN... I recommended the format and to add the phases of the moon and zodiac signs just for giggles. /b On Nov 23, 2009, at 8:13 PM, Adam Ford wrote: > > Andrew - that does sound like a useful patch, is it in svn or > unpublished? From brian at freeswitch.org Mon Nov 23 18:20:30 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 20:20:30 -0600 Subject: [Freeswitch-users] os x compile failure In-Reply-To: References: Message-ID: While I love that people report issues... can you elaborate on things a bit? OS X version? CPU? SVN Rev? /b On Nov 23, 2009, at 8:16 PM, James Budge wrote: > The compile fails after this. > > i686-apple-darwin10-gcc-4.2.1: -bundle not allowed with -dynamiclib From Prometheus001 at gmx.net Mon Nov 23 18:37:11 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 24 Nov 2009 03:37:11 +0100 Subject: [Freeswitch-users] Problems with Voicemail In-Reply-To: <4B099098.2040408@gmx.net> References: <4B073ACD.1090708@gmx.net> <976A0342-4F4B-4035-9201-D56F8625AE12@freeswitch.org> <4B07D999.4040004@gmx.net> <4B099098.2040408@gmx.net> Message-ID: <4B0B46D7.1050609@gmx.net> I sorted it out. Something went wrong with the odbc database. I deleted the voicemail database tables, restarted FS and let FS create the tables again. Now it works. I can even share the voicemails across 2 Freeswitch boxes. Best regards Peter Peter P GMX schrieb: > I now created a file inbox.PCMA and get the following: > > * inbox.PCMA is played > * the recorded voive mail file is not played (FS does not even try > to do that) > * then I hear > o "to listen to the recording press 1" > o "to save the recording press 2" > o ... > > Here's the debug output > 2009-11-22 20:17:43.701098 [DEBUG] switch_core_io.c:660 > sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] > 2009-11-22 20:17:44.278600 [DEBUG] switch_ivr_play_say.c:1428 done > playing file > 2009-11-22 20:17:44.386776 [INFO] mod_native_file.c:82 Opening File > [/usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA] 8000hz > 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1428 done > playing file > 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:118 No language > specified - Using [en] > 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-listen_to_recording.wav] (en:en) > 2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1136 Codec > Activated L16 at 8000hz 1 channels 20ms > 2009-11-22 20:17:45.201099 [DEBUG] switch_core_io.c:660 > sofia/internal/200 at sip1.mydomain.com receive message [TRANSCODING_NECESSARY] > 2009-11-22 20:17:46.419933 [DEBUG] switch_ivr_play_say.c:1428 done > playing file > > nGrepping port 3306 I can see that the correct filenames are retrieved > from the mysql/odbc database: > 1258894746.0.200.sip1.mydomain.com$d11c2a74-d766-11de-997b-bd7aecdc2a16.Gor > Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_c57a5e84-d766-11de-997b-bd7aecdc2a16.wav.4..B_NORMAL.....47 > 1258897120.0.200.sip1.mydomain.com$580dafee-d76c-11de-84d4-a1cd7fa320b3.Gor > Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_4d484a7e-d76c-11de-84d4-a1cd7fa320b3.wav.5..B_NORMAL......... > Both filenames can be read. > > Best regards > Peter > > Peter P GMX schrieb: > >> I installed all sounds from SVN, but >> >> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA >> >> isn't there. I checked another, older installation and couldn't this >> file either. >> >> I think that freeswitch tries to build a sound path for the file to be >> played, and some parts of the path are missing. >> I expect it would play a recorded message at that time in >> /usr/local/freeswitch/storage/voicemail/default/${domain} and the >> defined format is "wav" not pcma. >> >> I also set "storage_dir" explicitely in the voicemail configs,but this >> also didn't help. >> >> Best regards >> Peter >> >> >> Brian West schrieb: >> >> >>> I'm going to venture to guess maybe the file was recorded in a >>> different codec and NOT pcma? >>> >>> /b >>> >>> On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote: >>> >>> >>> >>> >>>> 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / >>>> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA >>>> >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From james at talent.com.au Mon Nov 23 18:44:30 2009 From: james at talent.com.au (James Budge) Date: Tue, 24 Nov 2009 12:44:30 +1000 Subject: [Freeswitch-users] os x compile failure In-Reply-To: References: Message-ID: <40B8DB3D-2F66-45BC-BB9E-9B773B707FA3@talent.com.au> 2GHz Intel Core Duo OS 10.6.2 Xcode 3.2.1 Updated to revision 15648. On 24/11/2009, at 12:20 PM, Brian West wrote: > While I love that people report issues... can you elaborate on things > a bit? OS X version? CPU? SVN Rev? > > /b > > On Nov 23, 2009, at 8:16 PM, James Budge wrote: > >> The compile fails after this. >> >> i686-apple-darwin10-gcc-4.2.1: -bundle not allowed with -dynamiclib > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Mon Nov 23 18:45:45 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Nov 2009 10:45:45 +0800 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <00e101ca6cab$c3525240$49f6f6c0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> <00e101ca6cab$c3525240$49f6f6c0$@net> Message-ID: <23f91030911231845w4247b7fcmc05f920a81dd288d@mail.gmail.com> 2009/11/24 Adam Ford > Yes, that wiki page is what I was referring to when I said it seems simple > enough to do it in the XML dialplan. So other than the one lua suggestion, > it seems the majority say XML is the way to go eh? > Yeah, I made that lua script when the latest XML time routing feature was unavailable. > > -Adam > > Andrew - that does sound like a useful patch, is it in svn or unpublished? > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew > Thompson > Sent: Monday, November 23, 2009 6:48 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Business/holiday hours routing > > On Mon, Nov 23, 2009 at 06:26:47PM -0700, Adam Ford wrote: > > Is there a standard module for FreeSWITCH out there that people use for > > routing calls based on business hours and a holiday schedule? Or is > everyone > > just creating their own in the XML dialplan?(which seems pretty simple) > > > > > > > > I can't seem to find anything on the wiki, but might just be searching > for > > the wrong thing. I am relatively new at FreeSWITCH and would rather > follow > > what the community has decided is the best practice, instead of trying to > > reinvent the wheel myself. > > > I assume you've seen this: > > http://wiki.freeswitch.org/wiki/Time_of_Day_Routing > > I have a patch that'll let you specify the nth day of the nth week via > wday="3,4" for the 4th tuesday in the month. This willl let you do > vacations like thanksgiving, MLK day, etc as well. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/791e0443/attachment-0002.html From brian at freeswitch.org Mon Nov 23 18:47:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Nov 2009 20:47:27 -0600 Subject: [Freeswitch-users] os x compile failure In-Reply-To: <40B8DB3D-2F66-45BC-BB9E-9B773B707FA3@talent.com.au> References: <40B8DB3D-2F66-45BC-BB9E-9B773B707FA3@talent.com.au> Message-ID: <8CD390C6-D194-483C-8A0A-732B5BFFCE09@freeswitch.org> Ok 32bit... we are currently working on that as I type. /b On Nov 23, 2009, at 8:44 PM, James Budge wrote: > 2GHz Intel Core Duo > > OS 10.6.2 > > Xcode 3.2.1 > > Updated to revision 15648. From john_platts at hotmail.com Mon Nov 23 20:33:03 2009 From: john_platts at hotmail.com (John Platts) Date: Mon, 23 Nov 2009 22:33:03 -0600 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH In-Reply-To: <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org> References: , <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org> Message-ID: I was using revision 15586. ---------------------------------------- > From: brian at freeswitch.org > Date: Mon, 23 Nov 2009 18:25:44 -0600 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH > > What rev exactly? > > /b > > On Nov 23, 2009, at 6:19 PM, John Platts wrote: > >> >> I actually checked out the latest version of FreeSWITCH in the SVN >> repository. >> >> I have the following configured in /usr/local/freeswitch/conf/ >> dialplan/default.xml: >> >> >> >> >> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/ From talk2ram at gmail.com Mon Nov 23 21:54:19 2009 From: talk2ram at gmail.com (ram) Date: Mon, 23 Nov 2009 21:54:19 -0800 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <4B0AD655.9070507@greatiam.com> References: <4B0AA8B6.2080305@greatiam.com> <4B0AD655.9070507@greatiam.com> Message-ID: On Mon, Nov 23, 2009 at 10:37 AM, Otis wrote: > Thanks. > > I have to get a centos box I guess. > > Much appreciated > > Samuel 'Otis' > > > how about trying Fusionpbx.com ( GUI) Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/b7943aca/attachment-0002.html From thangappan143 at gmail.com Mon Nov 23 21:56:44 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Tue, 24 Nov 2009 11:26:44 +0530 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> Message-ID: <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> The reason for waiting only for DTMF event is to handle the time outs in the IVR concept like response and inter digit time out. Using our own logic we 10 voice files in each play back if the voice files are more than 10. Now it works fine. Now the new problem has been raised. The problem is we are filtering only for DTMF events but we are getting COMMAND event . Because of this the DTMF digits are missing at the time . I am not able to proceed further. We are in the critical situation. Why this command event is occurring? How can I restrict this? What are the information it has? How can I get all the information in it ? ( If command event has info) Help me............ On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: > I am waiting only for DTMF events. That's why I am setting freeswitch > variable for knowing whether the playback has done. > > My question is "why this freeswitch variable is not setting properly when I > play back more than 10 files using playback_delimiter option?". > > When I play back lesser than ten voice files the variable has been set > properly. What could be the reason? > > > > ---------- Forwarded message ---------- > From: Thangappan.M > Date: Sat, Nov 21, 2009 at 2:52 PM > Subject: Problem while playing more than 10 voice files using playback > To: freeswitch-users > > > Dear all, > > I am in the process of implementing IVR using event outbound > socket (async mode). > I have implemented using Perl language. > > I did the following steps: > => Set the playback_delimiter variable > => Set the playback_sleep_val variable > => Set the event lock as true > => Set the freeswitch ( my own) variable as zero > => Wait in the loop until the variable is been set as > zero > => Playback the voice files ( Here I combined the voice > files with the delimiter value if more than one voice files are there) > => Set the freeswitch(my own) variable as true ( This is > used to identify whether the voice files are played > successfully). > => Wait in the loop until the variable is been set as > one. > => Set the Event lock as false > > => Trying to get the DTMF digits ( Have a assurance > that all the voice files are played). > > The problem is, > > The above steps are working fine when the voice file count is > lesser than or equal to 10. After the voice files are played only the > variable(my own freeswitch) is set. Based on the variable I am doing further > things. > > But when I tried to give the voice files count of more than 10 > the variable has been set while starting to play back the first voice file > itself . Because of this I am not able to proceed further. > > *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* > > *NOTE*: I also referred mod_file_string documentation. In that they > specified 128 files can be used to play back the voice files using > playback_delimiter option. > > Please help me................? > Thanks in advance. > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b475fc39/attachment-0002.html From yudha2008 at gmail.com Mon Nov 23 22:03:58 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 24 Nov 2009 11:33:58 +0530 Subject: [Freeswitch-users] DTMF javasript Message-ID: * Hi,* * * *I want to check value given to the javascript with conditions whether it is voicefile, extension or mobile Number when i press the dtmf value.* * * *Steps i need to check in javascript:* * * *When i Press the DTMF value 1 it should check the 3 condition* * * If the Value for argv[2]=vfsurya means it is a voice file so it should play the Voice file *If the Value for argv[2]=1001 means it is a extension. The call should Bridge the extension* *If the Value for argv[2]=9841799874 means it is a Mobile number. The call should Bridge the Mobile number* * * *var exit = false;* *var dtmf_digits = "";* *var repeat = 0;* *var argv[2]=vfsurya; // or var argv[2]=1001 or var argv[2]=Mobile Number* * * * * *function onInput( session, type, data, arg ) * *{* * if ( type == "dtmf" ) * * {* * console_log( "info", "Got digit " + data.digit + "\n" );* * if ( data.digit == "1" ) * * **{* * if(argv[2].startswith("vf"))* * **{* * **var voice2=voice.substring(2)+"
"* * **session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/"+voice2+".wav", onInput );* * **}* * **else if(argv[2].length==4)* * **{* * **console_log( "info", "Got voicefile " + argv[2] + "\n" );* * **session.execute("bridge", "sofia/internal/"+argv[2]+"%192.168.1.2", onInput ); * * **}* * **else* * **{* * **session.execute("bridge", "sofia/default/sip:"+argv[2]+"@ 192.168.1.135:5066", onInput ); * * **}* * }* * }* *}* * * *But if 1 is pressed there is no event trigger but it get the dtmf value as 1 in freeswitch console. * * * *can any one specify what is the error or correct me where i am wrong.* * -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/2da09a74/attachment-0002.html From mike at jerris.com Mon Nov 23 22:23:00 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 01:23:00 -0500 Subject: [Freeswitch-users] Requesting testing. Message-ID: I have done quite a few changes to the build system and correcting build problems and other platform specific problems the last few days. Could everyone on the list please take a little time out of their day and do a clean fresh svn trunk checkout of FreeSWITCH and do a full build and report any errors you encounter (if not already reported) to http://jira.freeswitch.org. We have fixed things for many platforms including bsd, solaris, linux, and especially issues on OS X. Please try these out to make sure all works. Thanks Mike From andrew at hijacked.us Mon Nov 23 22:45:09 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 24 Nov 2009 01:45:09 -0500 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <21CB5F92-98DE-4622-ADC5-013462A93BD2@freeswitch.org> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> <00e101ca6cab$c3525240$49f6f6c0$@net> <21CB5F92-98DE-4622-ADC5-013462A93BD2@freeswitch.org> Message-ID: <20091124064509.GA6360@hijacked.us> On Mon, Nov 23, 2009 at 08:17:46PM -0600, Brian West wrote: > He's working on it for SVN... I recommended the format and to add the > phases of the moon and zodiac signs just for giggles. > I'll probably get a patch in this week (or early next) I'm thinking of changing the format so that "week of month" becomes its own value so you could compare against mweek as well as wday so thanksgiving + extension becomes something like If I really get ambitious I'd also like to allow wday="mon-fri" so I don't always forget that days are 1-indexed from sunday :) Andrew From velu.technical at gmail.com Mon Nov 23 23:22:30 2009 From: velu.technical at gmail.com (velusamy velu) Date: Tue, 24 Nov 2009 12:52:30 +0530 Subject: [Freeswitch-users] DTMF Event is not coming while using playback terminators. Message-ID: <1452e2980911232322j106fe5eoe3efad59199f36e4@mail.gmail.com> Dear All, I am using Perl ESL::IVR module to develop a simple IVR. I have filtered DTMF events. I have also set playback_terminators to cut the playback when giving the digits. I have faced problem that DTMF event has not come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. Why the 'COMMAND' event came event filter is on? How to avoid this event in ESL? Thanks, Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b777c286/attachment-0002.html From siniypin at gmail.com Mon Nov 23 23:49:48 2009 From: siniypin at gmail.com (RobertT) Date: Tue, 24 Nov 2009 10:49:48 +0300 Subject: [Freeswitch-users] Requesting testing. In-Reply-To: References: Message-ID: <2160023e0911232349h6ef3a1f5m13c2cb21e12a70d2@mail.gmail.com> I've a problem building FS rev 15630 on Windows. One of mod_pocketsphinx related projects lack a code file. Regards, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/9ea7a70d/attachment-0002.html From mike at jerris.com Tue Nov 24 00:29:37 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 03:29:37 -0500 Subject: [Freeswitch-users] Requesting testing. In-Reply-To: <2160023e0911232349h6ef3a1f5m13c2cb21e12a70d2@mail.gmail.com> References: <2160023e0911232349h6ef3a1f5m13c2cb21e12a70d2@mail.gmail.com> Message-ID: <6C8B0117-27F5-4FEF-926A-6E6A97AA0309@jerris.com> please follow the procedures http://wiki.freeswitch.org/wiki/Reporting_Bugs to report bugs at http://jira.freeswitch.org. Also, you will need to provide far more detail than in this email for anyone to be able to have a possibility of fixing it. Please include details such as, what file is missing, what errors and warnings you get. How to reproduce it and preferably a patch to correct the problem if you can create one. Mike On Nov 24, 2009, at 2:49 AM, RobertT wrote: > I've a problem building FS rev 15630 on Windows. One of mod_pocketsphinx related projects lack a code file. > From mike at jerris.com Tue Nov 24 00:33:34 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 03:33:34 -0500 Subject: [Freeswitch-users] os x compile failure In-Reply-To: <8CD390C6-D194-483C-8A0A-732B5BFFCE09@freeswitch.org> References: <40B8DB3D-2F66-45BC-BB9E-9B773B707FA3@talent.com.au> <8CD390C6-D194-483C-8A0A-732B5BFFCE09@freeswitch.org> Message-ID: <363278A1-586F-4493-8A7E-BEEDAB036000@jerris.com> Please retest this with current svn trunk fresh checkout. Thanks Mike On Nov 23, 2009, at 9:47 PM, Brian West wrote: > Ok 32bit... we are currently working on that as I type. > > /b > > On Nov 23, 2009, at 8:44 PM, James Budge wrote: > >> 2GHz Intel Core Duo >> >> OS 10.6.2 >> >> Xcode 3.2.1 >> >> Updated to revision 15648. > From mike at jerris.com Tue Nov 24 00:39:16 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 03:39:16 -0500 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH In-Reply-To: References: , <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org> Message-ID: <6F2A2A62-CE26-477F-B402-358F313A3EC3@jerris.com> This was fixed in trunk yesterday about 8 hrs before you sent this message. (15619). Please update and try again. Mike On Nov 23, 2009, at 11:33 PM, John Platts wrote: > > I was using revision 15586. > > ---------------------------------------- >> From: brian at freeswitch.org >> Date: Mon, 23 Nov 2009 18:25:44 -0600 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH >> >> What rev exactly? >> >> /b >> >> On Nov 23, 2009, at 6:19 PM, John Platts wrote: >> >>> >>> I actually checked out the latest version of FreeSWITCH in the SVN >>> repository. >>> >>> I have the following configured in /usr/local/freeswitch/conf/ >>> dialplan/default.xml: >>> >>> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/3ece4be6/attachment-0002.html From mike at jerris.com Tue Nov 24 00:41:25 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 03:41:25 -0500 Subject: [Freeswitch-users] DTMF javasript In-Reply-To: References: Message-ID: Your not telling anything to call your callback. On Nov 24, 2009, at 1:03 AM, Baskar wrote: > Hi, > > I want to check value given to the javascript with conditions whether it is voicefile, extension or mobile Number when i press the dtmf value. > > Steps i need to check in javascript: > > When i Press the DTMF value 1 it should check the 3 condition > > If the Value for argv[2]=vfsurya means it is a voice file so it should play the Voice file > If the Value for argv[2]=1001 means it is a extension. The call should Bridge the extension > If the Value for argv[2]=9841799874 means it is a Mobile number. The call should Bridge the Mobile number > > var exit = false; > var dtmf_digits = ""; > var repeat = 0; > var argv[2]=vfsurya; // or var argv[2]=1001 or var argv[2]=Mobile Number > > > function onInput( session, type, data, arg ) > { > if ( type == "dtmf" ) > { > console_log( "info", "Got digit " + data.digit + "\n" ); > if ( data.digit == "1" ) > { > if(argv[2].startswith("vf")) > { > var voice2=voice.substring(2)+"
" > session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/"+voice2+".wav", onInput ); > } > else if(argv[2].length==4) > { > console_log( "info", "Got voicefile " + argv[2] + "\n" ); > session.execute("bridge", "sofia/internal/"+argv[2]+"%192.168.1.2", onInput ); > } > else > { > session.execute("bridge", "sofia/default/sip:"+argv[2]+"@192.168.1.135:5066", onInput ); > } > } > } > } > > But if 1 is pressed there is no event trigger but it get the dtmf value as 1 in freeswitch console. > > can any one specify what is the error or correct me where i am wrong. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/70af6ed1/attachment-0002.html From mike at jerris.com Tue Nov 24 00:42:07 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 03:42:07 -0500 Subject: [Freeswitch-users] DTMF Event is not coming while using playback terminators. In-Reply-To: <1452e2980911232322j106fe5eoe3efad59199f36e4@mail.gmail.com> References: <1452e2980911232322j106fe5eoe3efad59199f36e4@mail.gmail.com> Message-ID: <7EE00C1B-83CD-44C2-9CC0-F66120A9B534@jerris.com> async? On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: > Dear All, > I am using Perl ESL::IVR module to develop a simple IVR. I have filtered DTMF events. I have also set playback_terminators to cut the playback when giving the digits. I have faced problem that DTMF event has not come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. > > Why the 'COMMAND' event came event filter is on? > How to avoid this event in ESL? From velu.technical at gmail.com Tue Nov 24 00:59:29 2009 From: velu.technical at gmail.com (velusamy velu) Date: Tue, 24 Nov 2009 14:29:29 +0530 Subject: [Freeswitch-users] DTMF Event is not coming while using playback terminators. In-Reply-To: <7EE00C1B-83CD-44C2-9CC0-F66120A9B534@jerris.com> References: <1452e2980911232322j106fe5eoe3efad59199f36e4@mail.gmail.com> <7EE00C1B-83CD-44C2-9CC0-F66120A9B534@jerris.com> Message-ID: <1452e2980911240059n4223bd48u487a80b8306131da@mail.gmail.com> Yes, I am using async mode only.. On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris wrote: > async? > > On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: > > > Dear All, > > I am using Perl ESL::IVR module to develop a simple IVR. I have > filtered DTMF events. I have also set playback_terminators to cut the > playback when giving the digits. I have faced problem that DTMF event has > not come if DTMF given while playing voice files. I have received 'COMMAND' > event. I have the following questions. > > > > Why the 'COMMAND' event came event filter is on? > > How to avoid this event in ESL? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/2d65e5d2/attachment-0002.html From info at daccii.it Tue Nov 24 01:15:45 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Tue, 24 Nov 2009 10:15:45 +0100 Subject: [Freeswitch-users] User who answer the bridge in a execute_answer In-Reply-To: <9133578A-F706-46C2-9653-6C22D6E056CB@jerris.com> References: <4B0A65C9.10509@daccii.it> <9133578A-F706-46C2-9653-6C22D6E056CB@jerris.com> Message-ID: <4B0BA441.9060905@daccii.it> Hi, thanks for the suggestion! In the end i updated freeswitch using lastest source in the trunk and callee_id_number worked! Best Regard, Daniele Michael Jerris ha scritto: > Try running the info app there and see if the info is anywhere in that output . > > Mike > > On Nov 23, 2009, at 5:36 AM, Albano Daniele Salvatore - Lavoro wrote: > >> Hi, >> >> i'm writing some dialplan parts that get executed on execute_on_answer. In this dialplan that get executed i need to make a directory to handle recordings for record_session and my folder structure is: >> USER/YEAR/MONTH/HOUR-MINUTE-SECOND-CALLER_NUMBER.wav >> >> ------ >> >> >> ------ >> >> The call flow is: >> Call from external -> IVR -> Transfer to Group -> Execute on Answer -> system/bind_meta_app >> >> >> Pratically, i need the number (or better the user) that answered the call: what variable should i check? >> >> I tried with sip_from_user, callee_id_number and some other. >> >> >> Thank for your help, >> >> Best Regards, >> Daniele> -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/8a3593f2/attachment-0002.vcf From stevendt at primrosebank.net Tue Nov 24 02:29:08 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 24 Nov 2009 10:29:08 -0000 Subject: [Freeswitch-users] Call Transfer Help Please Message-ID: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> Hi, I'm trying to setup call transfer for a phone without a transfer button. I was on IRC last night and got some pointers to how this is setup in dialplan.xml and features.xml and what "bind meta app" does. Once it became clear how the transfer is initiated and that the transfer, in the default config, can only be initiated by the "b" leg of the call, I was able to make this work as configured in the defaults, i.e, to initiate a transfer (for an internal call) from the dialled extension to a new extension. Now the problem . . . I have an incoming PSTN line that rings a group of extensions, what I want to be able to do is to give whoever answers the PSTN call ability to transfer the call on to another extension. There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch extension of 1000, it rings the extension phones in the group. I'd hoped that the default transfer setup would handle this without modification - the incoming call on extension 1000 would be the "a" leg, the answering extension would be the "b" leg and a transfer from "b" would work as per the default config. This does not work for me though. I'm struggling a bit with the "bind meta app" options and can't seem to make it do what I want. Could someone please confirm that what I'm trying to do is feasible and perhaps suggest the right parameters to use in dialplan.xml and features.xml please ? Relevant section in the "is_transfer" section in features.xml And in default.xml from to I've tried posting a call log to the Pastebin (11252/3) but there was an error - it looks like the dump was too big. Not sure what the maximum size on pastebin dumps is ? My understanding (or lack of) of "a" and "b" are in the scenario described is not helping ... Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? Is the answering extension the "b" leg ? What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ? What is the correct "transfer" data string in features.xml ? Or am I totally on the wrong track here ? If it is possible to do what I want, and changes are required to the dialplan.xml and/or features.xml files, is it possible to have different logic in there such that the actions are different whether it is the "a" leg or "b" leg that's requesting the transfer ? regards Dave FreeSwitch Version 1.0.4 (14460) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b78cef61/attachment-0002.html From lakindia89 at gmail.com Tue Nov 24 02:57:05 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Tue, 24 Nov 2009 16:27:05 +0530 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> Message-ID: <7d79b3930911240257g4c22a09dhad954629ae49072d@mail.gmail.com> Yes Mr. Collins, I've tried with shed_api. But I was not able to control, if the user reject the call. I made a shed_api to originate a call to 1000 and If it is answered, I'll transfer the call to 9097 (So it comes to my program, refer the dialplan in my question). But what happens if the user 1000, reject the call. I can't control that. If the user 1000, reject the call, I need to call the user after some time. Any way to do this!! On Mon, Nov 23, 2009 at 11:21 PM, Michael Collins wrote: > > > On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> Hi, >> I'm using perl ESL to control the call in freeswitch. >> I'm having the following scenario, but not able to get it right. >> >> Dialplan: >> >> >> >> >> >> >> >> >> 1. User A calls to an extention (1000). >> 2. My ESL program will be running, and it answers the call. >> 3. Then the program will get a number from the user. >> 4. It will hangup the call. >> 5. The program has to call to the number that was given by the user. >> >> In the above scenario, I was able to do until the 4th step. After hangup >> the call, if I say originate it is not working. >> Any ideas on how to do this in ESL. >> >> > I want to make sure I understand what the script is supposed to be doing. > The caller will key in a phone number to your script and your script will > collect those digits. The script will then hangup on the caller and > originate a completely new call? Perhaps you could use sched_api to schedule > a new originate command for a few seconds into the future and then hangup? > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/d009686b/attachment-0002.html From oscav at hotmail.fr Tue Nov 24 04:23:52 2009 From: oscav at hotmail.fr (Oscav) Date: Tue, 24 Nov 2009 04:23:52 -0800 (PST) Subject: [Freeswitch-users] Execute on Answer with JavaScript In-Reply-To: References: <26476532.post@talk.nabble.com> Message-ID: <26494996.post@talk.nabble.com> Hi Mike, I understand. I just need to not use the session.answer(). Many thanks. Michael Jerris wrote: > > This is done automatically when you bridge 2 sessions together. > > Mike > > On Nov 23, 2009, at 6:45 AM, Oscav wrote: >> How can we send the answer to the caller only when the callee answers, in >> JavaScript?? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Execute-on-Answer-with-JavaScript-tp26476532p26494996.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From lakindia89 at gmail.com Tue Nov 24 04:27:40 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Tue, 24 Nov 2009 17:57:40 +0530 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> Message-ID: <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> I've tried the following program as per the suggestion that you've told. But it seems, no success. Once the connection is closed, I created a new connection and I send originate to originate a new call. But it is not working. require ESL; use IO::Socket::INET; use Data::Dumper; my $ip = "192.168.1.222"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; my $make_call; my $con; my $type = "user/"; for(;;) { my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { close($new_sock); next; } my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s to %s\n", $uuid, $info->getHeader("caller-caller-id-number"), $info->getHeader("caller-destination-number"); $con->filter("Unique-Id", $uuid); $con->events("plain", "all"); $con->execute("answer"); $con->setEventLock("true"); my $number=$con->execute("read","2 4 /usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav accnt_number 5000 #"); while($con->connected()) { my $e=$con->recvEvent(); my $ename=$e->getHeader("Event-Name"); my $app=$e->getHeader("Application"); if($ename eq "CHANNEL_EXECUTE_COMPLETE" and $app eq "read") { my $num=$e->getHeader("variable_accnt_number"); print "$num\n"; $con->execute("hangup"); } } if(!$con->connected()) { print "Connection not exists\n"; $con = new ESL::ESLconnection($fd); $con->api("originate","user/1000 &park()"); print "Hai\n"; } print "Bye\n------------------------------------------------------------------\n"; close($new_sock); } Output: Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097 1000 Connection not exists Hai Bye ------------------------------------------------------------------ The freeswitch log is in http://pastebin.freeswitch.org/11258 I also noted that, if I don't receive any events, especially "SERVER_DISCONNECTED", then the connection is in established state, but once I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it correct?? On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > or open a new outbound connection at the end of your script so you can send > your originate command. > Since the channel hanging up will close your existing connection since it's > only an outbound single session socket. > > > On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins wrote: > >> >> >> On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < >> lakindia89 at gmail.com> wrote: >> >>> Hi, >>> I'm using perl ESL to control the call in freeswitch. >>> I'm having the following scenario, but not able to get it right. >>> >>> Dialplan: >>> >>> >>> >>> >>> >>> >>> >>> >>> 1. User A calls to an extention (1000). >>> 2. My ESL program will be running, and it answers the call. >>> 3. Then the program will get a number from the user. >>> 4. It will hangup the call. >>> 5. The program has to call to the number that was given by the user. >>> >>> In the above scenario, I was able to do until the 4th step. After hangup >>> the call, if I say originate it is not working. >>> Any ideas on how to do this in ESL. >>> >>> >> I want to make sure I understand what the script is supposed to be doing. >> The caller will key in a phone number to your script and your script will >> collect those digits. The script will then hangup on the caller and >> originate a completely new call? Perhaps you could use sched_api to schedule >> a new originate command for a few seconds into the future and then hangup? >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b7ad5841/attachment-0002.html From oscav at hotmail.fr Tue Nov 24 04:30:05 2009 From: oscav at hotmail.fr (Oscav) Date: Tue, 24 Nov 2009 04:30:05 -0800 (PST) Subject: [Freeswitch-users] sched_broadcast doesn't execute In-Reply-To: <191c3a030911181128g35ba0652w9fc575d5586367dc@mail.gmail.com> References: <26408422.post@talk.nabble.com> <191c3a030911181128g35ba0652w9fc575d5586367dc@mail.gmail.com> Message-ID: <26495078.post@talk.nabble.com> Hi Anthony, Now it works very well. Thank you so much for your help. I'm having a lot of fun with this platform. Regards. Anthony Minessale-2 wrote: > > is that your exact code? > > ${uuid} will not be expanded by javascript > > var uuid = session.getVariable(uuid); > > new_session.execute("sched_broadcast", "+20 alloted_timeout " + uuid + " > playback:ivr-welcome_to_freeswitch.wav"); > > On Wed, Nov 18, 2009 at 10:07 AM, Oscav wrote: > >> >> Hi, >> >> I'm writing a script in Javascript that plays a message during a bridge. >> I'm >> trying to use a sched_broadcast to do it. The job is scheduled and then >> deleted but I never hear the wav file and I don't get the "OK Message >> Scheduled" in the log. It even doesn't display any error message if I >> specify a wrong file name. Someone could help me on this issue ?? >> >> new_session.execute("sched_broadcast", "+20 alloted_timeout ${uuid} >> playback:ivr-welcome_to_freeswitch.wav"); >> >> I already did some posts but I got no answer. This is very difficult to >> progress without help. >> >> Many thanks >> -- >> View this message in context: >> http://old.nabble.com/sched_broadcast-doesn%27t-execute-tp26408422p26408422.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/sched_broadcast-doesn%27t-execute-tp26408422p26495078.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yehavi.bourvine at gmail.com Tue Nov 24 05:05:34 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 24 Nov 2009 15:05:34 +0200 Subject: [Freeswitch-users] How to find whether the destination extension supports encryption Message-ID: Hello, We have a mix of phones that support RTP encryption and those that do not. I have to support both types in the meanwhile, and would like to have encryption enabled on the relevant leg, even if the other leg does not support it (why? one of our ATAs either must have it unencrypted or have it encrypted, but cannot have both). How do I find whether the *destination* supports encryption? I do not want to manage an additional table in the database... Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/7e41e5a6/attachment-0002.html From yehavi.bourvine at gmail.com Tue Nov 24 05:02:15 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 24 Nov 2009 15:02:15 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: <191c3a030911231157p44612c5dm3f0ee1e7b37e9cd3@mail.gmail.com> References: <4B0266F4.8070602@savion.huji.ac.il> <191c3a030911231157p44612c5dm3f0ee1e7b37e9cd3@mail.gmail.com> Message-ID: Hello Anthony, Indeed I see the reference to this channel variable in the code, but when trying to access it from the dial plan it is empty... I try to get the value of ${sip_profile_name} and it is empty. Thanks! __Yehavi: 2009/11/23 Anthony Minessale > Let's just do this: > > r15629 or higher > > look for sip_profile_name > > > > > On Tue, Nov 17, 2009 at 3:03 AM, Eli Hayun wrote: > >> Hi >> We have more then one profile. To make a call I have to enter : bridge >> sofia/profile/number at ip >> The problem is when I use : "${use_profile}" I am getting the caller >> profile, and I need the destination profile. >> >> How do I get this information? >> >> Thanks >> >> Eli >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b43fc10c/attachment-0002.html From achaloyan at yahoo.com Tue Nov 24 05:19:09 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Tue, 24 Nov 2009 05:19:09 -0800 (PST) Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <1258995685201-4052409.post@n2.nabble.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <552708.67071.qm@web111314.mail.gq1.yahoo.com> <1258949788572-4048969.post@n2.nabble.com> <858430.90192.qm@web111301.mail.gq1.yahoo.com> <1258995685201-4052409.post@n2.nabble.com> Message-ID: <900524.79591.qm@web111310.mail.gq1.yahoo.com> Hi Jeff, All is good, I have looked at the x64 related changes you made and will merge them back to UniMRCP tree most probably during the next week. Thanks, Arsen. ________________________________ From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Mon, November 23, 2009 9:01:25 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP Hi Arsen, I have merged your changes in now - thank you. Would you perhaps be able to look at the x64 changes I made to the projects and merge them back into your code to ease the future updating. Thanks Jeff Arsen Chaloyan wrote: > > Hi Jeff, > > > Your input would be very helpful, I just wanted to understand where the > problem is and contribute the way I can. > I see you're the assignee, so please go ahead and let me know if there is > anything left I can help with. > > Arsen. > > > > ________________________________ > From: Jeff Lenk > To: freeswitch-users at lists.freeswitch.org > Sent: Mon, November 23, 2009 8:16:28 AM > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > Hi Arsen, > > I would be happy to help with the FS integration if you want - please do > put your patch in a Jira. > > Jeff > > ________________________________ > Date: Sun, 22 Nov 2009 10:09:41 -0800 > From: [hidden email] > To: [hidden email] > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > > We discussed build integration related issues a few months ago with Mike > and seemed to find a solution which would work for both UniMRCP and > FreeSWITCH source trees. > > Now I've just got a chance to look into this a bit closer trying to > further complete VS2008 build integration in FreeSWITCH. So I've got it > working, the module is not only being built, but also is getting loaded. > Current build integration is not as seamless as I want it to be, but > probably we can start with what we have now and then discuss and identify > what can be done in the future. This concerns not only build integration > but overall integrity. > > So would you be interested in the patch? Where should I upload it? > I thought I had a Jira account, but not sure it exists any more. > > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org > > > > > > ________________________________ > From: Jeff Lenk <[hidden email]> > To: [hidden email] > Sent: Fri, November 20, 2009 7:59:28 PM > Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch & > uniMRCP > > > That module is not currently being built for Windows. Also the library > unimrcp needs build integration work with FS to make that happen under > windows. > > > ss1 wrote: > >> >> Hi Everyone, >> >> Please help freeswitch experts... !!! >> >> i have been working on freeswitch from last 2 days. i have downloaded >> freeswitch and unimrcp (server + client) for windows. >> I tested the unimrcp client and server, which is running fine with the >> command: run synth and run recog. I got both synth.pcm & recog.pcm files. >> >> But my objective is to call Freeswitch through x-lite, where freeswitch >> should call unimrcp client and return the PCM files. >> >> I tried it alot, but unable to do it. after lots of reading i found that >> i >> do not have mod_unimrcp. i do not know from where to download it and how >> to merge it into freeswitch. >> >> I would be very thankful if you may help. >> >> Thanks, >> ss >> >> -- > View this message in context: > http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > > View message @ > http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html > To unsubscribe from Re: need help !! Problem with freeswitch & uniMRCP, > click here. > > ________________________________ > Hotmail: Trusted email with powerful SPAM protection. Sign up now. > ________________________________ > View this message in context: RE: [Freeswitch-users] need help !! Problem > with freeswitch & uniMRCP > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4052409.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/2072e68b/attachment-0002.html From steve at justfone.com Tue Nov 24 05:25:06 2009 From: steve at justfone.com (Steven Brown) Date: Tue, 24 Nov 2009 13:25:06 +0000 Subject: [Freeswitch-users] Noise with openzap Message-ID: <3e6d7b0c0911240525o747f7b05y5c1f50ec6afe1179@mail.gmail.com> Hi, I have an Ubuntu box running FS1.0.4 which has been processing a good volume of calls between local users with soft phones (Xlite) and GSM handsets via a number or Portech gateways, this has worked very well for some time and audio quality is very good. I've now added a Sangoma A200 with 4 ports hooked up to 4 PSTN lines, configured openzap and I can originate and answer calls on the the openzap lines fine, however these calls via opezap all seem to suffer from significant noise, the audio path works fine in both directions but noise seems particularly bad at the local soft phone end. Quality of all other calls through the box is fine though, any ideas appreciated ?, NB A regular handset plugged directly into the PSTN lines has no problems though Thanks Steve From lei.tlfly at gmail.com Tue Nov 24 06:12:31 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 24 Nov 2009 22:12:31 +0800 Subject: [Freeswitch-users] FS cluster and how to get sofia gateway health status? Message-ID: <50c41b4e0911240612w7506a2f1qed0f25143c0b65d2@mail.gmail.com> Hi everyone, I'm setting up FS cluster In my application, I plan to use two FS server as front and four FS as backend, the incoming calls first send to the front FS, then the front FS forward the call to backend FS server by return 302 to invite message. The front FS need to known the backend FS's status, so it won't forward calls to a server if it's down. The question is, how to check the backend FS's status. As I known, fs can add gateways to sofia profile, the endpoint will check gateways's state by send ping message, I think it is the function what I need if I can get the gateways's status from fs, does someone known how to do it? or can someone give me some suggestion about how to setup FS cluster? -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/57da1325/attachment-0002.html From imthiyazg at gmail.com Tue Nov 24 06:54:23 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Tue, 24 Nov 2009 20:24:23 +0530 Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <1258732768082-4038514.post@n2.nabble.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> Message-ID: <8595daf70911240654y72f0440cm2ab1a50babc96f0f@mail.gmail.com> Hi Can we enable passive recording in freeswitch ,wanpipe ,openzap , we are using a sangoma tapping system with freeswitch. Thanks Imthiyaz From ovvenkatesan at gmail.com Tue Nov 24 04:49:34 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Tue, 24 Nov 2009 18:19:34 +0530 Subject: [Freeswitch-users] How to run IVR application Message-ID: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> Hi to all, I am very new this platform . I just downloaded freeswitch to my windows xp machine , compiled successfully and run. After that I dont have any idea what to do :( . I am not finding simple kind of tutorial on the net. could you please suggest me, how I have to proceed. My requirement is; I need to run IVR application on machine using SIP phone. I am very sorry to my bad English. Thanks and Regards, Venkat. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/595e26d5/attachment-0002.html From rajesh.npnr at yahoo.com Tue Nov 24 07:01:04 2009 From: rajesh.npnr at yahoo.com (rex.alex) Date: Tue, 24 Nov 2009 07:01:04 -0800 (PST) Subject: [Freeswitch-users] SoftPhone Message-ID: <1259074864897-4058292.post@n2.nabble.com> Hello, I have been going through FreeSWITCH for quite sometime now. I would like to develop my own SIP Client soft-phone in Java/etc., how do I start?. Will I get any SDK/APIs for this. Please assist. Thanks, Rex -- View this message in context: http://n2.nabble.com/SoftPhone-tp4058292p4058292.html Sent from the freeswitch-users mailing list archive at Nabble.com. From juanbackson at gmail.com Tue Nov 24 07:22:22 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 24 Nov 2009 23:22:22 +0800 Subject: [Freeswitch-users] remote_media_ip variable not set Message-ID: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> Hi, I tried to use the variable remote_media_ip from within dialplan, but it is not returning anything. Does anyone know when this variable gets set and how to have this variable to be set as soon as an INVITE hit freeswitch? Thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/66c5d26e/attachment-0002.html From edpimentl at gmail.com Tue Nov 24 07:30:48 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 24 Nov 2009 10:30:48 -0500 Subject: [Freeswitch-users] SoftPhone In-Reply-To: <1259074864897-4058292.post@n2.nabble.com> References: <1259074864897-4058292.post@n2.nabble.com> Message-ID: <9dc4a1670911240730q58f52e65o3b7586b30fd684e9@mail.gmail.com> Suggestion: Be one the first to integrate QuteCOm -E Gpro.ws -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/baefd770/attachment-0002.html From mike at jerris.com Tue Nov 24 07:43:44 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 10:43:44 -0500 Subject: [Freeswitch-users] need help !! Problem with freeswitch & uniMRCP In-Reply-To: <8595daf70911240654y72f0440cm2ab1a50babc96f0f@mail.gmail.com> References: <1258634740580-4031590.post@n2.nabble.com> <1258732768082-4038514.post@n2.nabble.com> <8595daf70911240654y72f0440cm2ab1a50babc96f0f@mail.gmail.com> Message-ID: <0636459B-CEAF-4332-84BD-D32DA0322A29@jerris.com> What does this have to do with uniMRCP? Mike On Nov 24, 2009, at 9:54 AM, Imthiyaz Ahmed wrote: > Hi > > Can we enable passive recording in freeswitch ,wanpipe ,openzap , we > are using a sangoma tapping system with freeswitch. From mrene_lists at avgs.ca Tue Nov 24 07:46:43 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 24 Nov 2009 10:46:43 -0500 Subject: [Freeswitch-users] remote_media_ip variable not set In-Reply-To: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> References: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> Message-ID: <2F929FDB-0E1B-49E0-A1E7-F4F1E2D548AD@avgs.ca> It gets set whenever the codec is negotiated. So it'll be NULL until (pre_)answer if you have late-negotiation on. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 24-Nov-09, at 10:22 AM, Juan Backson wrote: > Hi, > > I tried to use the variable remote_media_ip from within dialplan, > but it is not returning anything. > > Does anyone know when this variable gets set and how to have this > variable to be set as soon as an INVITE hit freeswitch? > > Thanks, > jb > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From juanbackson at gmail.com Tue Nov 24 07:56:17 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 24 Nov 2009 23:56:17 +0800 Subject: [Freeswitch-users] remote_media_ip variable not set In-Reply-To: <2F929FDB-0E1B-49E0-A1E7-F4F1E2D548AD@avgs.ca> References: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> <2F929FDB-0E1B-49E0-A1E7-F4F1E2D548AD@avgs.ca> Message-ID: <27c25bc40911240756k7842c80kd75be2d3d93441b9@mail.gmail.com> Hi, In the case of proxy_media=true, does it gets set at all then? thanks, jb On Tue, Nov 24, 2009 at 11:46 PM, Mathieu Rene wrote: > It gets set whenever the codec is negotiated. So it'll be NULL until > (pre_)answer if you have late-negotiation on. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 24-Nov-09, at 10:22 AM, Juan Backson wrote: > > > Hi, > > > > I tried to use the variable remote_media_ip from within dialplan, > > but it is not returning anything. > > > > Does anyone know when this variable gets set and how to have this > > variable to be set as soon as an INVITE hit freeswitch? > > > > Thanks, > > jb > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/30a1aeb9/attachment-0002.html From anthony.minessale at gmail.com Tue Nov 24 11:24:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Nov 2009 13:24:58 -0600 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> Message-ID: <191c3a030911241124r49679ad7je630a964dd60a5c3@mail.gmail.com> 1) Did you ever supply a log of your problem? 2) Are you using ESL lib or did you make your own event socket client, (if you did maybe you implemented the protocol client wrong) You are not supplying any specific information like traces of the connection or the version of the code you are using, weather you have tried the latest release or not etc. And lastly you are not using the events I told you about to tell exactly when the commands in question are being executed. getting a variable in a loop is a non-scalable memory consuming bad idea in how to program over a socket. On Mon, Nov 23, 2009 at 11:56 PM, Thangappan.M wrote: > The reason for waiting only for DTMF event is to handle the time outs in > the IVR concept like response and inter digit time out. Using our own logic > we 10 voice files in each play back if the voice files are more than 10. Now > it works fine. > > Now the new problem has been raised. The problem is we are filtering only > for DTMF events but we are getting COMMAND event . Because of this the DTMF > digits are missing at the time . I am not able to proceed further. We are > in the critical situation. > > Why this command event is occurring? > How can I restrict this? > What are the information it has? > How can I get all the information in it ? ( If command event has info) > > Help me............ > > > On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: > >> I am waiting only for DTMF events. That's why I am setting freeswitch >> variable for knowing whether the playback has done. >> >> My question is "why this freeswitch variable is not setting properly when >> I play back more than 10 files using playback_delimiter option?". >> >> When I play back lesser than ten voice files the variable has been set >> properly. What could be the reason? >> >> >> >> ---------- Forwarded message ---------- >> From: Thangappan.M >> Date: Sat, Nov 21, 2009 at 2:52 PM >> Subject: Problem while playing more than 10 voice files using playback >> To: freeswitch-users >> >> >> Dear all, >> >> I am in the process of implementing IVR using event outbound >> socket (async mode). >> I have implemented using Perl language. >> >> I did the following steps: >> => Set the playback_delimiter variable >> => Set the playback_sleep_val variable >> => Set the event lock as true >> => Set the freeswitch ( my own) variable as zero >> => Wait in the loop until the variable is been set as >> zero >> => Playback the voice files ( Here I combined the >> voice files with the delimiter value if more than one voice files are there) >> => Set the freeswitch(my own) variable as true ( This >> is used to identify whether the voice files are played >> successfully). >> => Wait in the loop until the variable is been set as >> one. >> => Set the Event lock as false >> >> => Trying to get the DTMF digits ( Have a assurance >> that all the voice files are played). >> >> The problem is, >> >> The above steps are working fine when the voice file count is >> lesser than or equal to 10. After the voice files are played only the >> variable(my own freeswitch) is set. Based on the variable I am doing further >> things. >> >> But when I tried to give the voice files count of more than >> 10 the variable has been set while starting to play back the first voice >> file itself . Because of this I am not able to proceed further. >> >> *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* >> >> *NOTE*: I also referred mod_file_string documentation. In that they >> specified 128 files can be used to play back the voice files using >> playback_delimiter option. >> >> Please help me................? >> Thanks in advance. >> >> >> -- >> Regards, >> Thangappan.M >> >> >> >> -- >> Regards, >> Thangappan.M >> > > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/5c44a390/attachment-0002.html From john_platts at hotmail.com Tue Nov 24 11:41:15 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 24 Nov 2009 13:41:15 -0600 Subject: [Freeswitch-users] Patch to allow gateways to be defined without the password parameter set Message-ID: I have modified sofia.c in mod_sofia so that I can define gateways without having to specify the password parameter. This is because I am using a SIP gateway that does not require SIP registration. The modified version still requires the password to be set on any gateway for which register is set to true. Attached is the diff file for these changes. _________________________________________________________________ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search?q=restaurants&form=MFESRP&publ=WLHMTAG&crea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 -------------- next part -------------- A non-text attachment was scrubbed... Name: sofia_password_patch.diff Type: application/octet-stream Size: 815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/1757affb/attachment-0002.obj From brian at freeswitch.org Tue Nov 24 11:58:12 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Nov 2009 13:58:12 -0600 Subject: [Freeswitch-users] Patch to allow gateways to be defined without the password parameter set In-Reply-To: References: Message-ID: John, If the remote end doesn't require a username or password then you don't need to create a gateway to send a call to that endpoint. You can simply do sofia/profile/number at remoteip and it'll work. Also can you put the patch on jira via http://jira.freeswitch.org /b On Nov 24, 2009, at 1:41 PM, John Platts wrote: > > I have modified sofia.c in mod_sofia so that I can define gateways > without having to specify the password parameter. This is because I > am using a SIP gateway that does not require SIP registration. The > modified version still requires the password to be set on any > gateway for which register is set to true. Attached is the diff file > for these changes. > > _________________________________________________________________ > Bing brings you maps, menus, and reviews organized in one place. > http://www.bing.com/search? > q > = > restaurants > &form > = > MFESRP > &publ > = > WLHMTAG > &crea > = > TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 > < > sofia_password_patch.diff > >_______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Nov 24 12:15:37 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 15:15:37 -0500 Subject: [Freeswitch-users] DTMF Event is not coming while using playback terminators. In-Reply-To: <1452e2980911240059n4223bd48u487a80b8306131da@mail.gmail.com> References: <1452e2980911232322j106fe5eoe3efad59199f36e4@mail.gmail.com> <7EE00C1B-83CD-44C2-9CC0-F66120A9B534@jerris.com> <1452e2980911240059n4223bd48u487a80b8306131da@mail.gmail.com> Message-ID: <6DBE01B2-75A5-4A0F-9BF9-C7434114DD92@jerris.com> 1. can you supply a trace of this esl communications. 2. is it inband or rfc2833 dtmf ? MIke On Nov 24, 2009, at 3:59 AM, velusamy velu wrote: > Yes, I am using async mode only.. > > On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris wrote: > async? > > On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: > > > Dear All, > > I am using Perl ESL::IVR module to develop a simple IVR. I have filtered DTMF events. I have also set playback_terminators to cut the playback when giving the digits. I have faced problem that DTMF event has not come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. > > > > Why the 'COMMAND' event came event filter is on? > > How to avoid this event in ESL? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/00974a44/attachment-0002.html From mike at jerris.com Tue Nov 24 12:19:17 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Nov 2009 15:19:17 -0500 Subject: [Freeswitch-users] Call Transfer Help Please In-Reply-To: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> References: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> Message-ID: On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: > Hi, > > I'm trying to setup call transfer for a phone without a transfer button. I was on IRC last night and got some pointers to how this is setup in dialplan.xml and features.xml and what "bind meta app" does. > > Once it became clear how the transfer is initiated and that the transfer, in the default config, can only be initiated by the "b" leg of the call, I was able to make this work as configured in the defaults, i.e, to initiate a transfer (for an internal call) from the dialled extension to a new extension. > > Now the problem . . . > > I have an incoming PSTN line that rings a group of extensions, what I want to be able to do is to give whoever answers the PSTN call ability to transfer the call on to another extension. > > There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch extension of 1000, it rings the extension phones in the group. > > I'd hoped that the default transfer setup would handle this without modification - the incoming call on extension 1000 would be the "a" leg, the answering extension would be the "b" leg and a transfer from "b" would work as per the default config. This does not work for me though. > > I'm struggling a bit with the "bind meta app" options and can't seem to make it do what I want. > > Could someone please confirm that what I'm trying to do is feasible and perhaps suggest the right parameters to use in dialplan.xml and features.xml please ? > > Relevant section in the "is_transfer" section in features.xml > > > And in default.xml from > to > > I've tried posting a call log to the Pastebin (11252/3) but there was an error - it looks like the dump was too big. Not sure what the maximum size on pastebin dumps is ? > > > My understanding (or lack of) of "a" and "b" are in the scenario described is not helping ... > > Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? Yes, the calling leg > Is the answering extension the "b" leg ? Yes > What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ? I don't understand this question > What is the correct "transfer" data string in features.xml ? > ditto > Or am I totally on the wrong track here ? > You should just need to make sure that the bind meta is called in this scenario so the b leg is able to do it, thats it. > If it is possible to do what I want, and changes are required to the dialplan.xml and/or features.xml files, is it possible to have different logic in there such that the actions are different whether it is the "a" leg or "b" leg that's requesting the transfer ? > > regards > Dave > > FreeSwitch Version 1.0.4 (14460) also, try the latest 1.0.5. pre release or svn trunk to confirm this is not an issue that has already been fixed. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/b1fe436d/attachment-0002.html From Prometheus001 at gmx.net Tue Nov 24 12:54:50 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 24 Nov 2009 21:54:50 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> Message-ID: <4B0C481A.8030309@gmx.net> Hello, I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console. I have explicitely set in the internal profile. When a phone is set up I get the following Snom Phone REGISTER => OpenSIPS=> Freeswitch Freeswitch OK => OpenSIPS=>Snom Phone Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch Freeswitch 202 Accepted => OpenSIPS=>Snom Phone Snom Phone PUBLISH => OpenSIPS=> Freeswitch Freeswitch 200 OK => OpenSIPS=>Snom Phone So presence generally seems to work. But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there. Any idea how to force the NOTIFY messages? Best regards Peter Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it: freeswitch at sip11.mydomain.com> nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 401 Unauthorized for REGISTER (6) nua(0x905a80): recv signal r_destroy nta_leg_destroy((nil)) nta: received PUBLISH sip:100 at sip1.mydomain.com SIP/2.0 (CSeq 3) nta: PUBLISH (3) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905f10): event i_publish 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905f10): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x905f10): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua(0x905f10): sent signal r_destroy nta: sent 200 OK for PUBLISH (3) nua(0x905f10): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:mod_sofia at 192.168.178.200:5062 SIP/2.0 (CSeq 2) nta: canonizing sip:mod_sofia at 192.168.178.200:5062 with contact nta: SUBSCRIBE (2) going to existing leg nua: nua_stack_process_request: entering nta: sent 200 OK for SUBSCRIBE (2) nua(0x905560): event i_subscribe 200 OK nua: nua_application_event: entering nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 8) nta: REGISTER (8) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5dc073ba0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5dc073ba0): sent signal r_respond nua(0x7fd5dc073ba0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua: nua_handle_destroy: entering nua(0x7fd5dc073ba0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (8) nua(0x7fd5dc073ba0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x8fc3d0): sent signal r_respond nua(0x8fc3d0): recv signal r_respond 200 OK nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (7) nua(0x8fc3d0): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:100 at sip1.mydomain.com;user=phone SIP/2.0 (CSeq 1) nta: SUBSCRIBE (1) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nta_leg_tcreate(0x7fd5dc03add0) nua(0x7fd5dc078b70): adding notify usage with event message-summary nua(0x7fd5dc078b70): event i_subscribe 100 Trying nua: nua_application_event: entering nua(): refresh notify after 3600 seconds (in [3600..3600]) nua: nua_respond: entering nua(0x7fd5dc078b70): sent signal r_respond nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted nua: nua_stack_set_params: entering nta: sent 202 Accepted for SUBSCRIBE (1) mayamatakeshi schrieb: > > On 9/12/09, *mayamatakeshi* > wrote: > > > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris > wrote: > > Following up, did a bug get created for this issue? > > > Hello, > yes. > http://jira.freeswitch.org/browse/MODSOFIA-26 > > > Just to simplify things in case someone searches the list: > Issue was solved on rev 14851. > Thank you all. > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From john_platts at hotmail.com Tue Nov 24 13:05:48 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 24 Nov 2009 15:05:48 -0600 Subject: [Freeswitch-users] Call forwarding problem Message-ID: I was having trouble doing call forwarding from my SIP phone that is connected to FreeSWITCH. It turns out that my SIP phone is actually sending 302 Moved Temporarily responses, but my SIP gateway does not support 302 Moved Temporarily or SIP REFER messages. How do I get FreeSWITCH to forward calls without sending 302 Moved Temporarily or SIP REFER messages? Here is the SIP debug from our gateway: Received: INVITE sip:+19725357722 at ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0 Record-Route: v: SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236 record-route: f: ;tag=dc7-13c4-3ec95a-3ad03068-3ec95a t: i: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989 CSeq: 1 INVITE Max-Forwards: 18 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208 m: c: application/SDP l: 210 P-Asserted-Identity: Privacy: none v=0 o=- 540754816 540754816 IN IP4 199.173.111.141 s=- c=IN IP4 199.173.111.141 t=0 0 m=audio 30056 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208 From: ;tag=dc7-13c4-3ec95a-3ad03068-3ec95a To: Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357722 at 168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: ;tag=14E93594-2488 To: Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246 Supported: timer,resource-priority,replaces Min-SE:? 1800 Cisco-Guid: 1208058493-3631485406-2892683743-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259096880 Contact: Expires: 180 Allow-Events: telephone-event Max-Forwards: 17 P-Asserted-Identity: Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 2925 1780 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.141 t=0 0 m=audio 30056 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.141 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 24 15:08:00.367 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: ;tag=14E93594-2488 To: Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE Timestamp: 1259096880 0.000342 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M Content-Length: 0 Nov 24 15:08:00.419 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: ;tag=14E93594-2488 To: ;tag=49aF8vtgHme2c Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 P-Asserted-Identity: "19725357722" Nov 24 15:08:00.427 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357722 at 168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: ;tag=14E93594-2488 To: ;tag=49aF8vtgHme2c Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: 4802BACC-D87411DE-AC70D9DF-3419A306 at 168.75.202.246 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 _________________________________________________________________ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 From john_platts at hotmail.com Tue Nov 24 13:28:09 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 24 Nov 2009 15:28:09 -0600 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH In-Reply-To: <6F2A2A62-CE26-477F-B402-358F313A3EC3@jerris.com> References: , , <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org>, , <6F2A2A62-CE26-477F-B402-358F313A3EC3@jerris.com> Message-ID: I actually checked out revision 15654 today, and I was still getting problems with proxy media and bypass media in FreeSWITCH. ________________________________ > From: mike at jerris.com > Date: Tue, 24 Nov 2009 03:39:16 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH > > > > This was fixed in trunk yesterday about 8 hrs before you sent this message. (15619). Please update and try again. > > > Mike > > On Nov 23, 2009, at 11:33 PM, John Platts wrote: > > > I was using revision 15586. > > ---------------------------------------- > From: brian at freeswitch.org > Date: Mon, 23 Nov 2009 18:25:44 -0600 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH > > What rev exactly? > > /b > > On Nov 23, 2009, at 6:19 PM, John Platts wrote: > > > I actually checked out the latest version of FreeSWITCH in the SVN > repository. > > I have the following configured in /usr/local/freeswitch/conf/ > dialplan/default.xml: > > > _________________________________________________________________ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/ From brian at freeswitch.org Tue Nov 24 13:32:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Nov 2009 15:32:44 -0600 Subject: [Freeswitch-users] Call forwarding problem In-Reply-To: References: Message-ID: <633E77B1-2EC0-41A2-90C9-E884B59AFC99@freeswitch.org> You'll have to hairpin the media thru your machine usually if they won't accept either of those. /b On Nov 24, 2009, at 3:05 PM, John Platts wrote: > How do I get FreeSWITCH to forward calls without sending 302 Moved > Temporarily or SIP REFER messages? From brian at freeswitch.org Tue Nov 24 13:33:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Nov 2009 15:33:06 -0600 Subject: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH In-Reply-To: References: , , <00B80748-F9C6-450F-ADFA-FB65599FDB76@freeswitch.org>, , <6F2A2A62-CE26-477F-B402-358F313A3EC3@jerris.com> Message-ID: Are you sure you did a make current? and can you outline the issue in more detail? /b On Nov 24, 2009, at 3:28 PM, John Platts wrote: > > I actually checked out revision 15654 today, and I was still getting > problems with proxy media and bypass media in FreeSWITCH. From anthony.minessale at gmail.com Tue Nov 24 13:59:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Nov 2009 15:59:42 -0600 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B0C481A.8030309@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> Message-ID: <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> connect to FS with fs_cli Issue the command: /events MESSAGE_QUERY MESSAGE_WAITING then leave some voice mails probably you have a mis-configuration where the user/domain/profile cannot be resolved to the correct sofia profile to send the notify The event starts out as a freeswitch event and is translated into the notify by mod_sofia but only if it can match the event to a real sip user On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX wrote: > Hello, > > I have a similar problem with Freeswitch behind OpenSIPS as a load > balancer: > When registering, Freeeswitch does not send a MWI NOTIFY message for a > Phone which has voicemails. Even after recording a new voicemail there > is no NOTIFY message sent. And there are no error messages on the console. > > I have explicitely set > in the internal profile. > > When a phone is set up I get the following > Snom Phone REGISTER => OpenSIPS=> Freeswitch > Freeswitch OK => OpenSIPS=>Snom Phone > > Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch > Freeswitch 202 Accepted => OpenSIPS=>Snom Phone > > Snom Phone PUBLISH => OpenSIPS=> Freeswitch > Freeswitch 200 OK => OpenSIPS=>Snom Phone > So presence generally seems to work. > > But ngrepping the Network traffic there's no MWI NOTIFY message coming > from Freeswitch to any phone. > FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before > should be already there. > > Any idea how to force the NOTIFY messages? > > > Best regards > Peter > > Here's the debug Level9 output for nta and nua when a phone with VMs > registers, seems like there is no error in it: > > freeswitch at sip11.mydomain.com> nta: received REGISTER > sip:sip1.mydomain.com SIP/2.0 (CSeq 7) > nta: REGISTER (7) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x7fd5d409c8f0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x7fd5d409c8f0): sent signal r_respond > nua: nua_handle_destroy: entering > nua(0x7fd5d409c8f0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized > nua: nua_stack_set_params: entering > nta: sent 401 Unauthorized for REGISTER (7) > nta: timer set to 32000 ms > nua(0x7fd5d409c8f0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 6) > nta: REGISTER (6) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x905a80): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x905a80): sent signal r_respond > nua: nua_handle_destroy: entering > nua(0x905a80): recv signal r_respond 401 Unauthorized > nua(0x905a80): sent signal r_destroy > nua: nua_stack_set_params: entering > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 401 Unauthorized for REGISTER (6) > nua(0x905a80): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received PUBLISH sip:100 at sip1.mydomain.comSIP/2.0 (CSeq 3) > nta: PUBLISH (3) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x905f10): event i_publish 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x905f10): sent signal r_respond > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nua(0x905f10): recv signal r_respond 200 OK > nua: nua_stack_set_params: entering > nua(0x905f10): sent signal r_destroy > nta: sent 200 OK for PUBLISH (3) > nua(0x905f10): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received SUBSCRIBE sip:mod_sofia at 192.168.178.200:5062 SIP/2.0 (CSeq > 2) > nta: canonizing sip:mod_sofia at 192.168.178.200:5062 with contact > nta: SUBSCRIBE (2) going to existing leg > nua: nua_stack_process_request: entering > nta: sent 200 OK for SUBSCRIBE (2) > nua(0x905560): event i_subscribe 200 OK > nua: nua_application_event: entering > nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 8) > nta: REGISTER (8) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x7fd5dc073ba0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x7fd5dc073ba0): sent signal r_respond > nua(0x7fd5dc073ba0): recv signal r_respond 200 OK > nua: nua_stack_set_params: entering > nua: nua_handle_destroy: entering > nua(0x7fd5dc073ba0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 200 OK for REGISTER (8) > nua(0x7fd5dc073ba0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) > nta: REGISTER (7) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x8fc3d0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x8fc3d0): sent signal r_respond > nua(0x8fc3d0): recv signal r_respond 200 OK > nua: nua_handle_destroy: entering > nua: nua_stack_set_params: entering > nua(0x8fc3d0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 200 OK for REGISTER (7) > nua(0x8fc3d0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received SUBSCRIBE sip:100 at sip1.mydomain.com;user=phone > SIP/2.0 > (CSeq 1) > nta: SUBSCRIBE (1) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nta_leg_tcreate(0x7fd5dc03add0) > nua(0x7fd5dc078b70): adding notify usage with event message-summary > nua(0x7fd5dc078b70): event i_subscribe 100 Trying > nua: nua_application_event: entering > nua(): refresh notify after 3600 seconds (in [3600..3600]) > nua: nua_respond: entering > nua(0x7fd5dc078b70): sent signal r_respond > nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted > nua: nua_stack_set_params: entering > nta: sent 202 Accepted for SUBSCRIBE (1) > > > > > > mayamatakeshi schrieb: > > > > On 9/12/09, *mayamatakeshi* > > wrote: > > > > > > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris > > wrote: > > > > Following up, did a bug get created for this issue? > > > > > > Hello, > > yes. > > http://jira.freeswitch.org/browse/MODSOFIA-26 > > > > > > Just to simplify things in case someone searches the list: > > Issue was solved on rev 14851. > > Thank you all. > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/8dabf774/attachment-0002.html From john_platts at hotmail.com Tue Nov 24 14:04:07 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 24 Nov 2009 16:04:07 -0600 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript Message-ID: I have considered writing JavaScript code to bridge two calls together. However, I would like to perform custom handling of the 302 Moved Temporarily response. How do I handle the 302 Moved Temporarily response if I use JavaScript? _________________________________________________________________ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search?q=restaurants&form=MFESRP&publ=WLHMTAG&crea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 From anthony.minessale at gmail.com Tue Nov 24 14:20:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Nov 2009 16:20:03 -0600 Subject: [Freeswitch-users] Noise with openzap In-Reply-To: <3e6d7b0c0911240525o747f7b05y5c1f50ec6afe1179@mail.gmail.com> References: <3e6d7b0c0911240525o747f7b05y5c1f50ec6afe1179@mail.gmail.com> Message-ID: <191c3a030911241420m7f9d649dicc05a13171f4a05f@mail.gmail.com> you may want to try the latest release of both wanpipe and FS openzap is still a moving target since its in constant development from both the hardware and software end On Tue, Nov 24, 2009 at 7:25 AM, Steven Brown wrote: > Hi, > > I have an Ubuntu box running FS1.0.4 which has been processing a good > volume of calls between local users with soft phones (Xlite) and GSM > handsets via a number or Portech gateways, this has worked very well > for some time and audio quality is very good. > > I've now added a Sangoma A200 with 4 ports hooked up to 4 PSTN lines, > configured openzap and I can originate and answer calls on the the > openzap lines fine, however these calls via opezap all seem to suffer > from significant noise, the audio path works fine in both directions > but noise seems particularly bad at the local soft phone end. Quality > of all other calls through the box is fine though, any ideas > appreciated ?, > > NB A regular handset plugged directly into the PSTN lines has no problems > though > > Thanks > > Steve > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/6473f8e2/attachment-0002.html From stevendt at primrosebank.net Tue Nov 24 14:36:36 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 24 Nov 2009 22:36:36 -0000 Subject: [Freeswitch-users] Call Transfer Help Please References: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> Message-ID: <5A7C3038838142B5A07F181C193754F6@bp1.ad.bp.com> Hi Mike, thanks for the reply. I am using the pre-compiled Windows binary - is there a 1.0.5 pre-release of that yet ? FreeSwitch reports its version as 1.0.4 (14460) but this is not correct, I was sure that I had previously loaded a later SVN Version, but just did it again, unless I'm not doing it right, the version number does not seem to be getting updated. The current build in the precompiled binaries area is reported to be 15604 and I've downloaded and installed that - although when the installer runs it tells me that it is version 15376. Either way, the "Version" command in FreeSwitch reports 1.0.4 (14460). The Transfer still does not work for me from the extension which answers the call. Sorry if my earlier questions were unclear ... "What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ?" What is the correct "transfer" data string in features.xml ? I don't understand this question(s) I was looking for clarification of the second two arguments in the bind_meta_app data call, i.e, that the "b" and "s" were the correct values and also that the "is transfer" "transfer" data argument was "-bleg" That is, that the arguments in the default dialplan are correct for this scenario - which they appear to be based on your previous reply to my query. So, is there anything else that I can check to see why this is not working ? regards Dave ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 24, 2009 8:19 PM Subject: Re: [Freeswitch-users] Call Transfer Help Please On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: Hi, I'm trying to setup call transfer for a phone without a transfer button. I was on IRC last night and got some pointers to how this is setup in dialplan.xml and features.xml and what "bind meta app" does. Once it became clear how the transfer is initiated and that the transfer, in the default config, can only be initiated by the "b" leg of the call, I was able to make this work as configured in the defaults, i.e, to initiate a transfer (for an internal call) from the dialled extension to a new extension. Now the problem . . . I have an incoming PSTN line that rings a group of extensions, what I want to be able to do is to give whoever answers the PSTN call ability to transfer the call on to another extension. There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch extension of 1000, it rings the extension phones in the group. I'd hoped that the default transfer setup would handle this without modification - the incoming call on extension 1000 would be the "a" leg, the answering extension would be the "b" leg and a transfer from "b" would work as per the default config. This does not work for me though. I'm struggling a bit with the "bind meta app" options and can't seem to make it do what I want. Could someone please confirm that what I'm trying to do is feasible and perhaps suggest the right parameters to use in dialplan.xml and features.xml please ? Relevant section in the "is_transfer" section in features.xml And in default.xml from to I've tried posting a call log to the Pastebin (11252/3) but there was an error - it looks like the dump was too big. Not sure what the maximum size on pastebin dumps is ? My understanding (or lack of) of "a" and "b" are in the scenario described is not helping ... Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? Yes, the calling leg Is the answering extension the "b" leg ? Yes What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ? I don't understand this question What is the correct "transfer" data string in features.xml ? ditto Or am I totally on the wrong track here ? You should just need to make sure that the bind meta is called in this scenario so the b leg is able to do it, thats it. If it is possible to do what I want, and changes are required to the dialplan.xml and/or features.xml files, is it possible to have different logic in there such that the actions are different whether it is the "a" leg or "b" leg that's requesting the transfer ? regards Dave FreeSwitch Version 1.0.4 (14460) also, try the latest 1.0.5. pre release or svn trunk to confirm this is not an issue that has already been fixed. Mike ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/57d90a13/attachment-0002.html From Prometheus001 at gmx.net Tue Nov 24 14:56:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 24 Nov 2009 23:56:25 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> Message-ID: <4B0C6499.4060504@gmx.net> Anthony, thanks for the hint, I receive events like the following RECV EVENT Event-Name: MESSAGE_WAITING Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 FreeSWITCH-Hostname: sip11.mydomain.com FreeSWITCH-IPv4: 192.168.178.200 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-24 23:33:13 Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT Event-Date-Timestamp: 1259101993918617 Event-Calling-File: mod_voicemail.c Event-Calling-Function: update_mwi Event-Calling-Line-Number: 1738 MWI-Messages-Waiting: yes MWI-Message-Account: 200 at sip1.mydomain.com MWI-Voice-Message: 4/1 (0/0) I think the problem may be the Freeswitch cluster we are working with. All phones register with realm (e.g. 200 at sip1.mydomain.com). The FS hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. With xml_curl we ensure that for both domain names a directory entry is passed back. That way it works nicely with registering phones, receiving voicemails, recording voicemails etc. but not for MWI. For recording and querying voicemails we use the realm instead of the domain name and that way it works. When a voicemail has finished recording - and at the time the above message occurs - I see 2 directory xml_curl requests with Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id One I expect is for retrieving the MWI data and the other one for sending the VM email (which is sucessfully sent). Any hint how we can workaround this? Or is there a parameter to tell mod_voicemail that is should use the realm instead of the local hostname for sending MWI? Best regards Peter Anthony Minessale schrieb: > connect to FS with fs_cli > > Issue the command: > > /events MESSAGE_QUERY MESSAGE_WAITING > > then leave some voice mails > > probably you have a mis-configuration where the user/domain/profile > cannot be resolved to the correct > sofia profile to send the notify > > The event starts out as a freeswitch event and is translated into the > notify by mod_sofia but only if it can > match the event to a real sip user > > > > > On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX > wrote: > > Hello, > > I have a similar problem with Freeswitch behind OpenSIPS as a load > balancer: > When registering, Freeeswitch does not send a MWI NOTIFY message for a > Phone which has voicemails. Even after recording a new voicemail there > is no NOTIFY message sent. And there are no error messages on the > console. > > I have explicitely set > in the internal > profile. > > When a phone is set up I get the following > Snom Phone REGISTER => OpenSIPS=> Freeswitch > Freeswitch OK => OpenSIPS=>Snom Phone > > Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch > Freeswitch 202 Accepted => OpenSIPS=>Snom Phone > > Snom Phone PUBLISH => OpenSIPS=> Freeswitch > Freeswitch 200 OK => OpenSIPS=>Snom Phone > So presence generally seems to work. > > But ngrepping the Network traffic there's no MWI NOTIFY message coming > from Freeswitch to any phone. > FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before > should be already there. > > Any idea how to force the NOTIFY messages? > > > Best regards > Peter > > Here's the debug Level9 output for nta and nua when a phone with VMs > registers, seems like there is no error in it: > > freeswitch at sip11.mydomain.com > > nta: received REGISTER > sip:sip1.mydomain.com SIP/2.0 (CSeq 7) > nta: REGISTER (7) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x7fd5d409c8f0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x7fd5d409c8f0): sent signal r_respond > nua: nua_handle_destroy: entering > nua(0x7fd5d409c8f0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized > nua: nua_stack_set_params: entering > nta: sent 401 Unauthorized for REGISTER (7) > nta: timer set to 32000 ms > nua(0x7fd5d409c8f0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received REGISTER sip:sip1.mydomain.com > SIP/2.0 (CSeq 6) > nta: REGISTER (6) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x905a80): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x905a80): sent signal r_respond > nua: nua_handle_destroy: entering > nua(0x905a80): recv signal r_respond 401 Unauthorized > nua(0x905a80): sent signal r_destroy > nua: nua_stack_set_params: entering > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 401 Unauthorized for REGISTER (6) > nua(0x905a80): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received PUBLISH sip:100 at sip1.mydomain.com > SIP/2.0 (CSeq 3) > nta: PUBLISH (3) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x905f10): event i_publish 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x905f10): sent signal r_respond > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nua(0x905f10): recv signal r_respond 200 OK > nua: nua_stack_set_params: entering > nua(0x905f10): sent signal r_destroy > nta: sent 200 OK for PUBLISH (3) > nua(0x905f10): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received SUBSCRIBE sip:mod_sofia at 192.168.178.200:5062 > SIP/2.0 (CSeq 2) > nta: canonizing sip:mod_sofia at 192.168.178.200:5062 > with contact > nta: SUBSCRIBE (2) going to existing leg > nua: nua_stack_process_request: entering > nta: sent 200 OK for SUBSCRIBE (2) > nua(0x905560): event i_subscribe 200 OK > nua: nua_application_event: entering > nta: received REGISTER sip:sip1.mydomain.com > SIP/2.0 (CSeq 8) > nta: REGISTER (8) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x7fd5dc073ba0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x7fd5dc073ba0): sent signal r_respond > nua(0x7fd5dc073ba0): recv signal r_respond 200 OK > nua: nua_stack_set_params: entering > nua: nua_handle_destroy: entering > nua(0x7fd5dc073ba0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 200 OK for REGISTER (8) > nua(0x7fd5dc073ba0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received REGISTER sip:sip1.mydomain.com > SIP/2.0 (CSeq 7) > nta: REGISTER (7) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nua(0x8fc3d0): event i_register 100 Trying > nua: nua_application_event: entering > nua: nua_respond: entering > nua(0x8fc3d0): sent signal r_respond > nua(0x8fc3d0): recv signal r_respond 200 OK > nua: nua_handle_destroy: entering > nua: nua_stack_set_params: entering > nua(0x8fc3d0): sent signal r_destroy > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nta: sent 200 OK for REGISTER (7) > nua(0x8fc3d0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: received SUBSCRIBE sip:100 at sip1.mydomain.com > ;user=phone SIP/2.0 > (CSeq 1) > nta: SUBSCRIBE (1) going to a default leg > nua: nua_stack_process_request: entering > nua: nh_create: entering > nua: nh_create_handle: entering > nua: nua_stack_set_params: entering > nta_leg_tcreate(0x7fd5dc03add0) > nua(0x7fd5dc078b70): adding notify usage with event message-summary > nua(0x7fd5dc078b70): event i_subscribe 100 Trying > nua: nua_application_event: entering > nua(): refresh notify after 3600 seconds (in [3600..3600]) > nua: nua_respond: entering > nua(0x7fd5dc078b70): sent signal r_respond > nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted > nua: nua_stack_set_params: entering > nta: sent 202 Accepted for SUBSCRIBE (1) > > > > > > mayamatakeshi schrieb: > > > > On 9/12/09, *mayamatakeshi* > > >> wrote: > > > > > > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris > > > >> wrote: > > > > Following up, did a bug get created for this issue? > > > > > > Hello, > > yes. > > http://jira.freeswitch.org/browse/MODSOFIA-26 > > > > > > Just to simplify things in case someone searches the list: > > Issue was solved on rev 14851. > > Thank you all. > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From john_platts at hotmail.com Tue Nov 24 15:24:23 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 24 Nov 2009 17:24:23 -0600 Subject: [Freeswitch-users] Call forwarding problem In-Reply-To: <633E77B1-2EC0-41A2-90C9-E884B59AFC99@freeswitch.org> References: , <633E77B1-2EC0-41A2-90C9-E884B59AFC99@freeswitch.org> Message-ID: Is there any way to tell FreeSWITCH to do the following when 302 Moved Temporarily is sent to FreeSWITCH: - End the session between FreeSWITCH and the phone - Bridge the original session with the number that the call is forwarded to ---------------------------------------- > From: brian at freeswitch.org > Date: Tue, 24 Nov 2009 15:32:44 -0600 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call forwarding problem > > You'll have to hairpin the media thru your machine usually if they > won't accept either of those. > > /b > > On Nov 24, 2009, at 3:05 PM, John Platts wrote: > >> How do I get FreeSWITCH to forward calls without sending 302 Moved >> Temporarily or SIP REFER messages? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/ From fanatikneo at gmx.de Tue Nov 24 14:35:17 2009 From: fanatikneo at gmx.de (Jan Thiemo Fricke) Date: Tue, 24 Nov 2009 23:35:17 +0100 Subject: [Freeswitch-users] mod_conference kick to abort invitations Message-ID: <000001ca6d56$66037c80$320a7580$@de> Hi members, I'm controlling freeswitch with the conference module via xmlrpc. Is it desired that the kick command can only kick users that are connected to the conference? Is there no chance abort an invitation? The kick command has no effect until the person I invited with the dial command is connected. Thanks in advance! Jan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/1dc073b9/attachment-0002.html From stevendt at primrosebank.net Tue Nov 24 16:11:24 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 25 Nov 2009 00:11:24 -0000 Subject: [Freeswitch-users] Call Transfer Help Please References: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> <5A7C3038838142B5A07F181C193754F6@bp1.ad.bp.com> Message-ID: Hi again folks, I have posted a dump into the Pastebin (11276), could someone have a look and perhaps suggest where the problem might be please ? I'm sure you'll be able to work it out, but the log is for a call where :- incoming on PSTN Line (ext 1000) Group exts, 111, 1001, 1001 Answered on 111 and requested transfer to 1001 with no success regards Dave ----- Original Message ----- From: Dave Stevenson To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 24, 2009 10:36 PM Subject: Re: [Freeswitch-users] Call Transfer Help Please Hi Mike, thanks for the reply. I am using the pre-compiled Windows binary - is there a 1.0.5 pre-release of that yet ? FreeSwitch reports its version as 1.0.4 (14460) but this is not correct, I was sure that I had previously loaded a later SVN Version, but just did it again, unless I'm not doing it right, the version number does not seem to be getting updated. The current build in the precompiled binaries area is reported to be 15604 and I've downloaded and installed that - although when the installer runs it tells me that it is version 15376. Either way, the "Version" command in FreeSwitch reports 1.0.4 (14460). The Transfer still does not work for me from the extension which answers the call. Sorry if my earlier questions were unclear ... "What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ?" What is the correct "transfer" data string in features.xml ? I don't understand this question(s) I was looking for clarification of the second two arguments in the bind_meta_app data call, i.e, that the "b" and "s" were the correct values and also that the "is transfer" "transfer" data argument was "-bleg" That is, that the arguments in the default dialplan are correct for this scenario - which they appear to be based on your previous reply to my query. So, is there anything else that I can check to see why this is not working ? regards Dave ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 24, 2009 8:19 PM Subject: Re: [Freeswitch-users] Call Transfer Help Please On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: Hi, I'm trying to setup call transfer for a phone without a transfer button. I was on IRC last night and got some pointers to how this is setup in dialplan.xml and features.xml and what "bind meta app" does. Once it became clear how the transfer is initiated and that the transfer, in the default config, can only be initiated by the "b" leg of the call, I was able to make this work as configured in the defaults, i.e, to initiate a transfer (for an internal call) from the dialled extension to a new extension. Now the problem . . . I have an incoming PSTN line that rings a group of extensions, what I want to be able to do is to give whoever answers the PSTN call ability to transfer the call on to another extension. There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch extension of 1000, it rings the extension phones in the group. I'd hoped that the default transfer setup would handle this without modification - the incoming call on extension 1000 would be the "a" leg, the answering extension would be the "b" leg and a transfer from "b" would work as per the default config. This does not work for me though. I'm struggling a bit with the "bind meta app" options and can't seem to make it do what I want. Could someone please confirm that what I'm trying to do is feasible and perhaps suggest the right parameters to use in dialplan.xml and features.xml please ? Relevant section in the "is_transfer" section in features.xml And in default.xml from to I've tried posting a call log to the Pastebin (11252/3) but there was an error - it looks like the dump was too big. Not sure what the maximum size on pastebin dumps is ? My understanding (or lack of) of "a" and "b" are in the scenario described is not helping ... Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? Yes, the calling leg Is the answering extension the "b" leg ? Yes What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ? I don't understand this question What is the correct "transfer" data string in features.xml ? ditto Or am I totally on the wrong track here ? You should just need to make sure that the bind meta is called in this scenario so the b leg is able to do it, thats it. If it is possible to do what I want, and changes are required to the dialplan.xml and/or features.xml files, is it possible to have different logic in there such that the actions are different whether it is the "a" leg or "b" leg that's requesting the transfer ? regards Dave FreeSwitch Version 1.0.4 (14460) also, try the latest 1.0.5. pre release or svn trunk to confirm this is not an issue that has already been fixed. Mike ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/f4e74177/attachment-0002.html From lei.tlfly at gmail.com Tue Nov 24 18:03:04 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 25 Nov 2009 10:03:04 +0800 Subject: [Freeswitch-users] How to run IVR application In-Reply-To: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> References: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> Message-ID: <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> you can do this in follow steps: 1.edit default.xml diaplan config file in your fs config directory(FS/conf/dialplan/default.xml), and section 2. edit your ivr script, your can refer to http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in lua. 3. connect your sip phone to fs, and dial 114, this will launch your ivr application 2009/11/24 ovvenkat > Hi to all, > > I am very new this platform . I just downloaded freeswitch to my windows xp > machine , compiled successfully and run. After that I dont have any idea > what to do :( . I am not finding simple kind of tutorial on the net. could > you please suggest me, how I have to proceed. My requirement is; I need to > run IVR application on machine using SIP phone. I am very sorry to my bad > English. > > Thanks and Regards, > Venkat. > > -- > > If you have come to help me, you are wasting your time. > If you have come to because your liberation is bound up in mine, we can > work together. > > > Regards > Venkatesan OV. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/c8e49dd1/attachment-0002.html From jlenk at frontiernet.net Tue Nov 24 19:54:28 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 24 Nov 2009 19:54:28 -0800 (PST) Subject: [Freeswitch-users] Call Transfer Help Please In-Reply-To: References: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com> <5A7C3038838142B5A07F181C193754F6@bp1.ad.bp.com> Message-ID: <1259121268571-4062810.post@n2.nabble.com> I do not see the meta app getting added in your log -> Dialplan: sofia/internal/1000 at 192.168.1.50 Action bind_meta_app(* Without this no meta actions will occur Dave Stevenson wrote: > > Hi again folks, > > I have posted a dump into the Pastebin (11276), could someone have a look > and perhaps suggest where the problem might be please ? > > I'm sure you'll be able to work it out, but the log is for a call where :- > > incoming on PSTN Line (ext 1000) > Group exts, 111, 1001, 1001 > Answered on 111 and requested transfer to 1001 with no success > > regards > Dave > > > ----- Original Message ----- > From: Dave Stevenson > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, November 24, 2009 10:36 PM > Subject: Re: [Freeswitch-users] Call Transfer Help Please > > > Hi Mike, > > thanks for the reply. I am using the pre-compiled Windows binary - is > there a 1.0.5 pre-release of that yet ? > > FreeSwitch reports its version as 1.0.4 (14460) but this is not correct, > I was sure that I had previously loaded a later SVN Version, but just did > it again, unless I'm not doing it right, the version number does not seem > to be getting updated. The current build in the precompiled binaries area > is reported to be 15604 and I've downloaded and installed that - although > when the installer runs it tells me that it is version 15376. Either way, > the "Version" command in FreeSwitch reports 1.0.4 (14460). > > The Transfer still does not work for me from the extension which answers > the call. > > Sorry if my earlier questions were unclear ... > "What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml > ?" > What is the correct "transfer" data string in features.xml ? > I don't understand this question(s) > > I was looking for clarification of the second two arguments in the > bind_meta_app data call, i.e, that the "b" and "s" were the correct values > and also that the "is transfer" "transfer" data argument was "-bleg" > > That is, that the arguments in the default dialplan are correct for this > scenario - which they appear to be based on your previous reply to my > query. > > So, is there anything else that I can check to see why this is not > working ? > > > regards > Dave > > > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, November 24, 2009 8:19 PM > Subject: Re: [Freeswitch-users] Call Transfer Help Please > > > > > On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: > > > Hi, > > I'm trying to setup call transfer for a phone without a transfer > button. I was on IRC last night and got some pointers to how this is setup > in dialplan.xml and features.xml and what "bind meta app" does. > > Once it became clear how the transfer is initiated and that the > transfer, in the default config, can only be initiated by the "b" leg of > the call, I was able to make this work as configured in the defaults, i.e, > to initiate a transfer (for an internal call) from the dialled extension > to a new extension. > > Now the problem . . . > > I have an incoming PSTN line that rings a group of extensions, what > I want to be able to do is to give whoever answers the PSTN call ability > to transfer the call on to another extension. > > There is an ATA (Linksys SPA3101) set up on the PSTN line with a > FreeSwitch extension of 1000, it rings the extension phones in the group. > > I'd hoped that the default transfer setup would handle this without > modification - the incoming call on extension 1000 would be the "a" leg, > the answering extension would be the "b" leg and a transfer from "b" would > work as per the default config. This does not work for me though. > > I'm struggling a bit with the "bind meta app" options and can't seem > to make it do what I want. > > Could someone please confirm that what I'm trying to do is feasible > and perhaps suggest the right parameters to use in dialplan.xml and > features.xml please ? > > Relevant section in the "is_transfer" section in features.xml > > > And in default.xml from > to > > > I've tried posting a call log to the Pastebin (11252/3) but there > was an error - it looks like the dump was too big. Not sure what the > maximum size on pastebin dumps is ? > > > My understanding (or lack of) of "a" and "b" are in the scenario > described is not helping ... > > Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? > > > Yes, the calling leg > > > Is the answering extension the "b" leg ? > > > Yes > > > What are the correct LISTEN_TO and RESPOND_ON entries in > dialplan.xml ? > > > I don't understand this question > > > What is the correct "transfer" data string in features.xml ? > > > > ditto > > > Or am I totally on the wrong track here ? > > > > You should just need to make sure that the bind meta is called in this > scenario so the b leg is able to do it, thats it. > > > If it is possible to do what I want, and changes are required to the > dialplan.xml and/or features.xml files, is it possible to have different > logic in there such that the actions are different whether it is the "a" > leg or "b" leg that's requesting the transfer ? > > regards > Dave > > FreeSwitch Version 1.0.4 (14460) > > > also, try the latest 1.0.5. pre release or svn trunk to confirm this > is not an issue that has already been fixed. > > > Mike > > > > > ---------------------------------------------------------------------------- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------------ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Call-Transfer-Help-Please-tp4056930p4062810.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Tue Nov 24 19:55:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Nov 2009 19:55:31 -0800 Subject: [Freeswitch-users] How to run IVR application In-Reply-To: <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> References: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> Message-ID: <87f2f3b90911241955v4e726111ked993c8dbb556f99@mail.gmail.com> On Tue, Nov 24, 2009 at 6:03 PM, Lei Tang wrote: > you can do this in follow steps: > 1.edit default.xml diaplan config file in your fs config > directory(FS/conf/dialplan/default.xml), and section > > > > > > 2. edit your ivr script, your can refer to > http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in > lua. > 3. connect your sip phone to fs, and dial 114, this will launch your ivr > application > > You can also do IVRs with static XML. I recommend you try out the demo IVR by dialing 5000. Now go look at the two main files that we used to build that IVR: conf/autoload_configs/ivr.conf.xml (menu structure) conf/lang/en/demo/demo-ivr.xml (phrase macros) it's overwhelming at first, however once you get the hang of it you'll appreciate how powerful it is. The wiki and the sample XML config files have lots of information so be sure to read as much as you can and try things. You can't break anything. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091124/3e78aadb/attachment-0002.html From jlenk at frontiernet.net Tue Nov 24 20:35:59 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 24 Nov 2009 20:35:59 -0800 (PST) Subject: [Freeswitch-users] register timeout / cisco 7960 In-Reply-To: <367751820911231434j36b9846dk46d058ddb77c634@mail.gmail.com> References: <367751820911231434j36b9846dk46d058ddb77c634@mail.gmail.com> Message-ID: <1259123759153-4062958.post@n2.nabble.com> People commonly use 60 sec registration refreshes to keep NAT routers happy Phillip Jones-2 wrote: > > hi there, > > I have set up some cisco 7960 up with fs. They work fine - but the only > way > I can keep them registered is to set the "timer_register_expires" in the > Cisco cfg file to something really short like 10s. > > Does anyone know the default register timeout for fs? And where I might > change this in fs? > > Thanks! > > > Phil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/register-timeout-cisco-7960-tp4054546p4062958.html Sent from the freeswitch-users mailing list archive at Nabble.com. From thangappan143 at gmail.com Tue Nov 24 22:09:24 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Wed, 25 Nov 2009 11:39:24 +0530 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> Message-ID: <7aa29e790911242209w7ee2912bhbde4b3475147628d@mail.gmail.com> FreeSWITCH version: freeswitch 1.0.4 I am using ESL library I attached the example Perl script which does the same steps that I posted already. ( Sample.pl) I supplied the log , Here I attached the output of the ESL log. (Output.txt) Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. But in the output I got only 2,4,5,4 ( DTMF 1 is missed) Output of Perl code could be like Wait for response time out EVENT [COMMAND] Wait for response time out EVENT [DTMF] DTMF digit 2 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 4 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 5 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 4 (2000) Wait for inter digit time out Buffer: 2454 BYE Why the first digit(1) is missed here? In ESL log there is no digit called 1 why? Why the COMMAND event is received instead of DTMF? How can I get all DTMF digits? On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M wrote: > The reason for waiting only for DTMF event is to handle the time outs in > the IVR concept like response and inter digit time out. Using our own logic > we 10 voice files in each play back if the voice files are more than 10. Now > it works fine. > > Now the new problem has been raised. The problem is we are filtering only > for DTMF events but we are getting COMMAND event . Because of this the DTMF > digits are missing at the time . I am not able to proceed further. We are > in the critical situation. > > Why this command event is occurring? > How can I restrict this? > What are the information it has? > How can I get all the information in it ? ( If command event has info) > > Help me............ > > > On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: > >> I am waiting only for DTMF events. That's why I am setting freeswitch >> variable for knowing whether the playback has done. >> >> My question is "why this freeswitch variable is not setting properly when >> I play back more than 10 files using playback_delimiter option?". >> >> When I play back lesser than ten voice files the variable has been set >> properly. What could be the reason? >> >> >> >> ---------- Forwarded message ---------- >> From: Thangappan.M >> Date: Sat, Nov 21, 2009 at 2:52 PM >> Subject: Problem while playing more than 10 voice files using playback >> To: freeswitch-users >> >> >> Dear all, >> >> I am in the process of implementing IVR using event outbound >> socket (async mode). >> I have implemented using Perl language. >> >> I did the following steps: >> => Set the playback_delimiter variable >> => Set the playback_sleep_val variable >> => Set the event lock as true >> => Set the freeswitch ( my own) variable as zero >> => Wait in the loop until the variable is been set as >> zero >> => Playback the voice files ( Here I combined the >> voice files with the delimiter value if more than one voice files are there) >> => Set the freeswitch(my own) variable as true ( This >> is used to identify whether the voice files are played >> successfully). >> => Wait in the loop until the variable is been set as >> one. >> => Set the Event lock as false >> >> => Trying to get the DTMF digits ( Have a assurance >> that all the voice files are played). >> >> The problem is, >> >> The above steps are working fine when the voice file count is >> lesser than or equal to 10. After the voice files are played only the >> variable(my own freeswitch) is set. Based on the variable I am doing further >> things. >> >> But when I tried to give the voice files count of more than >> 10 the variable has been set while starting to play back the first voice >> file itself . Because of this I am not able to proceed further. >> >> *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* >> >> *NOTE*: I also referred mod_file_string documentation. In that they >> specified 128 files can be used to play back the voice files using >> playback_delimiter option. >> >> Please help me................? >> Thanks in advance. >> >> >> -- >> Regards, >> Thangappan.M >> >> >> >> -- >> Regards, >> Thangappan.M >> > > > > -- > Regards, > Thangappan.M > -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/e557b247/attachment-0002.html -------------- next part -------------- [DEBUG] src/esl.c:995 esl_send() SEND myevents [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK Events Enabled] [DEBUG] src/esl.c:995 esl_send() SEND filter Event-Name DTMF [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK filter added. [Event-Name]=[DTMF]] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: answer event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: set execute-app-arg: playback_terminators=0123456789 event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: set execute-app-arg: playback_delimiter=! event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: set execute-app-arg: playback_sleep_val=1 event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: set execute-app-arg: IVRFlag=0 event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [7] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: playback execute-app-arg: /FreeSWITCH-IVR/VoiceFile/English/iconnectform.wav!/FreeSWITCH-IVR/VoiceFile/English/dial1.wav!/FreeSWITCH-IVR/VoiceFile/English/tcurrform.wav!/FreeSWITCH-IVR/VoiceFile/English/dial2.wav!/FreeSWITCH-IVR/VoiceFile/English/licpayform.wav!/FreeSWITCH-IVR/VoiceFile/English/dial3.wav!/FreeSWITCH-IVR/VoiceFile/English/bsnlform.wav!/FreeSWITCH-IVR/VoiceFile/English/dial4.wav!/FreeSWITCH-IVR/VoiceFile/English/encashform.wav!/FreeSWITCH-IVR/VoiceFile/English/dial5.wav event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND sendmsg call-command: execute execute-app-name: set execute-app-arg: IVRFlag=1 event-lock: true [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1516] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1516] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Name] = [DTMF] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Core-UUID] = [cfbc5248-d983-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-Hostname] = [debian] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv4] = [192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv6] = [::1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Local] = [2009-11-25 11:11:49] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-GMT] = [Wed, 25 Nov 2009 05:41:49 GMT] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Timestamp] = [1259127709436105] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-File] = [switch_channel.c] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Function] = [switch_channel_dequeue_dtmf] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Line-Number] = [388] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State] = [CS_EXECUTE] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State-Number] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Presence-Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Answer-State] = [answered] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Username] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Dialplan] = [XML] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Name] = [thangappan] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Number] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Network-Addr] = [192.168.8.100] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Destination-Number] = [200] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Source] = [mod_sofia] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Context] = [default] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Index] = [1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Answered-Time] = [1259127708528257] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Media-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Hangup-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Transfer-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Screen-Bit] = [true] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Name] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Number] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Digit] = [2] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Duration] = [2000] [DEBUG] src/esl.c:963 esl_recv_event() RECV EVENT Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-25 11:11:49 Event-Date-GMT: Wed, 25 Nov 2009 05:41:49 GMT Event-Date-Timestamp: 1259127709436105 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012 at 192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012 at 192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 2 DTMF-Duration: 2000 [DEBUG] src/esl.c:971 esl_recv_event() RECV MESSAGE Event-Name: COMMAND Content-Length: 1516 Content-Type: text/event-plain Content-Length: 1516 Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-11-25%2011%3A11%3A49 Event-Date-GMT: Wed,%2025%20Nov%202009%2005%3A41%3A49%20GMT Event-Date-Timestamp: 1259127709436105 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012%40192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012%40192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 2 DTMF-Duration: 2000 [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1516] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Name] = [DTMF] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Core-UUID] = [cfbc5248-d983-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-Hostname] = [debian] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv4] = [192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv6] = [::1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Local] = [2009-11-25 11:11:49] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-GMT] = [Wed, 25 Nov 2009 05:41:49 GMT] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Timestamp] = [1259127709636074] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-File] = [switch_channel.c] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Function] = [switch_channel_dequeue_dtmf] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Line-Number] = [388] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State] = [CS_EXECUTE] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State-Number] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Presence-Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Answer-State] = [answered] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Username] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Dialplan] = [XML] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Name] = [thangappan] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Number] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Network-Addr] = [192.168.8.100] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Destination-Number] = [200] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Source] = [mod_sofia] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Context] = [default] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Index] = [1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Answered-Time] = [1259127708528257] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Media-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Hangup-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Transfer-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Screen-Bit] = [true] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Name] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Number] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Digit] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Duration] = [2000] [DEBUG] src/esl.c:963 esl_recv_event() RECV EVENT Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-25 11:11:49 Event-Date-GMT: Wed, 25 Nov 2009 05:41:49 GMT Event-Date-Timestamp: 1259127709636074 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012 at 192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012 at 192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 4 DTMF-Duration: 2000 [DEBUG] src/esl.c:971 esl_recv_event() RECV MESSAGE Event-Name: COMMAND Content-Length: 1516 Content-Type: text/event-plain Content-Length: 1516 Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-11-25%2011%3A11%3A49 Event-Date-GMT: Wed,%2025%20Nov%202009%2005%3A41%3A49%20GMT Event-Date-Timestamp: 1259127709636074 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012%40192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012%40192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 4 DTMF-Duration: 2000 [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1516] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Name] = [DTMF] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Core-UUID] = [cfbc5248-d983-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-Hostname] = [debian] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv4] = [192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv6] = [::1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Local] = [2009-11-25 11:11:51] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-GMT] = [Wed, 25 Nov 2009 05:41:51 GMT] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Timestamp] = [1259127711031853] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-File] = [switch_channel.c] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Function] = [switch_channel_dequeue_dtmf] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Line-Number] = [388] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State] = [CS_EXECUTE] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State-Number] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Presence-Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Answer-State] = [answered] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Username] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Dialplan] = [XML] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Name] = [thangappan] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Number] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Network-Addr] = [192.168.8.100] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Destination-Number] = [200] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Source] = [mod_sofia] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Context] = [default] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Index] = [1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Answered-Time] = [1259127708528257] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Media-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Hangup-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Transfer-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Screen-Bit] = [true] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Name] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Number] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Digit] = [5] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Duration] = [2000] [DEBUG] src/esl.c:963 esl_recv_event() RECV EVENT Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-25 11:11:51 Event-Date-GMT: Wed, 25 Nov 2009 05:41:51 GMT Event-Date-Timestamp: 1259127711031853 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012 at 192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012 at 192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 5 DTMF-Duration: 2000 [DEBUG] src/esl.c:971 esl_recv_event() RECV MESSAGE Event-Name: COMMAND Content-Length: 1516 Content-Type: text/event-plain Content-Length: 1516 Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-11-25%2011%3A11%3A51 Event-Date-GMT: Wed,%2025%20Nov%202009%2005%3A41%3A51%20GMT Event-Date-Timestamp: 1259127711031853 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012%40192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012%40192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 5 DTMF-Duration: 2000 [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1516] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Name] = [DTMF] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Core-UUID] = [cfbc5248-d983-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-Hostname] = [debian] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv4] = [192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv6] = [::1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Local] = [2009-11-25 11:11:52] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-GMT] = [Wed, 25 Nov 2009 05:41:52 GMT] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Date-Timestamp] = [1259127712831585] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-File] = [switch_channel.c] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Function] = [switch_channel_dequeue_dtmf] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Event-Calling-Line-Number] = [388] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State] = [CS_EXECUTE] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-State-Number] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Presence-Call-Direction] = [inbound] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Answer-State] = [answered] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Read-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Name] = [PCMA] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Channel-Write-Codec-Rate] = [8000] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Username] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Dialplan] = [XML] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Name] = [thangappan] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Caller-ID-Number] = [1012] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Network-Addr] = [192.168.8.100] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Destination-Number] = [200] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Unique-ID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Source] = [mod_sofia] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Context] = [default] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Name] = [sofia/internal/1012 at 192.168.1.222] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Index] = [1] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Profile-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Created-Time] = [1259127708480239] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Answered-Time] = [1259127708528257] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Progress-Media-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Hangup-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Channel-Transfer-Time] = [0] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Screen-Bit] = [true] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Name] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [Caller-Privacy-Hide-Number] = [false] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Digit] = [4] [DEBUG] src/esl.c:938 esl_recv_event() RECV INNER HEADER [DTMF-Duration] = [2000] [DEBUG] src/esl.c:963 esl_recv_event() RECV EVENT Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-25 11:11:52 Event-Date-GMT: Wed, 25 Nov 2009 05:41:52 GMT Event-Date-Timestamp: 1259127712831585 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012 at 192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012 at 192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 4 DTMF-Duration: 2000 [DEBUG] src/esl.c:971 esl_recv_event() RECV MESSAGE Event-Name: COMMAND Content-Length: 1516 Content-Type: text/event-plain Content-Length: 1516 Event-Name: DTMF Core-UUID: cfbc5248-d983-11de-ae1f-af1380a5f9d0 FreeSWITCH-Hostname: debian FreeSWITCH-IPv4: 192.168.1.222 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-11-25%2011%3A11%3A52 Event-Date-GMT: Wed,%2025%20Nov%202009%2005%3A41%3A52%20GMT Event-Date-Timestamp: 1259127712831585 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 388 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1012%40192.168.1.222 Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Caller-Username: 1012 Caller-Dialplan: XML Caller-Caller-ID-Name: thangappan Caller-Caller-ID-Number: 1012 Caller-Network-Addr: 192.168.8.100 Caller-Destination-Number: 200 Caller-Unique-ID: 396a4dac-d985-11de-ae1f-af1380a5f9d0 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/1012%40192.168.1.222 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1259127708480239 Caller-Channel-Created-Time: 1259127708480239 Caller-Channel-Answered-Time: 1259127708528257 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: 4 DTMF-Duration: 2000 [DEBUG] src/esl.c:995 esl_send() SEND api uuid_getvar 396a4dac-d985-11de-ae1f-af1380a5f9d0 IVRFlag [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Type] = [text/disconnect-notice] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Controlled-Session-UUID] = [396a4dac-d985-11de-ae1f-af1380a5f9d0] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Disposition] = [disconnect] [DEBUG] src/esl.c:852 esl_recv_event() RECV HEADER [Content-Length] = [67] -------------- next part -------------- A non-text attachment was scrubbed... Name: Sample.pl Type: text/x-perl Size: 4984 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/e557b247/attachment-0002.bin From thangappan143 at gmail.com Tue Nov 24 22:18:27 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Wed, 25 Nov 2009 11:48:27 +0530 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911242209w7ee2912bhbde4b3475147628d@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> <7aa29e790911242209w7ee2912bhbde4b3475147628d@mail.gmail.com> Message-ID: <7aa29e790911242218l580b90eem3ec50676dfbc5536@mail.gmail.com> The example script is there in the following link http://pastebin.com/f332f2fda In the previous post I have attached it. But it was not shown. 2009/11/25 Thangappan.M > FreeSWITCH version: freeswitch 1.0.4 > I am using ESL library > I attached the example Perl script which does the same steps that I posted > already. ( Sample.pl) > I supplied the log , Here I attached the output of the ESL log. > (Output.txt) > > Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. > But in the output I got only 2,4,5,4 ( DTMF 1 is missed) > > Output of Perl code could be like > > Wait for response time out > EVENT [COMMAND] > Wait for response time out > EVENT [DTMF] > DTMF digit 2 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 5 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > Buffer: 2454 > BYE > > Why the first digit(1) is missed here? > In ESL log there is no digit called 1 why? > Why the COMMAND event is received instead of DTMF? > How can I get all DTMF digits? > > > > > > > > > > > > > > > > > On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M wrote: > >> The reason for waiting only for DTMF event is to handle the time outs in >> the IVR concept like response and inter digit time out. Using our own logic >> we 10 voice files in each play back if the voice files are more than 10. Now >> it works fine. >> >> Now the new problem has been raised. The problem is we are filtering only >> for DTMF events but we are getting COMMAND event . Because of this the DTMF >> digits are missing at the time . I am not able to proceed further. We are >> in the critical situation. >> >> Why this command event is occurring? >> How can I restrict this? >> What are the information it has? >> How can I get all the information in it ? ( If command event has info) >> >> Help me............ >> >> >> On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: >> >>> I am waiting only for DTMF events. That's why I am setting freeswitch >>> variable for knowing whether the playback has done. >>> >>> My question is "why this freeswitch variable is not setting properly when >>> I play back more than 10 files using playback_delimiter option?". >>> >>> When I play back lesser than ten voice files the variable has been set >>> properly. What could be the reason? >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Thangappan.M >>> Date: Sat, Nov 21, 2009 at 2:52 PM >>> Subject: Problem while playing more than 10 voice files using playback >>> To: freeswitch-users >>> >>> >>> Dear all, >>> >>> I am in the process of implementing IVR using event outbound >>> socket (async mode). >>> I have implemented using Perl language. >>> >>> I did the following steps: >>> => Set the playback_delimiter variable >>> => Set the playback_sleep_val variable >>> => Set the event lock as true >>> => Set the freeswitch ( my own) variable as zero >>> => Wait in the loop until the variable is been set as >>> zero >>> => Playback the voice files ( Here I combined the >>> voice files with the delimiter value if more than one voice files are there) >>> => Set the freeswitch(my own) variable as true ( This >>> is used to identify whether the voice files are played >>> successfully). >>> => Wait in the loop until the variable is been set as >>> one. >>> => Set the Event lock as false >>> >>> => Trying to get the DTMF digits ( Have a assurance >>> that all the voice files are played). >>> >>> The problem is, >>> >>> The above steps are working fine when the voice file count >>> is lesser than or equal to 10. After the voice files are played only the >>> variable(my own freeswitch) is set. Based on the variable I am doing further >>> things. >>> >>> But when I tried to give the voice files count of more than >>> 10 the variable has been set while starting to play back the first voice >>> file itself . Because of this I am not able to proceed further. >>> >>> *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* >>> >>> *NOTE*: I also referred mod_file_string documentation. In that they >>> specified 128 files can be used to play back the voice files using >>> playback_delimiter option. >>> >>> Please help me................? >>> Thanks in advance. >>> >>> >>> -- >>> Regards, >>> Thangappan.M >>> >>> >>> >>> -- >>> Regards, >>> Thangappan.M >>> >> >> >> >> -- >> Regards, >> Thangappan.M >> > > > > -- > Regards, > Thangappan.M > -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/043e14f4/attachment-0002.html From mike at jerris.com Tue Nov 24 22:34:13 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 01:34:13 -0500 Subject: [Freeswitch-users] Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911242218l580b90eem3ec50676dfbc5536@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> <7aa29e790911242209w7ee2912bhbde4b3475147628d@mail.gmail.com> <7aa29e790911242218l580b90eem3ec50676dfbc5536@mail.gmail.com> Message-ID: <4C8DCE59-7E49-4147-B930-227945D3D243@jerris.com> "you should use execute_complete events to tell when a command you tried to execute has finished and not poll the channel for a variable to be set because FreeSWITCH is an asynchronous application in the mode you are describing and you can never be sure of the timing." You are STILL polling for the variable. If you want help, perhaps you should at least attempt what is being suggested? Mike On Nov 25, 2009, at 1:18 AM, Thangappan.M wrote: > The example script is there in the following link > http://pastebin.com/f332f2fda > > In the previous post I have attached it. But it was not shown. > > 2009/11/25 Thangappan.M > FreeSWITCH version: freeswitch 1.0.4 > I am using ESL library > I attached the example Perl script which does the same steps that I posted already. ( Sample.pl) > I supplied the log , Here I attached the output of the ESL log. (Output.txt) > > Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. > But in the output I got only 2,4,5,4 ( DTMF 1 is missed) > > Output of Perl code could be like > > Wait for response time out > EVENT [COMMAND] > Wait for response time out > EVENT [DTMF] > DTMF digit 2 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 5 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > Buffer: 2454 > BYE > > Why the first digit(1) is missed here? > In ESL log there is no digit called 1 why? > Why the COMMAND event is received instead of DTMF? > How can I get all DTMF digits? > > > > > > > > > > > > > > > > > On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M wrote: > The reason for waiting only for DTMF event is to handle the time outs in the IVR concept like response and inter digit time out. Using our own logic we 10 voice files in each play back if the voice files are more than 10. Now it works fine. > > Now the new problem has been raised. The problem is we are filtering only for DTMF events but we are getting COMMAND event . Because of this the DTMF digits are missing at the time . I am not able to proceed further. We are in the critical situation. > > Why this command event is occurring? > How can I restrict this? > What are the information it has? > How can I get all the information in it ? ( If command event has info) > > Help me............ > > > On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: > I am waiting only for DTMF events. That's why I am setting freeswitch variable for knowing whether the playback has done. > > My question is "why this freeswitch variable is not setting properly when I play back more than 10 files using playback_delimiter option?". > > When I play back lesser than ten voice files the variable has been set properly. What could be the reason? > > > > ---------- Forwarded message ---------- > From: Thangappan.M > Date: Sat, Nov 21, 2009 at 2:52 PM > Subject: Problem while playing more than 10 voice files using playback > To: freeswitch-users > > > Dear all, > > I am in the process of implementing IVR using event outbound socket (async mode). > I have implemented using Perl language. > > I did the following steps: > => Set the playback_delimiter variable > => Set the playback_sleep_val variable > => Set the event lock as true > => Set the freeswitch ( my own) variable as zero > => Wait in the loop until the variable is been set as zero > => Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) > => Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played > successfully). > => Wait in the loop until the variable is been set as one. > => Set the Event lock as false > > => Trying to get the DTMF digits ( Have a assurance that all the voice files are played). > > The problem is, > > The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. > > But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. > > DID I MAKE ANY MISTAKE IN THE ABOVE STEPS? > > NOTE: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. > > Please help me................? > Thanks in advance. > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > > > > -- > Regards, > Thangappan.M > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/4aad26d9/attachment-0002.html From ovvenkatesan at gmail.com Tue Nov 24 22:36:20 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Wed, 25 Nov 2009 12:06:20 +0530 Subject: [Freeswitch-users] How to connect SIP phone to freeswitch Message-ID: <47d63d920911242236m2c4720a8g7c900fe5f02c05aa@mail.gmail.com> Hi . Could you please tell me, How to connect sip phone (which one is more friendly with freeswitch) to freeswitch. How I can check whether connection is properly established or not? -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/94fb5dc2/attachment-0002.html From mike at jerris.com Tue Nov 24 22:49:43 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 01:49:43 -0500 Subject: [Freeswitch-users] How to connect SIP phone to freeswitch In-Reply-To: <47d63d920911242236m2c4720a8g7c900fe5f02c05aa@mail.gmail.com> References: <47d63d920911242236m2c4720a8g7c900fe5f02c05aa@mail.gmail.com> Message-ID: <7465AD63-4C50-4D9C-993B-81B6621F98F4@jerris.com> http://wiki.freeswitch.org/wiki/Getting_Started_Guide http://wiki.freeswitch.org/wiki/Interop_List On Nov 25, 2009, at 1:36 AM, ovvenkat wrote: > Hi . > > Could you please tell me, How to connect sip phone (which one is more friendly with freeswitch) to freeswitch. How I can check whether connection is properly established or not? > From mcampbellsmith at gmail.com Tue Nov 24 23:46:11 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 25 Nov 2009 18:46:11 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> Message-ID: <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> Hi there Itamar, Does the SPA3102 support TLS or only SRTP? And what about Brians comments that 'It uses a sick twisted method of doing SRTP'. Do you have it working using SRTP together with FS? What about TLS? Otherwise are there any other ATA's that support TLS & SRTP? On Sun, Nov 22, 2009 at 8:41 PM, Itamar Reis Peixoto wrote: > it's support SRTP > > > On Sun, Nov 22, 2009 at 7:21 AM, Mark Campbell-Smith > wrote: >> Do LInksys devices support TLS and SRTP that FS supports? ?3102 at >> least doesn't according to this post > > > > > > -- > ------------ > > Itamar Reis Peixoto > > e-mail/msn/google talk/sip: itamar at ispbrasil.com.br > skype: itamarjp > icq: 81053601 > +55 11 4063 5033 > +55 34 3221 8599 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Wed Nov 25 00:14:53 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 25 Nov 2009 19:14:53 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> Message-ID: <20091125081453.GA28340@jdc.jasonjgw.net> Mark Campbell-Smith wrote: > Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... From ovvenkatesan at gmail.com Wed Nov 25 00:29:28 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Wed, 25 Nov 2009 13:59:28 +0530 Subject: [Freeswitch-users] How Register soft sip phones to FreeSWITCH with extension number. Message-ID: <47d63d920911250029v61538c37veefc1bd44e1bd072@mail.gmail.com> Hi to All, Any one please tell me , How to configure soft sip phone to freeswitch with extension number. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/dc8631a4/attachment-0002.html From mcampbellsmith at gmail.com Wed Nov 25 00:34:29 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 25 Nov 2009 19:34:29 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <20091125081453.GA28340@jdc.jasonjgw.net> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> Message-ID: <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> The only ATA mentioned on the WIKI that supports TLS/SRTP is the Grandstream HandyTone 503. But, again according to the wiki, that doesn't seem to behave to well with TLS ... On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: > Mark Campbell-Smith wrote: >> Does the SPA3102 support TLS or only SRTP? > > I don't know, but supporting only SRTP would be ridiculous, since the keys > would then be transmitted in the clear and therefore amenable to interception. > SRTP requires the SIP channel to be encrypted by TLS in order to be secure. > ZRTP, on the other hand, doesn't have this limitation: it works entirely in > RTP. > > I would be rather surprised were a hardware manufacturer to implement SRTP > without TLS for the SIP traffic. On the other hand, we've seen often in this > forum that some manufacturers are really clueless... > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mcampbellsmith at gmail.com Wed Nov 25 00:36:41 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 25 Nov 2009 19:36:41 +1100 Subject: [Freeswitch-users] How Register soft sip phones to FreeSWITCH with extension number. In-Reply-To: <47d63d920911250029v61538c37veefc1bd44e1bd072@mail.gmail.com> References: <47d63d920911250029v61538c37veefc1bd44e1bd072@mail.gmail.com> Message-ID: <33c87fa30911250036k4c0820d0pd26ef96d7971b024@mail.gmail.com> Didn't Michael already answer this? Best read the FS wiki and the softphone user guide for help with this. http://wiki.freeswitch.org/wiki/Getting_Started_Guide http://wiki.freeswitch.org/wiki/Interop_List On Wed, Nov 25, 2009 at 7:29 PM, ovvenkat wrote: > Hi to All, > > Any one please tell me , How to configure soft sip phone to freeswitch with > extension number. > > -- > > If you have come to help me, you are wasting your time. > If you have come to because your liberation is bound up in mine, we can work > together. > > > Regards > Venkatesan OV. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From stevendt at primrosebank.net Wed Nov 25 02:27:51 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Wed, 25 Nov 2009 10:27:51 -0000 Subject: [Freeswitch-users] Call Transfer Help Please References: <76F823D4525E409DA494ECD5BDDD3FF0@bp1.ad.bp.com><5A7C3038838142B5A07F181C193754F6@bp1.ad.bp.com> <1259121268571-4062810.post@n2.nabble.com> Message-ID: <19A54F6B1B3F419E8EB581FEA7F44725@bp1.ad.bp.com> Jeff, thanks very much for picking this up. You quickly spotted my mistake - I had the bind_meta_data call in the local extensions but not added it to the group extension (100). Appreciate you taking the time to have a look and point out my silly mistake - all working now, regards Dave ----- Original Message ----- From: "Jeff Lenk" To: Sent: Wednesday, November 25, 2009 3:54 AM Subject: Re: [Freeswitch-users] Call Transfer Help Please > > I do not see the meta app getting added in your log > -> > Dialplan: sofia/internal/1000 at 192.168.1.50 Action bind_meta_app(* > > Without this no meta actions will occur > > > > Dave Stevenson wrote: >> >> Hi again folks, >> >> I have posted a dump into the Pastebin (11276), could someone have a look >> and perhaps suggest where the problem might be please ? >> >> I'm sure you'll be able to work it out, but the log is for a call where >> :- >> >> incoming on PSTN Line (ext 1000) >> Group exts, 111, 1001, 1001 >> Answered on 111 and requested transfer to 1001 with no success >> >> regards >> Dave >> >> >> ----- Original Message ----- >> From: Dave Stevenson >> To: freeswitch-users at lists.freeswitch.org >> Sent: Tuesday, November 24, 2009 10:36 PM >> Subject: Re: [Freeswitch-users] Call Transfer Help Please >> >> >> Hi Mike, >> >> thanks for the reply. I am using the pre-compiled Windows binary - is >> there a 1.0.5 pre-release of that yet ? >> >> FreeSwitch reports its version as 1.0.4 (14460) but this is not >> correct, >> I was sure that I had previously loaded a later SVN Version, but just did >> it again, unless I'm not doing it right, the version number does not seem >> to be getting updated. The current build in the precompiled binaries area >> is reported to be 15604 and I've downloaded and installed that - although >> when the installer runs it tells me that it is version 15376. Either way, >> the "Version" command in FreeSwitch reports 1.0.4 (14460). >> >> The Transfer still does not work for me from the extension which >> answers >> the call. >> >> Sorry if my earlier questions were unclear ... >> "What are the correct LISTEN_TO and RESPOND_ON entries in >> dialplan.xml >> ?" >> What is the correct "transfer" data string in features.xml ? >> I don't understand this question(s) >> >> I was looking for clarification of the second two arguments in the >> bind_meta_app data call, i.e, that the "b" and "s" were the correct >> values >> and also that the "is transfer" "transfer" data argument was "-bleg" >> >> That is, that the arguments in the default dialplan are correct for >> this >> scenario - which they appear to be based on your previous reply to my >> query. >> >> So, is there anything else that I can check to see why this is not >> working ? >> >> >> regards >> Dave >> >> >> >> ----- Original Message ----- >> From: Michael Jerris >> To: freeswitch-users at lists.freeswitch.org >> Sent: Tuesday, November 24, 2009 8:19 PM >> Subject: Re: [Freeswitch-users] Call Transfer Help Please >> >> >> >> >> On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: >> >> >> Hi, >> >> I'm trying to setup call transfer for a phone without a transfer >> button. I was on IRC last night and got some pointers to how this is >> setup >> in dialplan.xml and features.xml and what "bind meta app" does. >> >> Once it became clear how the transfer is initiated and that the >> transfer, in the default config, can only be initiated by the "b" leg of >> the call, I was able to make this work as configured in the defaults, >> i.e, >> to initiate a transfer (for an internal call) from the dialled extension >> to a new extension. >> >> Now the problem . . . >> >> I have an incoming PSTN line that rings a group of extensions, what >> I want to be able to do is to give whoever answers the PSTN call ability >> to transfer the call on to another extension. >> >> There is an ATA (Linksys SPA3101) set up on the PSTN line with a >> FreeSwitch extension of 1000, it rings the extension phones in the group. >> >> I'd hoped that the default transfer setup would handle this without >> modification - the incoming call on extension 1000 would be the "a" leg, >> the answering extension would be the "b" leg and a transfer from "b" >> would >> work as per the default config. This does not work for me though. >> >> I'm struggling a bit with the "bind meta app" options and can't >> seem >> to make it do what I want. >> >> Could someone please confirm that what I'm trying to do is feasible >> and perhaps suggest the right parameters to use in dialplan.xml and >> features.xml please ? >> >> Relevant section in the "is_transfer" section in features.xml >> >> >> And in default.xml from >> to >> >> >> I've tried posting a call log to the Pastebin (11252/3) but there >> was an error - it looks like the dump was too big. Not sure what the >> maximum size on pastebin dumps is ? >> >> >> My understanding (or lack of) of "a" and "b" are in the scenario >> described is not helping ... >> >> Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? >> >> >> Yes, the calling leg >> >> >> Is the answering extension the "b" leg ? >> >> >> Yes >> >> >> What are the correct LISTEN_TO and RESPOND_ON entries in >> dialplan.xml ? >> >> >> I don't understand this question >> >> >> What is the correct "transfer" data string in features.xml ? >> >> >> >> ditto >> >> >> Or am I totally on the wrong track here ? >> >> >> >> You should just need to make sure that the bind meta is called in >> this >> scenario so the b leg is able to do it, thats it. >> >> >> If it is possible to do what I want, and changes are required to >> the >> dialplan.xml and/or features.xml files, is it possible to have different >> logic in there such that the actions are different whether it is the "a" >> leg or "b" leg that's requesting the transfer ? >> >> regards >> Dave >> >> FreeSwitch Version 1.0.4 (14460) >> >> >> also, try the latest 1.0.5. pre release or svn trunk to confirm this >> is not an issue that has already been fixed. >> >> >> Mike >> >> >> >> >> ---------------------------------------------------------------------------- >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> ------------------------------------------------------------------------------ >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Call-Transfer-Help-Please-tp4056930p4062810.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mattdfong at gmail.com Wed Nov 25 06:53:55 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 25 Nov 2009 06:53:55 -0800 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> Message-ID: <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> I tried removing the codec file extension from uuid_record and session_record but I'm still unable to record a file in native format for a bridged call. record WORKS!, but uuid_record and session_record do not want to record in native format. do uuid_record and session_record work with native format? or is it not going to be possible to record a bridged call in native format?...maybe because there are two different channels with a bridged call? If it isn't going to be possible, what's the best format to record bridged calls in that conserves the most processing power? .wav? Thanks. --matt DEBUG logs from console: http://pastebin.freeswitch.org/11283 Lua script: api = freeswitch.API(); --record = api:execute("sched_api", '+1 none uuid_record '..session:getVariable("uuid")..' start /tmp/my_recording'); --session:execute("record", "/tmp/my_recording"); session:execute("record_session", "/tmp/my_recording"); session:execute("playback", "somefile.wav"); On Mon, Nov 23, 2009 at 6:42 AM, Brian West wrote: > If you're doing native file you DO NOT put an extension on the file > name. > > /b > > On Nov 22, 2009, at 5:54 PM, Seven Du wrote: > > > did you try without any .wav or .PCMU? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/48bc296e/attachment-0002.html From kokoska.rokoska at post.cz Wed Nov 25 07:11:05 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 25 Nov 2009 16:11:05 +0100 Subject: [Freeswitch-users] how to enable short recordings Message-ID: <4B0D4909.7030009@post.cz> Hello all, is there a way how to enable very short recordings (1-3 seconds) in FreeSWITCH other than editing source code and recompiling? Thanks for your time! Best regards, kokoska.rokoska From brian at freeswitch.org Wed Nov 25 07:18:15 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 09:18:15 -0600 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> Message-ID: <133568C3-6EE1-40E2-8C0F-2CB174C2D94D@freeswitch.org> These two options attach media bugs on to the session. Which doesn't work with native files as far as I know. /b On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote: > record WORKS!, but uuid_record and session_record do not want to > record in native format. do uuid_record and session_record work with > native format? or is it not going to be possible to record a bridged > call in native format?...maybe because there are two different > channels with a bridged call? From brian at freeswitch.org Wed Nov 25 07:18:42 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 09:18:42 -0600 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <4B0D4909.7030009@post.cz> References: <4B0D4909.7030009@post.cz> Message-ID: <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> Is this standard recording? or voicemail? /b On Nov 25, 2009, at 9:11 AM, kokoska rokoska wrote: > Hello all, > > is there a way how to enable very short recordings (1-3 seconds) in > FreeSWITCH other than editing source code and recompiling? > > Thanks for your time! > > Best regards, > > kokoska.rokoska From jeff at jefflenk.com Wed Nov 25 07:32:17 2009 From: jeff at jefflenk.com (Jeff Lenk ) Date: Wed, 25 Nov 2009 15:32:17 +0000 Subject: [Freeswitch-users] how to enable short recordings Message-ID: Is this for vm? If so set min-record-len on the profile -----Original Message----- From: kokoska rokoska Sent: 11/25/2009 3:11:05 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] how to enable short recordings Hello all, is there a way how to enable very short recordings (1-3 seconds) in FreeSWITCH other than editing source code and recompiling? Thanks for your time! Best regards, kokoska.rokoska _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/1f1b152d/attachment-0002.html From imthiyazg at gmail.com Wed Nov 25 07:42:01 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Wed, 25 Nov 2009 21:12:01 +0530 Subject: [Freeswitch-users] passive recording Message-ID: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> hi is it possibe to enable passive recording in sangoma tdm interface in feeswich. pls advice Best Regards G.Imthiyaz Ahmed From mattdfong at gmail.com Wed Nov 25 07:44:41 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 25 Nov 2009 07:44:41 -0800 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <133568C3-6EE1-40E2-8C0F-2CB174C2D94D@freeswitch.org> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> <133568C3-6EE1-40E2-8C0F-2CB174C2D94D@freeswitch.org> Message-ID: <4256bf830911250744i3453f961h1e2c35f65222cab8@mail.gmail.com> so is using session_record with .wav my best option for recording bridged calls? --matt On Wed, Nov 25, 2009 at 7:18 AM, Brian West wrote: > These two options attach media bugs on to the session. Which doesn't > work with native files as far as I know. > > /b > > On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote: > > > record WORKS!, but uuid_record and session_record do not want to > > record in native format. do uuid_record and session_record work with > > native format? or is it not going to be possible to record a bridged > > call in native format?...maybe because there are two different > > channels with a bridged call? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/e2bde16f/attachment-0002.html From kokoska.rokoska at post.cz Wed Nov 25 07:46:37 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 25 Nov 2009 16:46:37 +0100 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> Message-ID: <4B0D515D.60805@post.cz> Thank you very much, Brian, for your interest! It is standard recording: Best regards, kokoska.rokoska Brian West napsal(a): > Is this standard recording? or voicemail? > > /b > > On Nov 25, 2009, at 9:11 AM, kokoska rokoska wrote: > >> Hello all, >> >> is there a way how to enable very short recordings (1-3 seconds) in >> FreeSWITCH other than editing source code and recompiling? >> >> Thanks for your time! >> >> Best regards, >> >> kokoska.rokoska > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Nov 25 07:51:56 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 09:51:56 -0600 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <4B0D515D.60805@post.cz> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> Message-ID: Really you want to keep 1-3 second files around? /b On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > Thank you very much, Brian, for your interest! > > It is standard recording: > > > > Best regards, > > kokoska.rokoska From rupa at rupa.com Wed Nov 25 07:56:14 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 25 Nov 2009 09:56:14 -0600 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <4256bf830911250744i3453f961h1e2c35f65222cab8@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> <133568C3-6EE1-40E2-8C0F-2CB174C2D94D@freeswitch.org> <4256bf830911250744i3453f961h1e2c35f65222cab8@mail.gmail.com> Message-ID: Yes On Wed, Nov 25, 2009 at 9:44 AM, Matthew Fong wrote: > so is using session_record with .wav my best option for recording bridged > calls? > > --matt > > > On Wed, Nov 25, 2009 at 7:18 AM, Brian West wrote: > >> These two options attach media bugs on to the session. Which doesn't >> work with native files as far as I know. >> >> /b >> >> On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote: >> >> > record WORKS!, but uuid_record and session_record do not want to >> > record in native format. do uuid_record and session_record work with >> > native format? or is it not going to be possible to record a bridged >> > call in native format?...maybe because there are two different >> > channels with a bridged call? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/a9948b61/attachment-0002.html From kokoska.rokoska at post.cz Wed Nov 25 08:03:45 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 25 Nov 2009 17:03:45 +0100 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> Message-ID: <4B0D5561.4020009@post.cz> Yes, Brian, I need them :-) They don't contain speech - instead, they contain few "computer generated" tones and I should store them in max quality for later proccessing (i.e. analysis)... Best regards, kokoska.rokoska Brian West napsal(a): > Really you want to keep 1-3 second files around? > > /b > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > >> Thank you very much, Brian, for your interest! >> >> It is standard recording: >> >> >> >> Best regards, >> >> kokoska.rokoska > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From juanbackson at gmail.com Wed Nov 25 08:10:30 2009 From: juanbackson at gmail.com (Juan Backson) Date: Thu, 26 Nov 2009 00:10:30 +0800 Subject: [Freeswitch-users] modify SDP for 200 OK Message-ID: <27c25bc40911250810x783fb1cbg49f50e624353bd51@mail.gmail.com> Hi, If I am using proxy_media=true, bypass_media=false, is there anyway of modifying o= and c= so that it won't show the IP of the far-end B leg? I am using fs as b2b2a and I want to hide the far-end ip as much as possible. I got to hide the IP for invite by modifying the sdp within C code, but I don't know how to do that for 200 OK. Any idea? Thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/76a73c27/attachment-0002.html From juliano.duque at terra.com.br Wed Nov 25 05:18:07 2009 From: juliano.duque at terra.com.br (Juliano - Terra) Date: Wed, 25 Nov 2009 11:18:07 -0200 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B0C6499.4060504@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> Message-ID: <4B0D2E8F.1030200@terra.com.br> Peter, I had a similar problem, the way I found to make it work was setting the mailbox ID in the phone to match the FS domain/hostname. For instance using a Linksys SPA962, I set the "Voice Mail Server" in the extension tab to extension at domain using FS hostname as the domain. Regards, Juliano Peter P GMX escreveu: > Anthony, thanks for the hint, > > I receive events like the following > RECV EVENT > Event-Name: MESSAGE_WAITING > Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 > FreeSWITCH-Hostname: sip11.mydomain.com > FreeSWITCH-IPv4: 192.168.178.200 > FreeSWITCH-IPv6: ::1 > Event-Date-Local: 2009-11-24 23:33:13 > Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT > Event-Date-Timestamp: 1259101993918617 > Event-Calling-File: mod_voicemail.c > Event-Calling-Function: update_mwi > Event-Calling-Line-Number: 1738 > MWI-Messages-Waiting: yes > MWI-Message-Account: 200 at sip1.mydomain.com > MWI-Voice-Message: 4/1 (0/0) > > I think the problem may be the Freeswitch cluster we are working with. > All phones register with realm (e.g. 200 at sip1.mydomain.com). The FS > hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. > With xml_curl we ensure that for both domain names a directory entry is > passed back. That way it works nicely with registering phones, receiving > voicemails, recording voicemails etc. but not for MWI. For recording and > querying voicemails we use the realm instead of the domain name and that > way it works. > > When a voicemail has finished recording - and at the time the above > message occurs - I see 2 directory xml_curl requests with > Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id > One I expect is for retrieving the MWI data and the other one for > sending the VM email (which is sucessfully sent). > > Any hint how we can workaround this? Or is there a parameter to tell > mod_voicemail that is should use the realm instead of the local hostname > for sending MWI? > > Best regards > Peter > > Anthony Minessale schrieb: > >> connect to FS with fs_cli >> >> Issue the command: >> >> /events MESSAGE_QUERY MESSAGE_WAITING >> >> then leave some voice mails >> >> probably you have a mis-configuration where the user/domain/profile >> cannot be resolved to the correct >> sofia profile to send the notify >> >> The event starts out as a freeswitch event and is translated into the >> notify by mod_sofia but only if it can >> match the event to a real sip user >> >> >> >> >> On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX > > wrote: >> >> Hello, >> >> I have a similar problem with Freeswitch behind OpenSIPS as a load >> balancer: >> When registering, Freeeswitch does not send a MWI NOTIFY message for a >> Phone which has voicemails. Even after recording a new voicemail there >> is no NOTIFY message sent. And there are no error messages on the >> console. >> >> I have explicitely set >> in the internal >> profile. >> >> When a phone is set up I get the following >> Snom Phone REGISTER => OpenSIPS=> Freeswitch >> Freeswitch OK => OpenSIPS=>Snom Phone >> >> Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch >> Freeswitch 202 Accepted => OpenSIPS=>Snom Phone >> >> Snom Phone PUBLISH => OpenSIPS=> Freeswitch >> Freeswitch 200 OK => OpenSIPS=>Snom Phone >> So presence generally seems to work. >> >> But ngrepping the Network traffic there's no MWI NOTIFY message coming >> from Freeswitch to any phone. >> FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before >> should be already there. >> >> Any idea how to force the NOTIFY messages? >> >> >> Best regards >> Peter >> >> Here's the debug Level9 output for nta and nua when a phone with VMs >> registers, seems like there is no error in it: >> >> freeswitch at sip11.mydomain.com >> > nta: received REGISTER >> sip:sip1.mydomain.com SIP/2.0 (CSeq 7) >> nta: REGISTER (7) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x7fd5d409c8f0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x7fd5d409c8f0): sent signal r_respond >> nua: nua_handle_destroy: entering >> nua(0x7fd5d409c8f0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized >> nua: nua_stack_set_params: entering >> nta: sent 401 Unauthorized for REGISTER (7) >> nta: timer set to 32000 ms >> nua(0x7fd5d409c8f0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 6) >> nta: REGISTER (6) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x905a80): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x905a80): sent signal r_respond >> nua: nua_handle_destroy: entering >> nua(0x905a80): recv signal r_respond 401 Unauthorized >> nua(0x905a80): sent signal r_destroy >> nua: nua_stack_set_params: entering >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 401 Unauthorized for REGISTER (6) >> nua(0x905a80): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received PUBLISH sip:100 at sip1.mydomain.com >> SIP/2.0 (CSeq 3) >> nta: PUBLISH (3) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x905f10): event i_publish 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x905f10): sent signal r_respond >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nua(0x905f10): recv signal r_respond 200 OK >> nua: nua_stack_set_params: entering >> nua(0x905f10): sent signal r_destroy >> nta: sent 200 OK for PUBLISH (3) >> nua(0x905f10): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received SUBSCRIBE sip:mod_sofia at 192.168.178.200:5062 >> SIP/2.0 (CSeq 2) >> nta: canonizing sip:mod_sofia at 192.168.178.200:5062 >> with contact >> nta: SUBSCRIBE (2) going to existing leg >> nua: nua_stack_process_request: entering >> nta: sent 200 OK for SUBSCRIBE (2) >> nua(0x905560): event i_subscribe 200 OK >> nua: nua_application_event: entering >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 8) >> nta: REGISTER (8) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x7fd5dc073ba0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x7fd5dc073ba0): sent signal r_respond >> nua(0x7fd5dc073ba0): recv signal r_respond 200 OK >> nua: nua_stack_set_params: entering >> nua: nua_handle_destroy: entering >> nua(0x7fd5dc073ba0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 200 OK for REGISTER (8) >> nua(0x7fd5dc073ba0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 7) >> nta: REGISTER (7) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x8fc3d0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x8fc3d0): sent signal r_respond >> nua(0x8fc3d0): recv signal r_respond 200 OK >> nua: nua_handle_destroy: entering >> nua: nua_stack_set_params: entering >> nua(0x8fc3d0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 200 OK for REGISTER (7) >> nua(0x8fc3d0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received SUBSCRIBE sip:100 at sip1.mydomain.com >> ;user=phone SIP/2.0 >> (CSeq 1) >> nta: SUBSCRIBE (1) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nta_leg_tcreate(0x7fd5dc03add0) >> nua(0x7fd5dc078b70): adding notify usage with event message-summary >> nua(0x7fd5dc078b70): event i_subscribe 100 Trying >> nua: nua_application_event: entering >> nua(): refresh notify after 3600 seconds (in [3600..3600]) >> nua: nua_respond: entering >> nua(0x7fd5dc078b70): sent signal r_respond >> nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted >> nua: nua_stack_set_params: entering >> nta: sent 202 Accepted for SUBSCRIBE (1) >> >> >> >> >> >> mayamatakeshi schrieb: >> > >> > On 9/12/09, *mayamatakeshi* > >> > > >> wrote: >> > >> > >> > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris >> >> > >> wrote: >> > >> > Following up, did a bug get created for this issue? >> > >> > >> > Hello, >> > yes. >> > http://jira.freeswitch.org/browse/MODSOFIA-26 >> > >> > >> > Just to simplify things in case someone searches the list: >> > Issue was solved on rev 14851. >> > Thank you all. >> > >> ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From brian at freeswitch.org Wed Nov 25 08:20:29 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 10:20:29 -0600 Subject: [Freeswitch-users] modify SDP for 200 OK In-Reply-To: <27c25bc40911250810x783fb1cbg49f50e624353bd51@mail.gmail.com> References: <27c25bc40911250810x783fb1cbg49f50e624353bd51@mail.gmail.com> Message-ID: You know FreeSWITCH will proxy media already if you turn off proxy_media and disable transcoding you'll get the same results and the IP's will be correct. Proxy media is for one purpose... T.38, it gains you NOTHING otherwise. /b On Nov 25, 2009, at 10:10 AM, Juan Backson wrote: > Hi, > > If I am using proxy_media=true, bypass_media=false, is there anyway > of modifying o= and c= so that it won't show the IP of the far-end B > leg? > > I am using fs as b2b2a and I want to hide the far-end ip as much as > possible. > > I got to hide the IP for invite by modifying the sdp within C code, > but I don't know how to do that for 200 OK. Any idea? > > Thanks, > jb From srinivas.ksvreddy at gmail.com Wed Nov 25 08:53:34 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Wed, 25 Nov 2009 22:23:34 +0530 Subject: [Freeswitch-users] Bypass_media and re_invite Message-ID: Hi All, goodmorning to all, i have a scenario, two pjsua clients are connected with Freeswitch and they are in call and bypass_media=true. i close the Freeswitch server, still they are in call, again i started the Freeswitch, and registerd these two endpoints, now how can i end the call(estabilished by the first Freeswitch)? if i call re_invite will it estabilish the call between two endpoints? any idea? Thanks Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/71317773/attachment-0002.html From imthiyazg at gmail.com Wed Nov 25 09:00:43 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Wed, 25 Nov 2009 22:30:43 +0530 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> Message-ID: <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> hi is it possibe to enable passive recording in sangoma tdm interface in feeswich. pls advice Best Regards G.Imthiyaz Ahmed -- Best Regards G.Imthiyaz Ahmed PeopleTech systems (P) ltd http://peopletech.co.in From anthony.minessale at gmail.com Wed Nov 25 09:07:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 11:07:04 -0600 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <4B0D5561.4020009@post.cz> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> <4B0D5561.4020009@post.cz> Message-ID: <191c3a030911250907x38aca94cs9128629fc2a1ba7c@mail.gmail.com> use the variable RECORD_MIN_SEC This was added in revision 15271 so if you are below that I recommend updating to latest trunk. On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska wrote: > Yes, Brian, I need them :-) > > They don't contain speech - instead, they contain few "computer > generated" tones and I should store them in max quality for later > proccessing (i.e. analysis)... > > Best regards, > > kokoska.rokoska > > > Brian West napsal(a): > > Really you want to keep 1-3 second files around? > > > > /b > > > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > > > >> Thank you very much, Brian, for your interest! > >> > >> It is standard recording: > >> > >> > >> > >> Best regards, > >> > >> kokoska.rokoska > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/d72acaf6/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 25 09:13:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 11:13:39 -0600 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> Message-ID: <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> What do you mean by passive encoding? On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed wrote: > hi > is it possibe to enable passive recording in sangoma tdm interface > in feeswich. pls advice > Best Regards > G.Imthiyaz Ahmed > > > > -- > Best Regards > G.Imthiyaz Ahmed > PeopleTech systems (P) ltd > http://peopletech.co.in > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/1725495a/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 25 09:13:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 11:13:47 -0600 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> Message-ID: <191c3a030911250913i41ec7571t17396b9af247eb2f@mail.gmail.com> What do you mean by passive encoding? On Wed, Nov 25, 2009 at 11:13 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > What do you mean by passive encoding? > > > On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed wrote: > >> hi >> is it possibe to enable passive recording in sangoma tdm interface >> in feeswich. pls advice >> Best Regards >> G.Imthiyaz Ahmed >> >> >> >> -- >> Best Regards >> G.Imthiyaz Ahmed >> PeopleTech systems (P) ltd >> http://peopletech.co.in >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/39e55637/attachment-0002.html From mshepet at gmail.com Mon Nov 23 09:09:33 2009 From: mshepet at gmail.com (Michael Shepet) Date: Mon, 23 Nov 2009 12:09:33 -0500 Subject: [Freeswitch-users] Server give-away Message-ID: We at Swifcore Technologies, a telephony and server management team, would like your help in reviewing our latest product. We have created a hosting platform around the FreeSWITCH engine (for obvious reasons of stability and extensibility) and would like your feedback so we continue to improve our service. To facilitate this, we are giving away ten (10) hosting packages (pre-configured with latest FreeSWITCH compiled trunk) along with a web diagnostic dashboard for a free 60 day trial, no strings attached (or credit card required)! Each private server comes with CentOS 5.4 64-bit, 512MB RAM, its own public IP (no NAT problems), a shared test DID number, 100 minutes/month call credit, and ssh access. All you have to do is tell us why you love FreeSWITCH and something creative you have done or plan on doing with it. The 10 best responses will be awarded servers. If you are interested, email your response to giveaway at swifcore.com. You can also go to http://www.swifcore.com/products/myswitch for more product details. Thank you! Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091123/1b98a0b3/attachment-0002.html From kokoska.rokoska at post.cz Wed Nov 25 09:20:06 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 25 Nov 2009 18:20:06 +0100 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <191c3a030911250907x38aca94cs9128629fc2a1ba7c@mail.gmail.com> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> <4B0D5561.4020009@post.cz> <191c3a030911250907x38aca94cs9128629fc2a1ba7c@mail.gmail.com> Message-ID: <4B0D6746.8060907@post.cz> Thank you very much, Anthony, for your help! I'm nearly at current trunk (15653) and works great :-) Many thanks once more! Best regards, kokoska.rokoska Anthony Minessale napsal(a): > use the variable RECORD_MIN_SEC > > This was added in revision 15271 so if you are below that I recommend > updating to latest trunk. > > > On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska > > wrote: > > Yes, Brian, I need them :-) > > They don't contain speech - instead, they contain few "computer > generated" tones and I should store them in max quality for later > proccessing (i.e. analysis)... > > Best regards, > > kokoska.rokoska > > > Brian West napsal(a): > > Really you want to keep 1-3 second files around? > > > > /b > > > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > > > >> Thank you very much, Brian, for your interest! > >> > >> It is standard recording: > >> > >> > >> > >> Best regards, > >> > >> kokoska.rokoska > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From imthiyazg at gmail.com Wed Nov 25 09:29:03 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Wed, 25 Nov 2009 22:59:03 +0530 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> Message-ID: <8595daf70911250929w26eeb3aboae0f95042f35393b@mail.gmail.com> I mean to tap tx and rx of a PRI line using sangoma tap and record the call information and actual calls without distrubing the existing line . freeswitch will work in passive mode like trunk side call recorder. Thanks Imthiyaz On Wed, Nov 25, 2009 at 10:43 PM, Anthony Minessale wrote: > What do you mean by passive encoding? > > On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed > wrote: >> >> hi >> ?is it possibe to enable passive recording in sangoma tdm interface >> in feeswich. pls advice >> Best Regards >> G.Imthiyaz Ahmed >> >> >> >> -- >> Best Regards >> G.Imthiyaz Ahmed >> PeopleTech systems (P) ltd >> http://peopletech.co.in >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Best Regards G.Imthiyaz Ahmed PeopleTech systems (P) ltd http://peopletech.co.in From dujinfang at gmail.com Wed Nov 25 09:32:51 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 26 Nov 2009 01:32:51 +0800 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <23f91030911221554m2438e6a8x7a65f989964bc46f@mail.gmail.com> <1E945EE3-7361-45DC-BD72-19E1E07B8695@freeswitch.org> <4256bf830911250653p76eb66dds78c9a22c8c73acab@mail.gmail.com> <133568C3-6EE1-40E2-8C0F-2CB174C2D94D@freeswitch.org> <4256bf830911250744i3453f961h1e2c35f65222cab8@mail.gmail.com> Message-ID: <23f91030911250932m65d22333sd299702b881c1891@mail.gmail.com> http://code.google.com/p/mod-recpld/ It's out-dated. I originally wrote it to record raw G.729 codec on passthrough mode. It worked before and then we abandoned that since We felt G729 cannot deliver good sound particularly on a cross-continent network. The code is written when I don't know much about FS internals, perhaps it's easier to write some mod with indicate no-transcoding in switch_rtp.c. 2009/11/25 Rupa Schomaker > Yes > > > On Wed, Nov 25, 2009 at 9:44 AM, Matthew Fong wrote: > >> so is using session_record with .wav my best option for recording bridged >> calls? >> >> --matt >> >> >> On Wed, Nov 25, 2009 at 7:18 AM, Brian West wrote: >> >>> These two options attach media bugs on to the session. Which doesn't >>> work with native files as far as I know. >>> >>> /b >>> >>> On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote: >>> >>> > record WORKS!, but uuid_record and session_record do not want to >>> > record in native format. do uuid_record and session_record work with >>> > native format? or is it not going to be possible to record a bridged >>> > call in native format?...maybe because there are two different >>> > channels with a bridged call? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/29ce5598/attachment-0002.html From dujinfang at gmail.com Wed Nov 25 09:36:20 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 26 Nov 2009 01:36:20 +0800 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <4B0D6746.8060907@post.cz> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> <4B0D5561.4020009@post.cz> <191c3a030911250907x38aca94cs9128629fc2a1ba7c@mail.gmail.com> <4B0D6746.8060907@post.cz> Message-ID: <23f91030911250936q65dc7c25oe8c3746d6caed9e7@mail.gmail.com> And you may also would like to update the wiki as well if the var is not there. 2009/11/26 kokoska rokoska > Thank you very much, Anthony, for your help! > > I'm nearly at current trunk (15653) and > > works great :-) > > Many thanks once more! > > Best regards, > > kokoska.rokoska > > > Anthony Minessale napsal(a): > > use the variable RECORD_MIN_SEC > > > > This was added in revision 15271 so if you are below that I recommend > > updating to latest trunk. > > > > > > On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska > > > wrote: > > > > Yes, Brian, I need them :-) > > > > They don't contain speech - instead, they contain few "computer > > generated" tones and I should store them in max quality for later > > proccessing (i.e. analysis)... > > > > Best regards, > > > > kokoska.rokoska > > > > > > Brian West napsal(a): > > > Really you want to keep 1-3 second files around? > > > > > > /b > > > > > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > > > > > >> Thank you very much, Brian, for your interest! > > >> > > >> It is standard recording: > > >> > > >> > > >> > > >> Best regards, > > >> > > >> kokoska.rokoska > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/72720bd3/attachment-0002.html From mike at jerris.com Wed Nov 25 09:39:10 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:39:10 -0500 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B0C6499.4060504@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> Message-ID: <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> Try an alias on the sip profile. Mike On Nov 24, 2009, at 5:56 PM, Peter P GMX wrote: > Anthony, thanks for the hint, > > I receive events like the following > RECV EVENT > Event-Name: MESSAGE_WAITING > Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 > FreeSWITCH-Hostname: sip11.mydomain.com > FreeSWITCH-IPv4: 192.168.178.200 > FreeSWITCH-IPv6: ::1 > Event-Date-Local: 2009-11-24 23:33:13 > Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT > Event-Date-Timestamp: 1259101993918617 > Event-Calling-File: mod_voicemail.c > Event-Calling-Function: update_mwi > Event-Calling-Line-Number: 1738 > MWI-Messages-Waiting: yes > MWI-Message-Account: 200 at sip1.mydomain.com > MWI-Voice-Message: 4/1 (0/0) > > I think the problem may be the Freeswitch cluster we are working with. > All phones register with realm (e.g. 200 at sip1.mydomain.com). The FS > hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. > With xml_curl we ensure that for both domain names a directory entry is > passed back. That way it works nicely with registering phones, receiving > voicemails, recording voicemails etc. but not for MWI. For recording and > querying voicemails we use the realm instead of the domain name and that > way it works. > > When a voicemail has finished recording - and at the time the above > message occurs - I see 2 directory xml_curl requests with > Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id > One I expect is for retrieving the MWI data and the other one for > sending the VM email (which is sucessfully sent). > > Any hint how we can workaround this? Or is there a parameter to tell > mod_voicemail that is should use the realm instead of the local hostname > for sending MWI? > > Best regards > Peter > > Anthony Minessale schrieb: >> connect to FS with fs_cli >> >> Issue the command: >> >> /events MESSAGE_QUERY MESSAGE_WAITING >> >> then leave some voice mails >> >> probably you have a mis-configuration where the user/domain/profile >> cannot be resolved to the correct >> sofia profile to send the notify >> >> The event starts out as a freeswitch event and is translated into the >> notify by mod_sofia but only if it can >> match the event to a real sip user >> >> >> >> >> On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX > > wrote: >> >> Hello, >> >> I have a similar problem with Freeswitch behind OpenSIPS as a load >> balancer: >> When registering, Freeeswitch does not send a MWI NOTIFY message for a >> Phone which has voicemails. Even after recording a new voicemail there >> is no NOTIFY message sent. And there are no error messages on the >> console. >> >> I have explicitely set >> in the internal >> profile. >> >> When a phone is set up I get the following >> Snom Phone REGISTER => OpenSIPS=> Freeswitch >> Freeswitch OK => OpenSIPS=>Snom Phone >> >> Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch >> Freeswitch 202 Accepted => OpenSIPS=>Snom Phone >> >> Snom Phone PUBLISH => OpenSIPS=> Freeswitch >> Freeswitch 200 OK => OpenSIPS=>Snom Phone >> So presence generally seems to work. >> >> But ngrepping the Network traffic there's no MWI NOTIFY message coming >> from Freeswitch to any phone. >> FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before >> should be already there. >> >> Any idea how to force the NOTIFY messages? >> >> >> Best regards >> Peter >> >> Here's the debug Level9 output for nta and nua when a phone with VMs >> registers, seems like there is no error in it: >> >> freeswitch at sip11.mydomain.com >> > nta: received REGISTER >> sip:sip1.mydomain.com SIP/2.0 (CSeq 7) >> nta: REGISTER (7) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x7fd5d409c8f0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x7fd5d409c8f0): sent signal r_respond >> nua: nua_handle_destroy: entering >> nua(0x7fd5d409c8f0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized >> nua: nua_stack_set_params: entering >> nta: sent 401 Unauthorized for REGISTER (7) >> nta: timer set to 32000 ms >> nua(0x7fd5d409c8f0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 6) >> nta: REGISTER (6) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x905a80): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x905a80): sent signal r_respond >> nua: nua_handle_destroy: entering >> nua(0x905a80): recv signal r_respond 401 Unauthorized >> nua(0x905a80): sent signal r_destroy >> nua: nua_stack_set_params: entering >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 401 Unauthorized for REGISTER (6) >> nua(0x905a80): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received PUBLISH sip:100 at sip1.mydomain.com >> SIP/2.0 (CSeq 3) >> nta: PUBLISH (3) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x905f10): event i_publish 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x905f10): sent signal r_respond >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nua(0x905f10): recv signal r_respond 200 OK >> nua: nua_stack_set_params: entering >> nua(0x905f10): sent signal r_destroy >> nta: sent 200 OK for PUBLISH (3) >> nua(0x905f10): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received SUBSCRIBE sip:mod_sofia at 192.168.178.200:5062 >> SIP/2.0 (CSeq 2) >> nta: canonizing sip:mod_sofia at 192.168.178.200:5062 >> with contact >> nta: SUBSCRIBE (2) going to existing leg >> nua: nua_stack_process_request: entering >> nta: sent 200 OK for SUBSCRIBE (2) >> nua(0x905560): event i_subscribe 200 OK >> nua: nua_application_event: entering >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 8) >> nta: REGISTER (8) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x7fd5dc073ba0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x7fd5dc073ba0): sent signal r_respond >> nua(0x7fd5dc073ba0): recv signal r_respond 200 OK >> nua: nua_stack_set_params: entering >> nua: nua_handle_destroy: entering >> nua(0x7fd5dc073ba0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 200 OK for REGISTER (8) >> nua(0x7fd5dc073ba0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received REGISTER sip:sip1.mydomain.com >> SIP/2.0 (CSeq 7) >> nta: REGISTER (7) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nua(0x8fc3d0): event i_register 100 Trying >> nua: nua_application_event: entering >> nua: nua_respond: entering >> nua(0x8fc3d0): sent signal r_respond >> nua(0x8fc3d0): recv signal r_respond 200 OK >> nua: nua_handle_destroy: entering >> nua: nua_stack_set_params: entering >> nua(0x8fc3d0): sent signal r_destroy >> nua: nua_handle_magic: entering >> nua: nua_handle_destroy: entering >> nta: sent 200 OK for REGISTER (7) >> nua(0x8fc3d0): recv signal r_destroy >> nta_leg_destroy((nil)) >> nta: received SUBSCRIBE sip:100 at sip1.mydomain.com >> ;user=phone SIP/2.0 >> (CSeq 1) >> nta: SUBSCRIBE (1) going to a default leg >> nua: nua_stack_process_request: entering >> nua: nh_create: entering >> nua: nh_create_handle: entering >> nua: nua_stack_set_params: entering >> nta_leg_tcreate(0x7fd5dc03add0) >> nua(0x7fd5dc078b70): adding notify usage with event message-summary >> nua(0x7fd5dc078b70): event i_subscribe 100 Trying >> nua: nua_application_event: entering >> nua(): refresh notify after 3600 seconds (in [3600..3600]) >> nua: nua_respond: entering >> nua(0x7fd5dc078b70): sent signal r_respond >> nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted >> nua: nua_stack_set_params: entering >> nta: sent 202 Accepted for SUBSCRIBE (1) >> >> >> >> >> >> mayamatakeshi schrieb: >>> >>> On 9/12/09, *mayamatakeshi* > >>> > >> wrote: >>> >>> >>> On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris >> >>> >> wrote: >>> >>> Following up, did a bug get created for this issue? >>> >>> >>> Hello, >>> yes. >>> http://jira.freeswitch.org/browse/MODSOFIA-26 >>> >>> >>> Just to simplify things in case someone searches the list: >>> Issue was solved on rev 14851. >>> Thank you all. >>> >> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Wed Nov 25 09:44:05 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:44:05 -0500 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: FreeSWITCH will kill the calls when you shut it down, if you intentionally kill the network without shutting down FreeSWITCH the only thing you can do is enable session timers or rtp timers in the soft phones to kill the call when FreeSWITCH dies or when the call is over. Mike On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > Hi All, > > goodmorning to all, i have a scenario, two pjsua clients are connected with Freeswitch and they are in call and bypass_media=true. i close the Freeswitch server, still they are in call, again i started the Freeswitch, and registerd these two endpoints, now how can i end the call(estabilished by the first Freeswitch)? if i call re_invite will it estabilish the call between two endpoints? > any idea? From mike at jerris.com Wed Nov 25 09:44:46 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:44:46 -0500 Subject: [Freeswitch-users] mod_conference kick to abort invitations In-Reply-To: <000001ca6d56$66037c80$320a7580$@de> References: <000001ca6d56$66037c80$320a7580$@de> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B@jerris.com> Its a feature we don't have, patches welcome. Mike On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > Hi members, > I?m controlling freeswitch with the conference module via xmlrpc. > > Is it desired that the kick command can only kick users that are connected to the conference? > Is there no chance abort an invitation? > The kick command has no effect until the person I invited with the dial command is connected. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0002.html From mike at jerris.com Wed Nov 25 09:45:50 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:45:50 -0500 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: Message-ID: In trunk there is a sofia profile setting to allow dialplan processing of 302 responses. This won't get you back into your same javascript, but you can probably do something clever from there. Mike On Nov 24, 2009, at 5:04 PM, John Platts wrote: > > I have considered writing JavaScript code to bridge two calls together. However, I would like to perform custom handling of the 302 Moved Temporarily response. How do I handle the 302 Moved Temporarily response if I use JavaScript? > From brian at freeswitch.org Wed Nov 25 09:46:05 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 11:46:05 -0600 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> Yes an alias will be required for every domain you run on the profile so it can find it. /b On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > Try an alias on the sip profile. > > Mike From mike at jerris.com Wed Nov 25 09:47:37 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:47:37 -0500 Subject: [Freeswitch-users] remote_media_ip variable not set In-Reply-To: <27c25bc40911240756k7842c80kd75be2d3d93441b9@mail.gmail.com> References: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> <2F929FDB-0E1B-49E0-A1E7-F4F1E2D548AD@avgs.ca> <27c25bc40911240756k7842c80kd75be2d3d93441b9@mail.gmail.com> Message-ID: It's possible it does not. I just added some code to set it on auto-adjust so it might be there sometimes now. You might need to add some code in mod_sofia to add it other times. Maybe it makes sense to move that var setting down to switch_rtp.c. Patches for this would be welcome. Thanks Mike On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > Hi, > > In the case of proxy_media=true, does it gets set at all then? From mike at jerris.com Wed Nov 25 09:48:39 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 12:48:39 -0500 Subject: [Freeswitch-users] How to find whether the destination extension supports encryption In-Reply-To: References: Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40@jerris.com> You can send the call with secure enabled and if it supports it it will use it. Mike On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > Hello, > > We have a mix of phones that support RTP encryption and those that do not. I have to support both types in the meanwhile, and would like to have encryption enabled on the relevant leg, even if the other leg does not support it (why? one of our ATAs either must have it unencrypted or have it encrypted, but cannot have both). > > How do I find whether the destination supports encryption? I do not want to manage an additional table in the database... > From srinivas.ksvreddy at gmail.com Wed Nov 25 09:55:01 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Wed, 25 Nov 2009 23:25:01 +0530 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: HI, thanks for your reply, my requirement is i am doing failover stuff with freeswitch. i dont want cut the calls when freeswitch dies, when failover happens mean one freeswitch dies we are going to start the second freeswitch, i dont want close call intiated by the first freeswtich, they are communicating with meida(bypass media). when one endpoing try to end the call at that time i want to close the call for the other end also. srinivas On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris wrote: > FreeSWITCH will kill the calls when you shut it down, if you intentionally > kill the network without shutting down FreeSWITCH the only thing you can do > is enable session timers or rtp timers in the soft phones to kill the call > when FreeSWITCH dies or when the call is over. > > Mike > > On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > > > Hi All, > > > > goodmorning to all, i have a scenario, two pjsua clients are connected > with Freeswitch and they are in call and bypass_media=true. i close the > Freeswitch server, still they are in call, again i started the Freeswitch, > and registerd these two endpoints, now how can i end the call(estabilished > by the first Freeswitch)? if i call re_invite will it estabilish the call > between two endpoints? > > any idea? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0002.html From stevecrozz at gmail.com Wed Nov 25 10:01:14 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 25 Nov 2009 10:01:14 -0800 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: Message-ID: <11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com> Surprisingly, I've found no way to access the HTTP response status code using mod_spidermonkey_curl. I'd love to see this feature added or discussed if it already exists and I'm missing it. --Stephen On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris wrote: > In trunk there is a sofia profile setting to allow dialplan processing of > 302 responses. This won't get you back into your same javascript, but you > can probably do something clever from there. > > Mike > > On Nov 24, 2009, at 5:04 PM, John Platts wrote: > > > > > I have considered writing JavaScript code to bridge two calls together. > However, I would like to perform custom handling of the 302 Moved > Temporarily response. How do I handle the 302 Moved Temporarily response if > I use JavaScript? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0002.html From tculjaga at gmail.com Wed Nov 25 10:04:56 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 25 Nov 2009 19:04:56 +0100 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: Message-ID: <65d96fc80911251004l401d5efbl8df3a2ac920207b8@mail.gmail.com> this is how i do it from the dialplan: On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris wrote: > In trunk there is a sofia profile setting to allow dialplan processing of > 302 responses. This won't get you back into your same javascript, but you > can probably do something clever from there. > > Mike > > On Nov 24, 2009, at 5:04 PM, John Platts wrote: > > > > > I have considered writing JavaScript code to bridge two calls together. > However, I would like to perform custom handling of the 302 Moved > Temporarily response. How do I handle the 302 Moved Temporarily response if > I use JavaScript? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment-0002.html From kokoska.rokoska at post.cz Wed Nov 25 10:09:38 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 25 Nov 2009 19:09:38 +0100 Subject: [Freeswitch-users] how to enable short recordings In-Reply-To: <23f91030911250936q65dc7c25oe8c3746d6caed9e7@mail.gmail.com> References: <4B0D4909.7030009@post.cz> <3332BEF3-DB28-4909-BC6D-BEFBB373094C@freeswitch.org> <4B0D515D.60805@post.cz> <4B0D5561.4020009@post.cz> <191c3a030911250907x38aca94cs9128629fc2a1ba7c@mail.gmail.com> <4B0D6746.8060907@post.cz> <23f91030911250936q65dc7c25oe8c3746d6caed9e7@mail.gmail.com> Message-ID: <4B0D72E2.10607@post.cz> It was the first I want to do - update wiki :-) But someone was much faster (00:14, 30 October 2009 :-) http://wiki.freeswitch.org/wiki/Variable_record_min_sec Last time I looked for some hint about the recording (few months ago), this page (and even the variable) didn't exist... Best regards, kokoska.rokoska Seven Du napsal(a): > And you may also would like to update the wiki as well if the var is not > there. > > 2009/11/26 kokoska rokoska > > > Thank you very much, Anthony, for your help! > > I'm nearly at current trunk (15653) and > > works great :-) > > Many thanks once more! > > Best regards, > > kokoska.rokoska > > > Anthony Minessale napsal(a): > > use the variable RECORD_MIN_SEC > > > > This was added in revision 15271 so if you are below that I recommend > > updating to latest trunk. > > > > > > On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska > > > >> > wrote: > > > > Yes, Brian, I need them :-) > > > > They don't contain speech - instead, they contain few "computer > > generated" tones and I should store them in max quality for later > > proccessing (i.e. analysis)... > > > > Best regards, > > > > kokoska.rokoska > > > > > > Brian West napsal(a): > > > Really you want to keep 1-3 second files around? > > > > > > /b > > > > > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: > > > > > >> Thank you very much, Brian, for your interest! > > >> > > >> It is standard recording: > > >> > > >> > > >> > > >> Best regards, > > >> > > >> kokoska.rokoska > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Wed Nov 25 10:17:21 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 13:17:21 -0500 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: For that you would need to fully exchange session state into the sip library, something that is not available in that lib at this time. On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: > HI, > thanks for your reply, my requirement is i am doing failover stuff with freeswitch. i dont want cut the calls when freeswitch dies, when failover happens mean one freeswitch dies we are going to start the second freeswitch, i dont want close call intiated by the first freeswtich, they are communicating with meida(bypass media). when one endpoing try to end the call at that time i want to close the call for the other end also. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/4bac4fdf/attachment-0002.html From lists at redbonez.net Wed Nov 25 10:21:25 2009 From: lists at redbonez.net (Adam Ford) Date: Wed, 25 Nov 2009 11:21:25 -0700 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <20091124064509.GA6360@hijacked.us> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> <00e101ca6cab$c3525240$49f6f6c0$@net> <21CB5F92-98DE-4622-ADC5-013462A93BD2@freeswitch.org> <20091124064509.GA6360@hijacked.us> Message-ID: <016701ca6dfc$1a8e9ae0$4fabd0a0$@net> Awesome, thanks Andrew, I will have to keep an eye out for that patch. To continue, last night I decided to tackle the business hours and holiday routing on my FreeSWITCH system. It turned out to not be quite as simple with the XML dialplan as I thought. After being up until 1am banging my head against the wall, I finally got the results I was after. I decided I would share what I tried, why I tried them, and the solution I ended with, in hopes of helping other newcomers such as myself. Goal: To have a single area to modify in order to affect business hours and holiday routing across all extensions. Sources: http://wiki.freeswitch.org/wiki/Dialplan_XML http://wiki.freeswitch.org/wiki/Time_of_Day_routing http://svn.freeswitch.org/svn/freeswitch/trunk/conf/dialplan/default.xml After reading the above documentation of time of day routing, my first thought was that following the main example on http://wiki.freeswitch.org/wiki/Time_of_Day_routing would be wildly inefficient for me. I didn't want to create 3 extensions for every 1 real extension, nor did I want to edit each individual extension if there were adjustments to the hours or holiday schedule. I was filled with hope when I re-read the top of the default.xml dialplan. As it implies that I could simply set a variable at the top of the dialplan that would be accessible to all following extensions. You veterans immediately realize this is silly, but that is what is implied to a newcomer reading the top of the default.xml diaplan. After lots of playing with different ways of trying to get this to work, I went back and re-read http://wiki.freeswitch.org/wiki/Time_of_Day_routing. The bottom of which points out why it won't work the way that it is in the default.xml dialplan, a "classic case of dialplan is parsed all at once." My next attempt was to make a catchall extension at the top of default.xml that would set the ${Status} variable, then pass the originally dial extension into a transfer application, as suggested by the bottom of the Time_of_Day_routing wiki page. This obviously didn't work with a catchall, as it just continued to loop through the catchall. However, it did set the variable and pass through the dialed extension, so I felt I was on the right track. After trying a few different things with no avail, I realized the catchall extension would work if I just had it jump contexts on transfer! So I moved my catchall to the public.xml, adjusted my OpenZAP context to be public instead of default (which is apparently the default), and viola everything worked flawlessly. Now I just had to add the ability to set ${Status} to closed on holidays. There could be better ways of organizing this, but I just created a holidays directory and included the xml, which added override conditions for holidays. ---------------------------------------------------------------------- public.xml - ---------------------------------------------------------------------- holidays/thanksgiving.xml ---------------------------------------------------------------------- default context extension - ---------------------------------------------------------------------- Call flow breakdown (for those who are new, so you can easily follow what is going on) - In this example, if someone calls the DID ending in 5651, the call is processed by the catchall 'business_hours' extension in the public.xml. The weekday, and hour of day, is checked and ${Status} set accordingly. The holiday conditions that were included are then processed, overriding the ${Status}, setting it to 'closed' if today meets the criteria of a holiday. The call is then transferred to extension 5651, in the default context, of the XML dialplan. In the default context dialplan, extension 5651 is processed, which checks to see if ${Status} = open. If ${Status}=open is true, the extension passes the call to the IVR 'coral_ivr'. If ${Status}=open is false, the extension passes the call to the IVR 'coral_after_hours_ivr'. The catchall in the public context works well for me, because my connection to the PSTN is direct, through OpenZAP. I do not have to use the public context to provide extension security, as the incoming dialed extensions is limited to my known DIDs. I imagine this is not recommended for those who use SIP providers, and need the public context to provide internal and external segregation of your extensions. For these situations, you could probably create a middle-man context with the business hours logic, in which to send all transfers from public to default through. To those experienced users, if I missed something, and am making this much more complicated that it needs to be, please let me know. Or if you see potential errors or problems with my configuration, please let me know. I am new to this after all. Thanks all, especially you FreeSWITCH developers/contributors. I love the clean, efficient, logical design of FreeSWITCH. -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Thompson Sent: Monday, November 23, 2009 11:45 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Business/holiday hours routing On Mon, Nov 23, 2009 at 08:17:46PM -0600, Brian West wrote: > He's working on it for SVN... I recommended the format and to add the > phases of the moon and zodiac signs just for giggles. > I'll probably get a patch in this week (or early next) I'm thinking of changing the format so that "week of month" becomes its own value so you could compare against mweek as well as wday so thanksgiving + extension becomes something like If I really get ambitious I'd also like to allow wday="mon-fri" so I don't always forget that days are 1-indexed from sunday :) Andrew _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From afritzlists at fritztech.com Wed Nov 25 10:36:43 2009 From: afritzlists at fritztech.com (Andrew Fritz) Date: Wed, 25 Nov 2009 12:36:43 -0600 Subject: [Freeswitch-users] Patch: VMD Configurable MIN_TIME Message-ID: <4B0D793B.5040700@fritztech.com> I've created a patch to override the value of MIN_TIME in the vmd modules using a channel variable. In this way, it can be configured on a call by call basis. The channel variable is name "vmd_min_time". I didn't add the other detection parameters, but doing so would be straight forward. So, in our app, we can catch T-Mobile and the other problematic cell carriers beeps. I did this because in our app, we would rather have false positives than miss the start of recording on a voice mail system. This way, anyone using the VMD module can configure the vmd module to be as touchy or hard to trigger as they would like. I not sure how to implement it (at least in the vmd module code), but a way to make mod_vmd more robust to false positives, especially with short beeps would be to have it look for short silence immediately proceeding and/or following the beep. I've noticed that it tends to trigger on noise if there is tone in the noise, if for example I extend a syllable in a word or I have music on in the background. However on a voice mail system there will likely be a short near silence before the tone and an indefinite silence after it. In fact, background noise should be non-existent, except for line noise which should be Gaussian and not look like a structured tone. Looking for a beep + near silence after it for some period should eliminate many false positive where tones are embedded in other sounds (e.g. music or someone holding a vowel for longer than normal). Andrew -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: mod_vmd.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/653e39fa/attachment-0002.txt From rupa at rupa.com Wed Nov 25 11:06:20 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 25 Nov 2009 13:06:20 -0600 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: <11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com> References: <11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com> Message-ID: Stephen, I think you've jumped into the middle of a thread about sip 302, not about http. Anyway, you might want to look at using mod_curl instead of mod_spidermonkey_curl. mod_curl can give you a json response which you can then parse easily in javascript or any other language. The json response has the http response code, all headers, and the body. On Wed, Nov 25, 2009 at 12:01 PM, Stephen Crosby wrote: > Surprisingly, I've found no way to access the HTTP response status code > using mod_spidermonkey_curl. I'd love to see this feature added or discussed > if it already exists and I'm missing it. > > --Stephen > > > On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris wrote: > >> In trunk there is a sofia profile setting to allow dialplan processing of >> 302 responses. This won't get you back into your same javascript, but you >> can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >> > >> > I have considered writing JavaScript code to bridge two calls together. >> However, I would like to perform custom handling of the 302 Moved >> Temporarily response. How do I handle the 302 Moved Temporarily response if >> I use JavaScript? >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/5fa958fc/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 25 11:13:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 13:13:25 -0600 Subject: [Freeswitch-users] Patch: VMD Configurable MIN_TIME In-Reply-To: <4B0D793B.5040700@fritztech.com> References: <4B0D793B.5040700@fritztech.com> Message-ID: <191c3a030911251113t777a4516y1b23824c147f1d45@mail.gmail.com> can you submit your patch to jira as an improvement under the "application modules" section please. On Wed, Nov 25, 2009 at 12:36 PM, Andrew Fritz wrote: > I've created a patch to override the value of MIN_TIME in the vmd modules > using a channel variable. In this way, it can be configured on a call by > call basis. The channel variable is name "vmd_min_time". I didn't add the > other detection parameters, but doing so would be straight forward. So, in > our app, we can catch T-Mobile and the other problematic cell carriers > beeps. > > I did this because in our app, we would rather have false positives than > miss the start of recording on a voice mail system. This way, anyone using > the VMD module can configure the vmd module to be as touchy or hard to > trigger as they would like. > > I not sure how to implement it (at least in the vmd module code), but a way > to make mod_vmd more robust to false positives, especially with short beeps > would be to have it look for short silence immediately proceeding and/or > following the beep. I've noticed that it tends to trigger on noise if there > is tone in the noise, if for example I extend a syllable in a word or I have > music on in the background. > > However on a voice mail system there will likely be a short near silence > before the tone and an indefinite silence after it. In fact, background > noise should be non-existent, except for line noise which should be Gaussian > and not look like a structured tone. Looking for a beep + near silence after > it for some period should eliminate many false positive where tones are > embedded in other sounds (e.g. music or someone holding a vowel for longer > than normal). > > Andrew > > Index: src/mod/applications/mod_vmd/mod_vmd.c > =================================================================== > --- src/mod/applications/mod_vmd/mod_vmd.c (revision 15668) > +++ src/mod/applications/mod_vmd/mod_vmd.c (working copy) > @@ -162,6 +162,8 @@ > /*! A count of how long a distinct beep was detected > * by the discreet energy separation algorithm. */ > switch_size_t timestamp; > + /*! The MIN_TIME to use for this call */ > + int minTime; > } vmd_session_info_t; > > static switch_bool_t process_data(vmd_session_info_t * vmd_info, > switch_frame_t * frame); > @@ -312,7 +314,7 @@ > > if (c < (POINTS - MAX_CHIRP)) { > vmd_info->state = BEEP_NOT_DETECTED; > - if (vmd_info->timestamp < MIN_TIME) { > + if (vmd_info->timestamp < vmd_info->minTime) { > break; > } > > @@ -541,6 +543,7 @@ > switch_channel_t *channel; > vmd_session_info_t *vmd_info; > int i; > + const char *minTimeString; > > if (session == NULL) > return; > @@ -588,6 +591,14 @@ > > switch_channel_set_private(channel, "_vmd_", bug); > > + minTimeString = switch_channel_get_variable(channel, > "vmd_min_time"); > + if (minTimeString != 0) { > + sscanf(minTimeString,"%d",&(vmd_info->minTime)); > + } else { > + vmd_info->minTime = MIN_TIME; > + } > + > + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "MIN_TIME for > call: %d\n",vmd_info->minTime); > } > > /*! \brief Called when the module shuts down > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/1df26015/attachment-0002.html From stevecrozz at gmail.com Wed Nov 25 11:26:49 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 25 Nov 2009 11:26:49 -0800 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: <11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com> Message-ID: <11990ade0911251126w1a6937fdga6ee6da79342305d@mail.gmail.com> My apologies, and thanks for the info. --Stephen On Wed, Nov 25, 2009 at 11:06 AM, Rupa Schomaker wrote: > Stephen, I think you've jumped into the middle of a thread about sip 302, > not about http. > > Anyway, you might want to look at using mod_curl instead of > mod_spidermonkey_curl. mod_curl can give you a json response which you can > then parse easily in javascript or any other language. The json response > has the http response code, all headers, and the body. > > > On Wed, Nov 25, 2009 at 12:01 PM, Stephen Crosby wrote: > >> Surprisingly, I've found no way to access the HTTP response status code >> using mod_spidermonkey_curl. I'd love to see this feature added or discussed >> if it already exists and I'm missing it. >> >> --Stephen >> >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan processing of >>> 302 responses. This won't get you back into your same javascript, but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>> > >>> > I have considered writing JavaScript code to bridge two calls together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily response if >>> I use JavaScript? >>> > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/9da7f56e/attachment-0002.html From samuelmukoti at gmail.com Wed Nov 25 11:40:00 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Wed, 25 Nov 2009 21:40:00 +0200 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: References: Message-ID: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> Hi all, I'm wanting to try out a my first large scale setup at the office, 200 extensions and 24 POTS incoming, also a T1 line once the telco guys are ready. I wanted assistance with choosing the most appropriate hardware. We already have about 150 analogue phones, and I was wondering what's best? A couple of grandstream FXS GXW4024? Also for my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma or digium card? The best voice quality is paramount. Lastly for T1 what cards are recommeded, I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, would that perform? Or do I need hardware transcoding? Thank you, Sam Twitter: twitter.com/samuelmukoti On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > Today's Topics: > > 1. Re: mod_conference kick to abort invitations (Michael Jerris) > 2. Re: Handling the 302 Moved Temporarily response from > JavaScript (Michael Jerris) > 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) > 4. Re: remote_media_ip variable not set (Michael Jerris) > 5. Re: How to find whether the destination extension supports > encryption (Michael Jerris) > 6. Re: Bypass_media and re_invite (srinivasula reddy) > 7. Re: Handling the 302 Moved Temporarily response from > JavaScript (Stephen Crosby) > 8. Re: Handling the 302 Moved Temporarily response from > JavaScript (Tihomir Culjaga) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 25 Nov 2009 12:44:46 -0500 > From: Michael Jerris > Subject: Re: [Freeswitch-users] mod_conference kick to abort > invitations > To: freeswitch-users at lists.freeswitch.org > Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> > Content-Type: text/plain; charset="windows-1252" > > Its a feature we don't have, patches welcome. > > Mike > > On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > >> Hi members, >> I?m controlling freeswitch with the conference module via xmlrpc. >> >> Is it desired that the kick command can only kick users that are >> connected to the conference? >> Is there no chance abort an invitation? >> The kick command has no effect until the person I invited with the >> dial command is connected. > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Wed, 25 Nov 2009 12:45:50 -0500 > From: Michael Jerris > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > response from JavaScript > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset=us-ascii > > In trunk there is a sofia profile setting to allow dialplan > processing of 302 responses. This won't get you back into your same > javascript, but you can probably do something clever from there. > > Mike > > On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> >> I have considered writing JavaScript code to bridge two calls >> together. However, I would like to perform custom handling of the >> 302 Moved Temporarily response. How do I handle the 302 Moved >> Temporarily response if I use JavaScript? >> > > > > ------------------------------ > > Message: 3 > Date: Wed, 25 Nov 2009 11:46:05 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via > proxy. > To: freeswitch-users at lists.freeswitch.org > Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> > Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes > > Yes an alias will be required for every domain you run on the profile > so it can find it. > > /b > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > >> Try an alias on the sip profile. >> >> Mike > > > > > ------------------------------ > > Message: 4 > Date: Wed, 25 Nov 2009 12:47:37 -0500 > From: Michael Jerris > Subject: Re: [Freeswitch-users] remote_media_ip variable not set > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset=us-ascii > > It's possible it does not. I just added some code to set it on auto- > adjust so it might be there sometimes now. You might need to add > some code in mod_sofia to add it other times. Maybe it makes sense > to move that var setting down to switch_rtp.c. Patches for this > would be welcome. > > Thanks > > Mike > > On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > >> Hi, >> >> In the case of proxy_media=true, does it gets set at all then? > > > > > ------------------------------ > > Message: 5 > Date: Wed, 25 Nov 2009 12:48:39 -0500 > From: Michael Jerris > Subject: Re: [Freeswitch-users] How to find whether the destination > extension supports encryption > To: freeswitch-users at lists.freeswitch.org > Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> > Content-Type: text/plain; charset=us-ascii > > You can send the call with secure enabled and if it supports it it > will use it. > > Mike > > On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > >> Hello, >> >> We have a mix of phones that support RTP encryption and those that >> do not. I have to support both types in the meanwhile, and would >> like to have encryption enabled on the relevant leg, even if the >> other leg does not support it (why? one of our ATAs either must >> have it unencrypted or have it encrypted, but cannot have both). >> >> How do I find whether the destination supports encryption? I do not >> want to manage an additional table in the database... >> > > > > ------------------------------ > > Message: 6 > Date: Wed, 25 Nov 2009 23:25:01 +0530 > From: srinivasula reddy > Subject: Re: [Freeswitch-users] Bypass_media and re_invite > To: freeswitch-users at lists.freeswitch.org > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > HI, > thanks for your reply, my requirement is i am doing failover stuff > with > freeswitch. i dont want cut the calls when freeswitch dies, when > failover > happens mean one freeswitch dies we are going to start the second > freeswitch, i dont want close call intiated by the first > freeswtich, they > are communicating with meida(bypass media). when one endpoing try to > end the > call at that time i want to close the call for the other end also. > > > srinivas > > On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris > wrote: > >> FreeSWITCH will kill the calls when you shut it down, if you >> intentionally >> kill the network without shutting down FreeSWITCH the only thing >> you can do >> is enable session timers or rtp timers in the soft phones to kill >> the call >> when FreeSWITCH dies or when the call is over. >> >> Mike >> >> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >> >>> Hi All, >>> >>> goodmorning to all, i have a scenario, two pjsua clients are >>> connected >> with Freeswitch and they are in call and bypass_media=true. i >> close the >> Freeswitch server, still they are in call, again i started the >> Freeswitch, >> and registerd these two endpoints, now how can i end the call >> (estabilished >> by the first Freeswitch)? if i call re_invite will it estabilish >> the call >> between two endpoints? >>> any idea? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > > > -- > Srinivasula Reddy K > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html > > ------------------------------ > > Message: 7 > Date: Wed, 25 Nov 2009 10:01:14 -0800 > From: Stephen Crosby > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > response from JavaScript > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Surprisingly, I've found no way to access the HTTP response status > code > using mod_spidermonkey_curl. I'd love to see this feature added or > discussed > if it already exists and I'm missing it. > > --Stephen > > On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris > wrote: > >> In trunk there is a sofia profile setting to allow dialplan >> processing of >> 302 responses. This won't get you back into your same javascript, >> but you >> can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. >> However, I would like to perform custom handling of the 302 Moved >> Temporarily response. How do I handle the 302 Moved Temporarily >> response if >> I use JavaScript? >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html > > ------------------------------ > > Message: 8 > Date: Wed, 25 Nov 2009 19:04:56 +0100 > From: Tihomir Culjaga > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > response from JavaScript > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > this is how i do it from the dialplan: > > > > > > expression="^(300030)(.*)|^\+(300030)(.*)"> > > > > > data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> > data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: > 1:32} : > ${caller_id_number})}"/> > > data="aPfx=${caller_id_number:0:6}"/> > data="aNum=${caller_id_number:6:16}"/> > data="IP_ADDR=${network_addr}:5060"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris > wrote: > >> In trunk there is a sofia profile setting to allow dialplan >> processing of >> 302 responses. This won't get you back into your same javascript, >> but you >> can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. >> However, I would like to perform custom handling of the 302 Moved >> Temporarily response. How do I handle the 302 Moved Temporarily >> response if >> I use JavaScript? >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > End of FreeSWITCH-users Digest, Vol 41, Issue 189 > ************************************************* From srinivas.ksvreddy at gmail.com Wed Nov 25 11:47:43 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Thu, 26 Nov 2009 01:17:43 +0530 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: can please tell me how can i exchange session state into sip library. Thanks srinivas On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: > For that you would need to fully exchange session state into the sip > library, something that is not available in that lib at this time. > > > On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: > > HI, > thanks for your reply, my requirement is i am doing failover stuff with > freeswitch. i dont want cut the calls when freeswitch dies, when failover > happens mean one freeswitch dies we are going to start the second > freeswitch, i dont want close call intiated by the first freeswtich, they > are communicating with meida(bypass media). when one endpoing try to end the > call at that time i want to close the call for the other end also. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/fda28014/attachment-0002.html From chris.chen2004 at gmail.com Wed Nov 25 11:55:19 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Wed, 25 Nov 2009 14:55:19 -0500 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> Message-ID: <507898380911251155k29c52989v30d0e39bb18d4ac1@mail.gmail.com> One suggestion to you, please never consider the GXW4108 for any business use unless just in LAB. The GXW4108 will work when it is working,but I can tell you within one year you will be regretting your choice for use of GXW4108 if you put into production for business use. Chris On Wed, Nov 25, 2009 at 2:40 PM, Samuel Mukoti wrote: > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org > wrote: > > > Send FreeSWITCH-users mailing list submissions to > > freeswitch-users at lists.freeswitch.org > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > or, via email, send a message with subject or body 'help' to > > freeswitch-users-request at lists.freeswitch.org > > > > You can reach the person managing the list at > > freeswitch-users-owner at lists.freeswitch.org > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of FreeSWITCH-users digest..." > > > > > > Today's Topics: > > > > 1. Re: mod_conference kick to abort invitations (Michael Jerris) > > 2. Re: Handling the 302 Moved Temporarily response from > > JavaScript (Michael Jerris) > > 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) > > 4. Re: remote_media_ip variable not set (Michael Jerris) > > 5. Re: How to find whether the destination extension supports > > encryption (Michael Jerris) > > 6. Re: Bypass_media and re_invite (srinivasula reddy) > > 7. Re: Handling the 302 Moved Temporarily response from > > JavaScript (Stephen Crosby) > > 8. Re: Handling the 302 Moved Temporarily response from > > JavaScript (Tihomir Culjaga) > > > > > > ---------------------------------------------------------------------- > > > > Message: 1 > > Date: Wed, 25 Nov 2009 12:44:46 -0500 > > From: Michael Jerris > > Subject: Re: [Freeswitch-users] mod_conference kick to abort > > invitations > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> > > Content-Type: text/plain; charset="windows-1252" > > > > Its a feature we don't have, patches welcome. > > > > Mike > > > > On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > > > >> Hi members, > >> I?m controlling freeswitch with the conference module via xmlrpc. > >> > >> Is it desired that the kick command can only kick users that are > >> connected to the conference? > >> Is there no chance abort an invitation? > >> The kick command has no effect until the person I invited with the > >> dial command is connected. > > > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html > > > > ------------------------------ > > > > Message: 2 > > Date: Wed, 25 Nov 2009 12:45:50 -0500 > > From: Michael Jerris > > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > > response from JavaScript > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: > > Content-Type: text/plain; charset=us-ascii > > > > In trunk there is a sofia profile setting to allow dialplan > > processing of 302 responses. This won't get you back into your same > > javascript, but you can probably do something clever from there. > > > > Mike > > > > On Nov 24, 2009, at 5:04 PM, John Platts wrote: > > > >> > >> I have considered writing JavaScript code to bridge two calls > >> together. However, I would like to perform custom handling of the > >> 302 Moved Temporarily response. How do I handle the 302 Moved > >> Temporarily response if I use JavaScript? > >> > > > > > > > > ------------------------------ > > > > Message: 3 > > Date: Wed, 25 Nov 2009 11:46:05 -0600 > > From: Brian West > > Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via > > proxy. > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> > > Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes > > > > Yes an alias will be required for every domain you run on the profile > > so it can find it. > > > > /b > > > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > > >> Try an alias on the sip profile. > >> > >> Mike > > > > > > > > > > ------------------------------ > > > > Message: 4 > > Date: Wed, 25 Nov 2009 12:47:37 -0500 > > From: Michael Jerris > > Subject: Re: [Freeswitch-users] remote_media_ip variable not set > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: > > Content-Type: text/plain; charset=us-ascii > > > > It's possible it does not. I just added some code to set it on auto- > > adjust so it might be there sometimes now. You might need to add > > some code in mod_sofia to add it other times. Maybe it makes sense > > to move that var setting down to switch_rtp.c. Patches for this > > would be welcome. > > > > Thanks > > > > Mike > > > > On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > > > >> Hi, > >> > >> In the case of proxy_media=true, does it gets set at all then? > > > > > > > > > > ------------------------------ > > > > Message: 5 > > Date: Wed, 25 Nov 2009 12:48:39 -0500 > > From: Michael Jerris > > Subject: Re: [Freeswitch-users] How to find whether the destination > > extension supports encryption > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> > > Content-Type: text/plain; charset=us-ascii > > > > You can send the call with secure enabled and if it supports it it > > will use it. > > > > Mike > > > > On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > > > >> Hello, > >> > >> We have a mix of phones that support RTP encryption and those that > >> do not. I have to support both types in the meanwhile, and would > >> like to have encryption enabled on the relevant leg, even if the > >> other leg does not support it (why? one of our ATAs either must > >> have it unencrypted or have it encrypted, but cannot have both). > >> > >> How do I find whether the destination supports encryption? I do not > >> want to manage an additional table in the database... > >> > > > > > > > > ------------------------------ > > > > Message: 6 > > Date: Wed, 25 Nov 2009 23:25:01 +0530 > > From: srinivasula reddy > > Subject: Re: [Freeswitch-users] Bypass_media and re_invite > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: > > > > Content-Type: text/plain; charset="iso-8859-1" > > > > HI, > > thanks for your reply, my requirement is i am doing failover stuff > > with > > freeswitch. i dont want cut the calls when freeswitch dies, when > > failover > > happens mean one freeswitch dies we are going to start the second > > freeswitch, i dont want close call intiated by the first > > freeswtich, they > > are communicating with meida(bypass media). when one endpoing try to > > end the > > call at that time i want to close the call for the other end also. > > > > > > srinivas > > > > On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris > > wrote: > > > >> FreeSWITCH will kill the calls when you shut it down, if you > >> intentionally > >> kill the network without shutting down FreeSWITCH the only thing > >> you can do > >> is enable session timers or rtp timers in the soft phones to kill > >> the call > >> when FreeSWITCH dies or when the call is over. > >> > >> Mike > >> > >> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > >> > >>> Hi All, > >>> > >>> goodmorning to all, i have a scenario, two pjsua clients are > >>> connected > >> with Freeswitch and they are in call and bypass_media=true. i > >> close the > >> Freeswitch server, still they are in call, again i started the > >> Freeswitch, > >> and registerd these two endpoints, now how can i end the call > >> (estabilished > >> by the first Freeswitch)? if i call re_invite will it estabilish > >> the call > >> between two endpoints? > >>> any idea? > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Srinivasula Reddy K > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html > > > > ------------------------------ > > > > Message: 7 > > Date: Wed, 25 Nov 2009 10:01:14 -0800 > > From: Stephen Crosby > > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > > response from JavaScript > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: > > <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> > > Content-Type: text/plain; charset="utf-8" > > > > Surprisingly, I've found no way to access the HTTP response status > > code > > using mod_spidermonkey_curl. I'd love to see this feature added or > > discussed > > if it already exists and I'm missing it. > > > > --Stephen > > > > On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris > > wrote: > > > >> In trunk there is a sofia profile setting to allow dialplan > >> processing of > >> 302 responses. This won't get you back into your same javascript, > >> but you > >> can probably do something clever from there. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> > >>> > >>> I have considered writing JavaScript code to bridge two calls > >>> together. > >> However, I would like to perform custom handling of the 302 Moved > >> Temporarily response. How do I handle the 302 Moved Temporarily > >> response if > >> I use JavaScript? > >>> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html > > > > ------------------------------ > > > > Message: 8 > > Date: Wed, 25 Nov 2009 19:04:56 +0100 > > From: Tihomir Culjaga > > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > > response from JavaScript > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: > > <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> > > Content-Type: text/plain; charset="iso-8859-1" > > > > this is how i do it from the dialplan: > > > > > > > > > > > > > expression="^(300030)(.*)|^\+(300030)(.*)"> > > > > > > > > > > > data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> > > > data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: > > 1:32} : > > ${caller_id_number})}"/> > > > > > data="aPfx=${caller_id_number:0:6}"/> > > > data="aNum=${caller_id_number:6:16}"/> > > > data="IP_ADDR=${network_addr}:5060"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris > > wrote: > > > >> In trunk there is a sofia profile setting to allow dialplan > >> processing of > >> 302 responses. This won't get you back into your same javascript, > >> but you > >> can probably do something clever from there. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> > >>> > >>> I have considered writing JavaScript code to bridge two calls > >>> together. > >> However, I would like to perform custom handling of the 302 Moved > >> Temporarily response. How do I handle the 302 Moved Temporarily > >> response if > >> I use JavaScript? > >>> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html > > > > ------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > > > > End of FreeSWITCH-users Digest, Vol 41, Issue 189 > > ************************************************* > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/79bd1f8e/attachment-0002.html From brian at freeswitch.org Wed Nov 25 11:59:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 13:59:03 -0600 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <507898380911251155k29c52989v30d0e39bb18d4ac1@mail.gmail.com> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <507898380911251155k29c52989v30d0e39bb18d4ac1@mail.gmail.com> Message-ID: Or if you're dancing with the stars!!!!!! /b On Nov 25, 2009, at 1:55 PM, Chris Chen wrote: > One suggestion to you, please never consider the GXW4108 for any > business use unless just in LAB. The GXW4108 will work when it is > working,but I can tell you within one year you will be regretting > your choice for use of GXW4108 if you put into production for > business use. > > Chris From samuelmukoti at gmail.com Wed Nov 25 12:16:39 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Wed, 25 Nov 2009 22:16:39 +0200 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> Message-ID: Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews at sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org > wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> In trunk there is a sofia profile setting to allow dialplan >> processing of 302 responses. This won't get you back into your same >> javascript, but you can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. However, I would like to perform custom handling of the >>> 302 Moved Temporarily response. How do I handle the 302 Moved >>> Temporarily response if I use JavaScript? >>> >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Wed, 25 Nov 2009 11:46:05 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> proxy. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes >> >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >>> Try an alias on the sip profile. >>> >>> Mike >> >> >> >> >> ------------------------------ >> >> Message: 4 >> Date: Wed, 25 Nov 2009 12:47:37 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> It's possible it does not. I just added some code to set it on auto- >> adjust so it might be there sometimes now. You might need to add >> some code in mod_sofia to add it other times. Maybe it makes sense >> to move that var setting down to switch_rtp.c. Patches for this >> would be welcome. >> >> Thanks >> >> Mike >> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: >> >>> Hi, >>> >>> In the case of proxy_media=true, does it gets set at all then? >> >> >> >> >> ------------------------------ >> >> Message: 5 >> Date: Wed, 25 Nov 2009 12:48:39 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] How to find whether the destination >> extension supports encryption >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> >> Content-Type: text/plain; charset=us-ascii >> >> You can send the call with secure enabled and if it supports it it >> will use it. >> >> Mike >> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: >> >>> Hello, >>> >>> We have a mix of phones that support RTP encryption and those that >>> do not. I have to support both types in the meanwhile, and would >>> like to have encryption enabled on the relevant leg, even if the >>> other leg does not support it (why? one of our ATAs either must >>> have it unencrypted or have it encrypted, but cannot have both). >>> >>> How do I find whether the destination supports encryption? I do not >>> want to manage an additional table in the database... >>> >> >> >> >> ------------------------------ >> >> Message: 6 >> Date: Wed, 25 Nov 2009 23:25:01 +0530 >> From: srinivasula reddy >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> >> Content-Type: text/plain; charset="iso-8859-1" >> >> HI, >> thanks for your reply, my requirement is i am doing failover stuff >> with >> freeswitch. i dont want cut the calls when freeswitch dies, when >> failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the first >> freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to >> end the >> call at that time i want to close the call for the other end also. >> >> >> srinivas >> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris >> wrote: >> >>> FreeSWITCH will kill the calls when you shut it down, if you >>> intentionally >>> kill the network without shutting down FreeSWITCH the only thing >>> you can do >>> is enable session timers or rtp timers in the soft phones to kill >>> the call >>> when FreeSWITCH dies or when the call is over. >>> >>> Mike >>> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >>> >>>> Hi All, >>>> >>>> goodmorning to all, i have a scenario, two pjsua clients are >>>> connected >>> with Freeswitch and they are in call and bypass_media=true. i >>> close the >>> Freeswitch server, still they are in call, again i started the >>> Freeswitch, >>> and registerd these two endpoints, now how can i end the call >>> (estabilished >>> by the first Freeswitch)? if i call re_invite will it estabilish >>> the call >>> between two endpoints? >>>> any idea? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Srinivasula Reddy K >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html >> >> ------------------------------ >> >> Message: 7 >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> From: Stephen Crosby >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Surprisingly, I've found no way to access the HTTP response status >> code >> using mod_spidermonkey_curl. I'd love to see this feature added or >> discussed >> if it already exists and I'm missing it. >> >> --Stephen >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html >> >> ------------------------------ >> >> Message: 8 >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> From: Tihomir Culjaga >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> this is how i do it from the dialplan: >> >> >> >> >> >> > expression="^(300030)(.*)|^\+(300030)(.*)"> >> >> >> >> >> > data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> >> > data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: >> 1:32} : >> ${caller_id_number})}"/> >> >> > data="aPfx=${caller_id_number:0:6}"/> >> > data="aNum=${caller_id_number:6:16}"/> >> > data="IP_ADDR=${network_addr}:5060"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 >> ************************************************* > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From kristian.kielhofner at gmail.com Wed Nov 25 12:30:09 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 25 Nov 2009 15:30:09 -0500 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <507898380911251155k29c52989v30d0e39bb18d4ac1@mail.gmail.com> Message-ID: <2d9149cd0911251230h71ebc7b8n1203628bf97ed218@mail.gmail.com> That was a *GREAT* e-mail. On Wed, Nov 25, 2009 at 2:59 PM, Brian West wrote: > Or if you're dancing with the stars!!!!!! > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From mike at jerris.com Wed Nov 25 12:35:20 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 15:35:20 -0500 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: "something that is not available in that lib at this time." Mike On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote: > can please tell me how can i exchange session state into sip library. > > Thanks > srinivas > > On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: > For that you would need to fully exchange session state into the sip library, something that is not available in that lib at this time. > > > On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: > >> HI, >> thanks for your reply, my requirement is i am doing failover stuff with freeswitch. i dont want cut the calls when freeswitch dies, when failover happens mean one freeswitch dies we are going to start the second freeswitch, i dont want close call intiated by the first freeswtich, they are communicating with meida(bypass media). when one endpoing try to end the call at that time i want to close the call for the other end also. >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/c9abfc4c/attachment-0002.html From chris.chen2004 at gmail.com Wed Nov 25 12:40:25 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Wed, 25 Nov 2009 15:40:25 -0500 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> Message-ID: <507898380911251240q1a71b234x1458284bbdccc093@mail.gmail.com> You haven't really put it into production for more than one year. The issue with GXW4108 is that after some time, say a couple of months, either all FXO ports not working, or worse, some FXO ports not working, but after power recycling, they will come back to work for some time until on strike again at some time you have no control. This had been reported for a couple of years without improvement. Go google search you will find out, this has happened to many GXW4108 users. On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti wrote: > Thank you for those tips, > > I do have some small setups using gxw4108 they work or, except CID > doesn't seem to work. I will try the channel bank route - just don't > know too much about the setup options or how you'd purchase the > correct config, eg. For 150 FXS channel bank, can I get a single PCI > card for that? > > I may end up using the grandstream fxs gateways then use the T1 > channel bank from sangoma, > > Thank you all.. > > Lastly, I know asterisk now has an offical skype_ module, Is there > anything similar I could use? > > > On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: > > > Samuel - you could go with FXS gateways or channel banks. If you go > > the gateway route Grandstream or Audiocodes would work fine. > > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > > you could use a 24FXO channel bank cross connected with Rhino T1 > > cards, or individual FXO gateways but you may have a hard time > > finding 24 ports of FXO in a single GW. Best performing T1 cards in > > my experience (thousands of deployments) are Sangoma. Your server > > configuration looks fine. > > > > Cory J. Andrews > > Director New Market Initiatives > > > > Sayers Media Group > > VoIP Supply, LLC > > 454 Sonwil Drive > > Buffalo, NY 14225 > > 716-250-3402 OFFICE > > 716-630-1548 FAX > > 716-601-4474 MOBILE > > candrews at sayersmedia.com > > > > > > Have I exceeded your expectations? Please share your experience > > with my boss, Benjamin P. Sayers, CEO > > > > NOTICE: The information contained in this email and any document > > attached hereto is intended only for the named recipient(s). It is > > the property of the VoIP Supply, LLC and shall not be used, > > disclosed or reproduced without the express written consent of VoIP > > Supply, LLC. If you are not the intended recipient, nor the employee > > or agent responsible for delivering this message in confidence to > > the intended recipient(s), you are hereby notified that you have > > received this transmittal in error, and any review, dissemination, > > distribution or copying of this transmittal or its attachments is > > strictly prohibited. If you have received this transmittal and/or > > attachments in error, please notify me immediately by reply e-mail > > or telephone and then delete this message, including any > > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > > 14225 USA. > > > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Samuel Mukoti > > Sent: Wednesday, November 25, 2009 2:40 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Grandstream gateways > > > > Hi all, > > > > I'm wanting to try out a my first large scale setup at the office, 200 > > extensions and 24 POTS incoming, also a T1 line once the telco guys > > are ready. I wanted assistance with choosing the most appropriate > > hardware. We already have about 150 analogue phones, and I was > > wondering what's best? A couple of grandstream FXS GXW4024? Also for > > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > > or digium card? The best voice quality is paramount. Lastly for T1 > > what cards are recommeded, > > > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > > would that perform? Or do I need hardware transcoding? > > > > Thank you, > > > > Sam > > > > Twitter: twitter.com/samuelmukoti > > > > > > On 25 Nov,2009, at 8:05 PM, > freeswitch-users-request at lists.freeswitch.org > > wrote: > > > >> Send FreeSWITCH-users mailing list submissions to > >> freeswitch-users at lists.freeswitch.org > >> > >> To subscribe or unsubscribe via the World Wide Web, visit > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> or, via email, send a message with subject or body 'help' to > >> freeswitch-users-request at lists.freeswitch.org > >> > >> You can reach the person managing the list at > >> freeswitch-users-owner at lists.freeswitch.org > >> > >> When replying, please edit your Subject line so it is more specific > >> than "Re: Contents of FreeSWITCH-users digest..." > >> > >> > >> Today's Topics: > >> > >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) > >> 2. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Michael Jerris) > >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) > >> 4. Re: remote_media_ip variable not set (Michael Jerris) > >> 5. Re: How to find whether the destination extension supports > >> encryption (Michael Jerris) > >> 6. Re: Bypass_media and re_invite (srinivasula reddy) > >> 7. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Stephen Crosby) > >> 8. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Tihomir Culjaga) > >> > >> > >> --- > >> ------------------------------------------------------------------- > >> > >> Message: 1 > >> Date: Wed, 25 Nov 2009 12:44:46 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] mod_conference kick to abort > >> invitations > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> > >> Content-Type: text/plain; charset="windows-1252" > >> > >> Its a feature we don't have, patches welcome. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > >> > >>> Hi members, > >>> I?m controlling freeswitch with the conference module via xmlrpc. > >>> > >>> Is it desired that the kick command can only kick users that are > >>> connected to the conference? > >>> Is there no chance abort an invitation? > >>> The kick command has no effect until the person I invited with the > >>> dial command is connected. > >> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 2 > >> Date: Wed, 25 Nov 2009 12:45:50 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii > >> > >> In trunk there is a sofia profile setting to allow dialplan > >> processing of 302 responses. This won't get you back into your same > >> javascript, but you can probably do something clever from there. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> > >>> > >>> I have considered writing JavaScript code to bridge two calls > >>> together. However, I would like to perform custom handling of the > >>> 302 Moved Temporarily response. How do I handle the 302 Moved > >>> Temporarily response if I use JavaScript? > >>> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 3 > >> Date: Wed, 25 Nov 2009 11:46:05 -0600 > >> From: Brian West > >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via > >> proxy. > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> > >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes > >> > >> Yes an alias will be required for every domain you run on the profile > >> so it can find it. > >> > >> /b > >> > >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > >> > >>> Try an alias on the sip profile. > >>> > >>> Mike > >> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 4 > >> Date: Wed, 25 Nov 2009 12:47:37 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii > >> > >> It's possible it does not. I just added some code to set it on auto- > >> adjust so it might be there sometimes now. You might need to add > >> some code in mod_sofia to add it other times. Maybe it makes sense > >> to move that var setting down to switch_rtp.c. Patches for this > >> would be welcome. > >> > >> Thanks > >> > >> Mike > >> > >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > >> > >>> Hi, > >>> > >>> In the case of proxy_media=true, does it gets set at all then? > >> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 5 > >> Date: Wed, 25 Nov 2009 12:48:39 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] How to find whether the destination > >> extension supports encryption > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> > >> Content-Type: text/plain; charset=us-ascii > >> > >> You can send the call with secure enabled and if it supports it it > >> will use it. > >> > >> Mike > >> > >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > >> > >>> Hello, > >>> > >>> We have a mix of phones that support RTP encryption and those that > >>> do not. I have to support both types in the meanwhile, and would > >>> like to have encryption enabled on the relevant leg, even if the > >>> other leg does not support it (why? one of our ATAs either must > >>> have it unencrypted or have it encrypted, but cannot have both). > >>> > >>> How do I find whether the destination supports encryption? I do not > >>> want to manage an additional table in the database... > >>> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 6 > >> Date: Wed, 25 Nov 2009 23:25:01 +0530 > >> From: srinivasula reddy > >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> > >> Content-Type: text/plain; charset="iso-8859-1" > >> > >> HI, > >> thanks for your reply, my requirement is i am doing failover stuff > >> with > >> freeswitch. i dont want cut the calls when freeswitch dies, when > >> failover > >> happens mean one freeswitch dies we are going to start the second > >> freeswitch, i dont want close call intiated by the first > >> freeswtich, they > >> are communicating with meida(bypass media). when one endpoing try to > >> end the > >> call at that time i want to close the call for the other end also. > >> > >> > >> srinivas > >> > >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris > >> wrote: > >> > >>> FreeSWITCH will kill the calls when you shut it down, if you > >>> intentionally > >>> kill the network without shutting down FreeSWITCH the only thing > >>> you can do > >>> is enable session timers or rtp timers in the soft phones to kill > >>> the call > >>> when FreeSWITCH dies or when the call is over. > >>> > >>> Mike > >>> > >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > >>> > >>>> Hi All, > >>>> > >>>> goodmorning to all, i have a scenario, two pjsua clients are > >>>> connected > >>> with Freeswitch and they are in call and bypass_media=true. i > >>> close the > >>> Freeswitch server, still they are in call, again i started the > >>> Freeswitch, > >>> and registerd these two endpoints, now how can i end the call > >>> (estabilished > >>> by the first Freeswitch)? if i call re_invite will it estabilish > >>> the call > >>> between two endpoints? > >>>> any idea? > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Srinivasula Reddy K > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 7 > >> Date: Wed, 25 Nov 2009 10:01:14 -0800 > >> From: Stephen Crosby > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> > >> Content-Type: text/plain; charset="utf-8" > >> > >> Surprisingly, I've found no way to access the HTTP response status > >> code > >> using mod_spidermonkey_curl. I'd love to see this feature added or > >> discussed > >> if it already exists and I'm missing it. > >> > >> --Stephen > >> > >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris > >> wrote: > >> > >>> In trunk there is a sofia profile setting to allow dialplan > >>> processing of > >>> 302 responses. This won't get you back into your same javascript, > >>> but you > >>> can probably do something clever from there. > >>> > >>> Mike > >>> > >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >>> > >>>> > >>>> I have considered writing JavaScript code to bridge two calls > >>>> together. > >>> However, I would like to perform custom handling of the 302 Moved > >>> Temporarily response. How do I handle the 302 Moved Temporarily > >>> response if > >>> I use JavaScript? > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 8 > >> Date: Wed, 25 Nov 2009 19:04:56 +0100 > >> From: Tihomir Culjaga > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> > >> Content-Type: text/plain; charset="iso-8859-1" > >> > >> this is how i do it from the dialplan: > >> > >> > >> > >> > >> > >> >> expression="^(300030)(.*)|^\+(300030)(.*)"> > >> > >> > >> > >> > >> >> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> > >> >> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: > >> 1:32} : > >> ${caller_id_number})}"/> > >> > >> >> data="aPfx=${caller_id_number:0:6}"/> > >> >> data="aNum=${caller_id_number:6:16}"/> > >> >> data="IP_ADDR=${network_addr}:5060"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris > >> wrote: > >> > >>> In trunk there is a sofia profile setting to allow dialplan > >>> processing of > >>> 302 responses. This won't get you back into your same javascript, > >>> but you > >>> can probably do something clever from there. > >>> > >>> Mike > >>> > >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >>> > >>>> > >>>> I have considered writing JavaScript code to bridge two calls > >>>> together. > >>> However, I would like to perform custom handling of the 302 Moved > >>> Temporarily response. How do I handle the 302 Moved Temporarily > >>> response if > >>> I use JavaScript? > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html > >> > >> ------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > >> > >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 > >> ************************************************* > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/f6d17116/attachment-0002.html From brian at freeswitch.org Wed Nov 25 12:45:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Nov 2009 14:45:01 -0600 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <507898380911251240q1a71b234x1458284bbdccc093@mail.gmail.com> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <507898380911251240q1a71b234x1458284bbdccc093@mail.gmail.com> Message-ID: <5D71F7D7-7E93-499C-AFCA-61846CB3217F@freeswitch.org> Kill it, sunshine. /b On Nov 25, 2009, at 2:40 PM, Chris Chen wrote: > You haven't really put it into production for more than one year. > The issue with GXW4108 is that after some time, say a couple of > months, either all FXO ports not working, or worse, some FXO ports > not working, but after power recycling, they will come back to work > for some time until on strike again at some time you have no control. > > This had been reported for a couple of years without improvement. Go > google search you will find out, this has happened to many GXW4108 > users. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/9e386b99/attachment-0002.html From srinivas.ksvreddy at gmail.com Wed Nov 25 12:58:33 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Thu, 26 Nov 2009 02:28:33 +0530 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: thanks for your reply mike, is there any api in freeswitch or any thing else to update lib programatically from pjsua. srinivas On Thu, Nov 26, 2009 at 2:05 AM, Michael Jerris wrote: > "something that is not available in that lib at this time." > > Mike > > On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote > > can please tell me how can i exchange session state into sip library. > > Thanks > srinivas > > On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: > >> For that you would need to fully exchange session state into the sip >> library, *something that is not available in that lib at this time.* >> >> >> On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: >> >> HI, >> thanks for your reply, my requirement is i am doing failover stuff with >> freeswitch. i dont want cut the calls when freeswitch dies, when failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the first freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to end the >> call at that time i want to close the call for the other end also. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/bdac13c8/attachment-0002.html From testeador01 at gmail.com Wed Nov 25 13:00:14 2009 From: testeador01 at gmail.com (Milena) Date: Wed, 25 Nov 2009 16:00:14 -0500 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <507898380911251240q1a71b234x1458284bbdccc093@mail.gmail.com> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <507898380911251240q1a71b234x1458284bbdccc093@mail.gmail.com> Message-ID: Hello, Samuel: We also have some GXW4104 gateways, in small production/testing environments; our caller id works fine and none of them has failed in over a year of being used. The thing that i dislike about the GXW series is that it has no support for early media. Everyone: What FXO devices do you currently use / recommend? 2009/11/25 Chris Chen > You haven't really put it into production for more than one year. The issue > with GXW4108 is that after some time, say a couple of months, either all FXO > ports not working, or worse, some FXO ports not working, but after power > recycling, they will come back to work for some time until on strike again > at some time you have no control. > > This had been reported for a couple of years without improvement. Go google > search you will find out, this has happened to many GXW4108 users. > > > > On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti wrote: > >> Thank you for those tips, >> >> I do have some small setups using gxw4108 they work or, except CID >> doesn't seem to work. I will try the channel bank route - just don't >> know too much about the setup options or how you'd purchase the >> correct config, eg. For 150 FXS channel bank, can I get a single PCI >> card for that? >> >> I may end up using the grandstream fxs gateways then use the T1 >> channel bank from sangoma, >> >> Thank you all.. >> >> Lastly, I know asterisk now has an offical skype_ module, Is there >> anything similar I could use? >> >> >> On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: >> >> > Samuel - you could go with FXS gateways or channel banks. If you go >> > the gateway route Grandstream or Audiocodes would work fine. >> > Audiocodes are a bit more telco grade. If you have 25 POTS incoming >> > you could use a 24FXO channel bank cross connected with Rhino T1 >> > cards, or individual FXO gateways but you may have a hard time >> > finding 24 ports of FXO in a single GW. Best performing T1 cards in >> > my experience (thousands of deployments) are Sangoma. Your server >> > configuration looks fine. >> > >> > Cory J. Andrews >> > Director New Market Initiatives >> > >> > Sayers Media Group >> > VoIP Supply, LLC >> > 454 Sonwil Drive >> > Buffalo, NY 14225 >> > 716-250-3402 OFFICE >> > 716-630-1548 FAX >> > 716-601-4474 MOBILE >> > candrews at sayersmedia.com >> > >> > >> > Have I exceeded your expectations? Please share your experience >> > with my boss, Benjamin P. Sayers, CEO >> > >> > NOTICE: The information contained in this email and any document >> > attached hereto is intended only for the named recipient(s). It is >> > the property of the VoIP Supply, LLC and shall not be used, >> > disclosed or reproduced without the express written consent of VoIP >> > Supply, LLC. If you are not the intended recipient, nor the employee >> > or agent responsible for delivering this message in confidence to >> > the intended recipient(s), you are hereby notified that you have >> > received this transmittal in error, and any review, dissemination, >> > distribution or copying of this transmittal or its attachments is >> > strictly prohibited. If you have received this transmittal and/or >> > attachments in error, please notify me immediately by reply e-mail >> > or telephone and then delete this message, including any >> > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY >> > 14225 USA. >> >> > >> > >> > >> > -----Original Message----- >> > From: freeswitch-users-bounces at lists.freeswitch.org >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> > Samuel Mukoti >> > Sent: Wednesday, November 25, 2009 2:40 PM >> > To: freeswitch-users at lists.freeswitch.org >> > Subject: [Freeswitch-users] Grandstream gateways >> > >> > Hi all, >> > >> > I'm wanting to try out a my first large scale setup at the office, 200 >> > extensions and 24 POTS incoming, also a T1 line once the telco guys >> > are ready. I wanted assistance with choosing the most appropriate >> > hardware. We already have about 150 analogue phones, and I was >> > wondering what's best? A couple of grandstream FXS GXW4024? Also for >> > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma >> > or digium card? The best voice quality is paramount. Lastly for T1 >> > what cards are recommeded, >> > >> > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, >> > would that perform? Or do I need hardware transcoding? >> > >> > Thank you, >> > >> > Sam >> > >> > Twitter: twitter.com/samuelmukoti >> > >> > >> > On 25 Nov,2009, at 8:05 PM, >> freeswitch-users-request at lists.freeswitch.org >> > wrote: >> > >> >> Send FreeSWITCH-users mailing list submissions to >> >> freeswitch-users at lists.freeswitch.org >> >> >> >> To subscribe or unsubscribe via the World Wide Web, visit >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> or, via email, send a message with subject or body 'help' to >> >> freeswitch-users-request at lists.freeswitch.org >> >> >> >> You can reach the person managing the list at >> >> freeswitch-users-owner at lists.freeswitch.org >> >> >> >> When replying, please edit your Subject line so it is more specific >> >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> >> >> >> Today's Topics: >> >> >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> >> 2. Re: Handling the 302 Moved Temporarily response from >> >> JavaScript (Michael Jerris) >> >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> >> 5. Re: How to find whether the destination extension supports >> >> encryption (Michael Jerris) >> >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> >> 7. Re: Handling the 302 Moved Temporarily response from >> >> JavaScript (Stephen Crosby) >> >> 8. Re: Handling the 302 Moved Temporarily response from >> >> JavaScript (Tihomir Culjaga) >> >> >> >> >> >> --- >> >> ------------------------------------------------------------------- >> >> >> >> Message: 1 >> >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> >> From: Michael Jerris >> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> >> invitations >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> >> >> Content-Type: text/plain; charset="windows-1252" >> >> >> >> Its a feature we don't have, patches welcome. >> >> >> >> Mike >> >> >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >> >> >>> Hi members, >> >>> I?m controlling freeswitch with the conference module via xmlrpc. >> >>> >> >>> Is it desired that the kick command can only kick users that are >> >>> connected to the conference? >> >>> Is there no chance abort an invitation? >> >>> The kick command has no effect until the person I invited with the >> >>> dial command is connected. >> >> >> >> -------------- next part -------------- >> >> An HTML attachment was scrubbed... >> >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html >> >> >> >> ------------------------------ >> >> >> >> Message: 2 >> >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> >> From: Michael Jerris >> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> >> response from JavaScript >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: >> >> Content-Type: text/plain; charset=us-ascii >> >> >> >> In trunk there is a sofia profile setting to allow dialplan >> >> processing of 302 responses. This won't get you back into your same >> >> javascript, but you can probably do something clever from there. >> >> >> >> Mike >> >> >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >> >> >>> >> >>> I have considered writing JavaScript code to bridge two calls >> >>> together. However, I would like to perform custom handling of the >> >>> 302 Moved Temporarily response. How do I handle the 302 Moved >> >>> Temporarily response if I use JavaScript? >> >>> >> >> >> >> >> >> >> >> ------------------------------ >> >> >> >> Message: 3 >> >> Date: Wed, 25 Nov 2009 11:46:05 -0600 >> >> From: Brian West >> >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> >> proxy. >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> >> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes >> >> >> >> Yes an alias will be required for every domain you run on the profile >> >> so it can find it. >> >> >> >> /b >> >> >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >> >> >>> Try an alias on the sip profile. >> >>> >> >>> Mike >> >> >> >> >> >> >> >> >> >> ------------------------------ >> >> >> >> Message: 4 >> >> Date: Wed, 25 Nov 2009 12:47:37 -0500 >> >> From: Michael Jerris >> >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: >> >> Content-Type: text/plain; charset=us-ascii >> >> >> >> It's possible it does not. I just added some code to set it on auto- >> >> adjust so it might be there sometimes now. You might need to add >> >> some code in mod_sofia to add it other times. Maybe it makes sense >> >> to move that var setting down to switch_rtp.c. Patches for this >> >> would be welcome. >> >> >> >> Thanks >> >> >> >> Mike >> >> >> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: >> >> >> >>> Hi, >> >>> >> >>> In the case of proxy_media=true, does it gets set at all then? >> >> >> >> >> >> >> >> >> >> ------------------------------ >> >> >> >> Message: 5 >> >> Date: Wed, 25 Nov 2009 12:48:39 -0500 >> >> From: Michael Jerris >> >> Subject: Re: [Freeswitch-users] How to find whether the destination >> >> extension supports encryption >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> >> >> Content-Type: text/plain; charset=us-ascii >> >> >> >> You can send the call with secure enabled and if it supports it it >> >> will use it. >> >> >> >> Mike >> >> >> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: >> >> >> >>> Hello, >> >>> >> >>> We have a mix of phones that support RTP encryption and those that >> >>> do not. I have to support both types in the meanwhile, and would >> >>> like to have encryption enabled on the relevant leg, even if the >> >>> other leg does not support it (why? one of our ATAs either must >> >>> have it unencrypted or have it encrypted, but cannot have both). >> >>> >> >>> How do I find whether the destination supports encryption? I do not >> >>> want to manage an additional table in the database... >> >>> >> >> >> >> >> >> >> >> ------------------------------ >> >> >> >> Message: 6 >> >> Date: Wed, 25 Nov 2009 23:25:01 +0530 >> >> From: srinivasula reddy >> >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: >> >> >> >> Content-Type: text/plain; charset="iso-8859-1" >> >> >> >> HI, >> >> thanks for your reply, my requirement is i am doing failover stuff >> >> with >> >> freeswitch. i dont want cut the calls when freeswitch dies, when >> >> failover >> >> happens mean one freeswitch dies we are going to start the second >> >> freeswitch, i dont want close call intiated by the first >> >> freeswtich, they >> >> are communicating with meida(bypass media). when one endpoing try to >> >> end the >> >> call at that time i want to close the call for the other end also. >> >> >> >> >> >> srinivas >> >> >> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris >> >> wrote: >> >> >> >>> FreeSWITCH will kill the calls when you shut it down, if you >> >>> intentionally >> >>> kill the network without shutting down FreeSWITCH the only thing >> >>> you can do >> >>> is enable session timers or rtp timers in the soft phones to kill >> >>> the call >> >>> when FreeSWITCH dies or when the call is over. >> >>> >> >>> Mike >> >>> >> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >> >>> >> >>>> Hi All, >> >>>> >> >>>> goodmorning to all, i have a scenario, two pjsua clients are >> >>>> connected >> >>> with Freeswitch and they are in call and bypass_media=true. i >> >>> close the >> >>> Freeswitch server, still they are in call, again i started the >> >>> Freeswitch, >> >>> and registerd these two endpoints, now how can i end the call >> >>> (estabilished >> >>> by the first Freeswitch)? if i call re_invite will it estabilish >> >>> the call >> >>> between two endpoints? >> >>>> any idea? >> >>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>> users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> >> >> >> >> -- >> >> Srinivasula Reddy K >> >> -------------- next part -------------- >> >> An HTML attachment was scrubbed... >> >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html >> >> >> >> ------------------------------ >> >> >> >> Message: 7 >> >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> >> From: Stephen Crosby >> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> >> response from JavaScript >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: >> >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> >> >> Content-Type: text/plain; charset="utf-8" >> >> >> >> Surprisingly, I've found no way to access the HTTP response status >> >> code >> >> using mod_spidermonkey_curl. I'd love to see this feature added or >> >> discussed >> >> if it already exists and I'm missing it. >> >> >> >> --Stephen >> >> >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris >> >> wrote: >> >> >> >>> In trunk there is a sofia profile setting to allow dialplan >> >>> processing of >> >>> 302 responses. This won't get you back into your same javascript, >> >>> but you >> >>> can probably do something clever from there. >> >>> >> >>> Mike >> >>> >> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >> >>>> >> >>>> I have considered writing JavaScript code to bridge two calls >> >>>> together. >> >>> However, I would like to perform custom handling of the 302 Moved >> >>> Temporarily response. How do I handle the 302 Moved Temporarily >> >>> response if >> >>> I use JavaScript? >> >>>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>> users >> >>> http://www.freeswitch.org >> >>> >> >> -------------- next part -------------- >> >> An HTML attachment was scrubbed... >> >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html >> >> >> >> ------------------------------ >> >> >> >> Message: 8 >> >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> >> From: Tihomir Culjaga >> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> >> response from JavaScript >> >> To: freeswitch-users at lists.freeswitch.org >> >> Message-ID: >> >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> >> >> Content-Type: text/plain; charset="iso-8859-1" >> >> >> >> this is how i do it from the dialplan: >> >> >> >> >> >> >> >> >> >> >> >> > >> expression="^(300030)(.*)|^\+(300030)(.*)"> >> >> >> >> >> >> >> >> >> >> > >> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> >> >> > >> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: >> >> 1:32} : >> >> ${caller_id_number})}"/> >> >> >> >> > >> data="aPfx=${caller_id_number:0:6}"/> >> >> > >> data="aNum=${caller_id_number:6:16}"/> >> >> > >> data="IP_ADDR=${network_addr}:5060"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris >> >> wrote: >> >> >> >>> In trunk there is a sofia profile setting to allow dialplan >> >>> processing of >> >>> 302 responses. This won't get you back into your same javascript, >> >>> but you >> >>> can probably do something clever from there. >> >>> >> >>> Mike >> >>> >> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >> >>>> >> >>>> I have considered writing JavaScript code to bridge two calls >> >>>> together. >> >>> However, I would like to perform custom handling of the 302 Moved >> >>> Temporarily response. How do I handle the 302 Moved Temporarily >> >>> response if >> >>> I use JavaScript? >> >>>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>> users >> >>> http://www.freeswitch.org >> >>> >> >> -------------- next part -------------- >> >> An HTML attachment was scrubbed... >> >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html >> >> >> >> ------------------------------ >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >> users >> >> http://www.freeswitch.org >> >> >> >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 >> >> ************************************************* >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > users >> > http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/368005c3/attachment-0002.html From mrene_lists at avgs.ca Wed Nov 25 13:01:27 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 25 Nov 2009 16:01:27 -0500 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: You can read all about the sip library at http://sofia-sip.sourceforge.net/refdocs/ Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 25-Nov-09, at 3:58 PM, srinivasula reddy wrote: > thanks for your reply mike, > is there any api in freeswitch or any thing else to update lib > programatically from pjsua. > > srinivas > > On Thu, Nov 26, 2009 at 2:05 AM, Michael Jerris > wrote: > "something that is not available in that lib at this time." > > Mike > > On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote > >> can please tell me how can i exchange session state into sip library. >> >> Thanks >> srinivas >> >> On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris >> wrote: >> For that you would need to fully exchange session state into the >> sip library, something that is not available in that lib at this >> time. >> >> >> On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: >> >>> HI, >>> thanks for your reply, my requirement is i am doing failover stuff >>> with freeswitch. i dont want cut the calls when freeswitch dies, >>> when failover happens mean one freeswitch dies we are going to >>> start the second freeswitch, i dont want close call intiated by >>> the first freeswtich, they are communicating with meida(bypass >>> media). when one endpoing try to end the call at that time i want >>> to close the call for the other end also. >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Srinivasula Reddy K >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ce968534/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 25 13:10:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 15:10:57 -0600 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: References: Message-ID: <191c3a030911251310h9f8bd1epf0d445c746e968a5@mail.gmail.com> I can spare you the pain and let you know outright that this sort of functionality will cost somewhere in the range of 125,000.00 to 150,000.00 to properly implement by assembling a team of consultants including members of the development team from both FreeSWITCH and Sofia-SIP and even if you have the money, finding the time to implement it would also be a factor as it's a few thousand man-hours of work. On Wed, Nov 25, 2009 at 3:01 PM, Mathieu Rene wrote: > You can read all about the sip library at > http://sofia-sip.sourceforge.net/refdocs/ > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 25-Nov-09, at 3:58 PM, srinivasula reddy wrote: > > thanks for your reply mike, > is there any api in freeswitch or any thing else to update lib > programatically from pjsua. > > srinivas > > On Thu, Nov 26, 2009 at 2:05 AM, Michael Jerris wrote: > >> "something that is not available in that lib at this time." >> >> Mike >> >> On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote >> >> can please tell me how can i exchange session state into sip library. >> >> Thanks >> srinivas >> >> On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: >> >>> For that you would need to fully exchange session state into the sip >>> library, *something that is not available in that lib at this time.* >>> >>> >>> On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: >>> >>> HI, >>> thanks for your reply, my requirement is i am doing failover stuff with >>> freeswitch. i dont want cut the calls when freeswitch dies, when failover >>> happens mean one freeswitch dies we are going to start the second >>> freeswitch, i dont want close call intiated by the first freeswtich, they >>> are communicating with meida(bypass media). when one endpoing try to end the >>> call at that time i want to close the call for the other end also. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Srinivasula Reddy K >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Srinivasula Reddy K > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/93b12dba/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 25 13:19:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 15:19:56 -0600 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> Message-ID: <191c3a030911251319g60cdd5a3t33a82a560faf7a2b@mail.gmail.com> The processor power saved is negligible between PCMU and raw PCM and not worth the fuss. If you didn't decode the audio first you would not be able to mix the stream to produce a single file. So if we went to the trouble of making native media bugs to be able to do that you could barely use them so it would not be worth the 5k or more bounty to develop that functionality. On Sun, Nov 22, 2009 at 12:48 PM, Matthew Fong wrote: > I'm trying to conserve processor power by recording in native file format, > PCMU in my case. It works great with the following line > > session:execute("record", > "/tmp/my_recording."..session:getVariable("read_codec")); > > however it fails to work with > > session:execute("record_session", > "/tmp/my_recording."..session:getVariable("read_codec")); > or > record = api:execute("sched_api", '+1 none uuid_record > '..session:getVariable("uuid")..' start > /tmp/my_recording.'..session:getVariable("read_codec")); > > Why is it that it works with record, but not with record_session or > uuid_record? Is there something I'm over looking? In the latter two the > consul reports > > 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File > [/tmp/my_recording.PCMU] 8000hz > > as if it's recording, but /tmp/my_recording.PCMU never shows up. However if > I change it to .wav instead of .PCMU it works. Any ideas? > > --matt > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/a49b5d12/attachment-0002.html From lists at redbonez.net Wed Nov 25 14:18:34 2009 From: lists at redbonez.net (Adam Ford) Date: Wed, 25 Nov 2009 15:18:34 -0700 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> Message-ID: <01cb01ca6e1d$3c289540$b479bfc0$@net> Samuel, FreeSWITCH has a Skype module that uses Skype client instances to connect to the Skype network, you can read about it at http://wiki.freeswitch.org/wiki/Skypiax As far as an official Skype module for non-Asterisk PBX-es, it looks like it is in beta right now - http://www.skype.com/business/products/pbx-systems/sip/ -AF -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Samuel Mukoti Sent: Wednesday, November 25, 2009 1:17 PM Cc: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Grandstream gateways Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews at sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org > wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ 288d63a0/attachment-0001.html >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> In trunk there is a sofia profile setting to allow dialplan >> processing of 302 responses. This won't get you back into your same >> javascript, but you can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. However, I would like to perform custom handling of the >>> 302 Moved Temporarily response. How do I handle the 302 Moved >>> Temporarily response if I use JavaScript? >>> >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Wed, 25 Nov 2009 11:46:05 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> proxy. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes >> >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >>> Try an alias on the sip profile. >>> >>> Mike >> >> >> >> >> ------------------------------ >> >> Message: 4 >> Date: Wed, 25 Nov 2009 12:47:37 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> It's possible it does not. I just added some code to set it on auto- >> adjust so it might be there sometimes now. You might need to add >> some code in mod_sofia to add it other times. Maybe it makes sense >> to move that var setting down to switch_rtp.c. Patches for this >> would be welcome. >> >> Thanks >> >> Mike >> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: >> >>> Hi, >>> >>> In the case of proxy_media=true, does it gets set at all then? >> >> >> >> >> ------------------------------ >> >> Message: 5 >> Date: Wed, 25 Nov 2009 12:48:39 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] How to find whether the destination >> extension supports encryption >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> >> Content-Type: text/plain; charset=us-ascii >> >> You can send the call with secure enabled and if it supports it it >> will use it. >> >> Mike >> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: >> >>> Hello, >>> >>> We have a mix of phones that support RTP encryption and those that >>> do not. I have to support both types in the meanwhile, and would >>> like to have encryption enabled on the relevant leg, even if the >>> other leg does not support it (why? one of our ATAs either must >>> have it unencrypted or have it encrypted, but cannot have both). >>> >>> How do I find whether the destination supports encryption? I do not >>> want to manage an additional table in the database... >>> >> >> >> >> ------------------------------ >> >> Message: 6 >> Date: Wed, 25 Nov 2009 23:25:01 +0530 >> From: srinivasula reddy >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> >> Content-Type: text/plain; charset="iso-8859-1" >> >> HI, >> thanks for your reply, my requirement is i am doing failover stuff >> with >> freeswitch. i dont want cut the calls when freeswitch dies, when >> failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the first >> freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to >> end the >> call at that time i want to close the call for the other end also. >> >> >> srinivas >> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris >> wrote: >> >>> FreeSWITCH will kill the calls when you shut it down, if you >>> intentionally >>> kill the network without shutting down FreeSWITCH the only thing >>> you can do >>> is enable session timers or rtp timers in the soft phones to kill >>> the call >>> when FreeSWITCH dies or when the call is over. >>> >>> Mike >>> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >>> >>>> Hi All, >>>> >>>> goodmorning to all, i have a scenario, two pjsua clients are >>>> connected >>> with Freeswitch and they are in call and bypass_media=true. i >>> close the >>> Freeswitch server, still they are in call, again i started the >>> Freeswitch, >>> and registerd these two endpoints, now how can i end the call >>> (estabilished >>> by the first Freeswitch)? if i call re_invite will it estabilish >>> the call >>> between two endpoints? >>>> any idea? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Srinivasula Reddy K >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ ec246f47/attachment-0001.html >> >> ------------------------------ >> >> Message: 7 >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> From: Stephen Crosby >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Surprisingly, I've found no way to access the HTTP response status >> code >> using mod_spidermonkey_curl. I'd love to see this feature added or >> discussed >> if it already exists and I'm missing it. >> >> --Stephen >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ b8ea2be6/attachment-0001.html >> >> ------------------------------ >> >> Message: 8 >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> From: Tihomir Culjaga >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> this is how i do it from the dialplan: >> >> >> >> >> >> > expression="^(300030)(.*)|^\+(300030)(.*)"> >> >> >> >> >> > data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> >> > data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: >> 1:32} : >> ${caller_id_number})}"/> >> >> > data="aPfx=${caller_id_number:0:6}"/> >> > data="aNum=${caller_id_number:6:16}"/> >> > data="IP_ADDR=${network_addr}:5060"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ 638a2202/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 >> ************************************************* > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mike at jerris.com Wed Nov 25 14:38:18 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 17:38:18 -0500 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <01cb01ca6e1d$3c289540$b479bfc0$@net> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <01cb01ca6e1d$3c289540$b479bfc0$@net> Message-ID: <2ABF7CA4-5FBF-4E7D-8BE3-6D0C92717C92@jerris.com> On Nov 25, 2009, at 5:18 PM, Adam Ford wrote: > Samuel, > > FreeSWITCH has a Skype module that uses Skype client instances to connect to > the Skype network, you can read about it at > http://wiki.freeswitch.org/wiki/Skypiax > > As far as an official Skype module for non-Asterisk PBX-es, it looks like it > is in beta right now - > http://www.skype.com/business/products/pbx-systems/sip/ > > -AF If by in beta you mean they turned off all the servers the beta testers could talk to, then yes, it is indeed. Mike From anthony.minessale at gmail.com Wed Nov 25 14:40:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 16:40:45 -0600 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <01cb01ca6e1d$3c289540$b479bfc0$@net> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <01cb01ca6e1d$3c289540$b479bfc0$@net> Message-ID: <191c3a030911251440n46ec3ee8h92d1305b1542fc0@mail.gmail.com> Skype for SIP is just a SIP account you can get from skype that is somehow tied to skype, probably using the scary 2 year commerical endeavor to make skype work in asterisk. We should all thank Giovanni Maruzzelli for giving us a free solution. On Wed, Nov 25, 2009 at 4:18 PM, Adam Ford wrote: > Samuel, > > FreeSWITCH has a Skype module that uses Skype client instances to connect > to > the Skype network, you can read about it at > http://wiki.freeswitch.org/wiki/Skypiax > > As far as an official Skype module for non-Asterisk PBX-es, it looks like > it > is in beta right now - > http://www.skype.com/business/products/pbx-systems/sip/ > > -AF > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Samuel > Mukoti > Sent: Wednesday, November 25, 2009 1:17 PM > Cc: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Grandstream gateways > > Thank you for those tips, > > I do have some small setups using gxw4108 they work or, except CID > doesn't seem to work. I will try the channel bank route - just don't > know too much about the setup options or how you'd purchase the > correct config, eg. For 150 FXS channel bank, can I get a single PCI > card for that? > > I may end up using the grandstream fxs gateways then use the T1 > channel bank from sangoma, > > Thank you all.. > > Lastly, I know asterisk now has an offical skype_ module, Is there > anything similar I could use? > > > On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: > > > Samuel - you could go with FXS gateways or channel banks. If you go > > the gateway route Grandstream or Audiocodes would work fine. > > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > > you could use a 24FXO channel bank cross connected with Rhino T1 > > cards, or individual FXO gateways but you may have a hard time > > finding 24 ports of FXO in a single GW. Best performing T1 cards in > > my experience (thousands of deployments) are Sangoma. Your server > > configuration looks fine. > > > > Cory J. Andrews > > Director New Market Initiatives > > > > Sayers Media Group > > VoIP Supply, LLC > > 454 Sonwil Drive > > Buffalo, NY 14225 > > 716-250-3402 OFFICE > > 716-630-1548 FAX > > 716-601-4474 MOBILE > > candrews at sayersmedia.com > > > > > > Have I exceeded your expectations? Please share your experience > > with my boss, Benjamin P. Sayers, CEO > > > > NOTICE: The information contained in this email and any document > > attached hereto is intended only for the named recipient(s). It is > > the property of the VoIP Supply, LLC and shall not be used, > > disclosed or reproduced without the express written consent of VoIP > > Supply, LLC. If you are not the intended recipient, nor the employee > > or agent responsible for delivering this message in confidence to > > the intended recipient(s), you are hereby notified that you have > > received this transmittal in error, and any review, dissemination, > > distribution or copying of this transmittal or its attachments is > > strictly prohibited. If you have received this transmittal and/or > > attachments in error, please notify me immediately by reply e-mail > > or telephone and then delete this message, including any > > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > > 14225 USA. > > > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Samuel Mukoti > > Sent: Wednesday, November 25, 2009 2:40 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Grandstream gateways > > > > Hi all, > > > > I'm wanting to try out a my first large scale setup at the office, 200 > > extensions and 24 POTS incoming, also a T1 line once the telco guys > > are ready. I wanted assistance with choosing the most appropriate > > hardware. We already have about 150 analogue phones, and I was > > wondering what's best? A couple of grandstream FXS GXW4024? Also for > > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > > or digium card? The best voice quality is paramount. Lastly for T1 > > what cards are recommeded, > > > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > > would that perform? Or do I need hardware transcoding? > > > > Thank you, > > > > Sam > > > > Twitter: twitter.com/samuelmukoti > > > > > > On 25 Nov,2009, at 8:05 PM, > freeswitch-users-request at lists.freeswitch.org > > wrote: > > > >> Send FreeSWITCH-users mailing list submissions to > >> freeswitch-users at lists.freeswitch.org > >> > >> To subscribe or unsubscribe via the World Wide Web, visit > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> or, via email, send a message with subject or body 'help' to > >> freeswitch-users-request at lists.freeswitch.org > >> > >> You can reach the person managing the list at > >> freeswitch-users-owner at lists.freeswitch.org > >> > >> When replying, please edit your Subject line so it is more specific > >> than "Re: Contents of FreeSWITCH-users digest..." > >> > >> > >> Today's Topics: > >> > >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) > >> 2. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Michael Jerris) > >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) > >> 4. Re: remote_media_ip variable not set (Michael Jerris) > >> 5. Re: How to find whether the destination extension supports > >> encryption (Michael Jerris) > >> 6. Re: Bypass_media and re_invite (srinivasula reddy) > >> 7. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Stephen Crosby) > >> 8. Re: Handling the 302 Moved Temporarily response from > >> JavaScript (Tihomir Culjaga) > >> > >> > >> --- > >> ------------------------------------------------------------------- > >> > >> Message: 1 > >> Date: Wed, 25 Nov 2009 12:44:46 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] mod_conference kick to abort > >> invitations > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> > >> Content-Type: text/plain; charset="windows-1252" > >> > >> Its a feature we don't have, patches welcome. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > >> > >>> Hi members, > >>> I?m controlling freeswitch with the conference module via xmlrpc. > >>> > >>> Is it desired that the kick command can only kick users that are > >>> connected to the conference? > >>> Is there no chance abort an invitation? > >>> The kick command has no effect until the person I invited with the > >>> dial command is connected. > >> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ > 288d63a0/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 2 > >> Date: Wed, 25 Nov 2009 12:45:50 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii > >> > >> In trunk there is a sofia profile setting to allow dialplan > >> processing of 302 responses. This won't get you back into your same > >> javascript, but you can probably do something clever from there. > >> > >> Mike > >> > >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> > >>> > >>> I have considered writing JavaScript code to bridge two calls > >>> together. However, I would like to perform custom handling of the > >>> 302 Moved Temporarily response. How do I handle the 302 Moved > >>> Temporarily response if I use JavaScript? > >>> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 3 > >> Date: Wed, 25 Nov 2009 11:46:05 -0600 > >> From: Brian West > >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via > >> proxy. > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> > >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes > >> > >> Yes an alias will be required for every domain you run on the profile > >> so it can find it. > >> > >> /b > >> > >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > >> > >>> Try an alias on the sip profile. > >>> > >>> Mike > >> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 4 > >> Date: Wed, 25 Nov 2009 12:47:37 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii > >> > >> It's possible it does not. I just added some code to set it on auto- > >> adjust so it might be there sometimes now. You might need to add > >> some code in mod_sofia to add it other times. Maybe it makes sense > >> to move that var setting down to switch_rtp.c. Patches for this > >> would be welcome. > >> > >> Thanks > >> > >> Mike > >> > >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > >> > >>> Hi, > >>> > >>> In the case of proxy_media=true, does it gets set at all then? > >> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 5 > >> Date: Wed, 25 Nov 2009 12:48:39 -0500 > >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] How to find whether the destination > >> extension supports encryption > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> > >> Content-Type: text/plain; charset=us-ascii > >> > >> You can send the call with secure enabled and if it supports it it > >> will use it. > >> > >> Mike > >> > >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > >> > >>> Hello, > >>> > >>> We have a mix of phones that support RTP encryption and those that > >>> do not. I have to support both types in the meanwhile, and would > >>> like to have encryption enabled on the relevant leg, even if the > >>> other leg does not support it (why? one of our ATAs either must > >>> have it unencrypted or have it encrypted, but cannot have both). > >>> > >>> How do I find whether the destination supports encryption? I do not > >>> want to manage an additional table in the database... > >>> > >> > >> > >> > >> ------------------------------ > >> > >> Message: 6 > >> Date: Wed, 25 Nov 2009 23:25:01 +0530 > >> From: srinivasula reddy > >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> > >> Content-Type: text/plain; charset="iso-8859-1" > >> > >> HI, > >> thanks for your reply, my requirement is i am doing failover stuff > >> with > >> freeswitch. i dont want cut the calls when freeswitch dies, when > >> failover > >> happens mean one freeswitch dies we are going to start the second > >> freeswitch, i dont want close call intiated by the first > >> freeswtich, they > >> are communicating with meida(bypass media). when one endpoing try to > >> end the > >> call at that time i want to close the call for the other end also. > >> > >> > >> srinivas > >> > >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris > >> wrote: > >> > >>> FreeSWITCH will kill the calls when you shut it down, if you > >>> intentionally > >>> kill the network without shutting down FreeSWITCH the only thing > >>> you can do > >>> is enable session timers or rtp timers in the soft phones to kill > >>> the call > >>> when FreeSWITCH dies or when the call is over. > >>> > >>> Mike > >>> > >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > >>> > >>>> Hi All, > >>>> > >>>> goodmorning to all, i have a scenario, two pjsua clients are > >>>> connected > >>> with Freeswitch and they are in call and bypass_media=true. i > >>> close the > >>> Freeswitch server, still they are in call, again i started the > >>> Freeswitch, > >>> and registerd these two endpoints, now how can i end the call > >>> (estabilished > >>> by the first Freeswitch)? if i call re_invite will it estabilish > >>> the call > >>> between two endpoints? > >>>> any idea? > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Srinivasula Reddy K > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ > ec246f47/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 7 > >> Date: Wed, 25 Nov 2009 10:01:14 -0800 > >> From: Stephen Crosby > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> > >> Content-Type: text/plain; charset="utf-8" > >> > >> Surprisingly, I've found no way to access the HTTP response status > >> code > >> using mod_spidermonkey_curl. I'd love to see this feature added or > >> discussed > >> if it already exists and I'm missing it. > >> > >> --Stephen > >> > >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris > >> wrote: > >> > >>> In trunk there is a sofia profile setting to allow dialplan > >>> processing of > >>> 302 responses. This won't get you back into your same javascript, > >>> but you > >>> can probably do something clever from there. > >>> > >>> Mike > >>> > >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >>> > >>>> > >>>> I have considered writing JavaScript code to bridge two calls > >>>> together. > >>> However, I would like to perform custom handling of the 302 Moved > >>> Temporarily response. How do I handle the 302 Moved Temporarily > >>> response if > >>> I use JavaScript? > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ > b8ea2be6/attachment-0001.html > >> > >> ------------------------------ > >> > >> Message: 8 > >> Date: Wed, 25 Nov 2009 19:04:56 +0100 > >> From: Tihomir Culjaga > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily > >> response from JavaScript > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> > >> Content-Type: text/plain; charset="iso-8859-1" > >> > >> this is how i do it from the dialplan: > >> > >> > >> > >> > >> > >> >> expression="^(300030)(.*)|^\+(300030)(.*)"> > >> > >> > >> > >> > >> >> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> > >> >> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: > >> 1:32} : > >> ${caller_id_number})}"/> > >> > >> >> data="aPfx=${caller_id_number:0:6}"/> > >> >> data="aNum=${caller_id_number:6:16}"/> > >> >> data="IP_ADDR=${network_addr}:5060"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris > >> wrote: > >> > >>> In trunk there is a sofia profile setting to allow dialplan > >>> processing of > >>> 302 responses. This won't get you back into your same javascript, > >>> but you > >>> can probably do something clever from there. > >>> > >>> Mike > >>> > >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >>> > >>>> > >>>> I have considered writing JavaScript code to bridge two calls > >>>> together. > >>> However, I would like to perform custom handling of the 302 Moved > >>> Temporarily response. How do I handle the 302 Moved Temporarily > >>> response if > >>> I use JavaScript? > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ > 638a2202/attachment.html > >> > >> ------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > >> > >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 > >> ************************************************* > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/7800dfaa/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 25 14:58:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 16:58:17 -0600 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <2ABF7CA4-5FBF-4E7D-8BE3-6D0C92717C92@jerris.com> References: <270A2C12-D937-4C5B-BCE9-B175790BEDBA@gmail.com> <01cb01ca6e1d$3c289540$b479bfc0$@net> <2ABF7CA4-5FBF-4E7D-8BE3-6D0C92717C92@jerris.com> Message-ID: <191c3a030911251458t5475dfc9lb9665878de91b8c9@mail.gmail.com> exactly On Wed, Nov 25, 2009 at 4:38 PM, Michael Jerris wrote: > > On Nov 25, 2009, at 5:18 PM, Adam Ford wrote: > > > Samuel, > > > > FreeSWITCH has a Skype module that uses Skype client instances to connect > to > > the Skype network, you can read about it at > > http://wiki.freeswitch.org/wiki/Skypiax > > > > As far as an official Skype module for non-Asterisk PBX-es, it looks like > it > > is in beta right now - > > http://www.skype.com/business/products/pbx-systems/sip/ > > > > -AF > > If by in beta you mean they turned off all the servers the beta testers > could talk to, then yes, it is indeed. > > Mike > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/12c9df3f/attachment-0002.html From anthony.minessale at gmail.com Wed Nov 25 15:10:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Nov 2009 17:10:04 -0600 Subject: [Freeswitch-users] remote_media_ip variable not set In-Reply-To: References: <27c25bc40911240722vfe90d0dr497ceec9f03bfecf@mail.gmail.com> <2F929FDB-0E1B-49E0-A1E7-F4F1E2D548AD@avgs.ca> <27c25bc40911240756k7842c80kd75be2d3d93441b9@mail.gmail.com> Message-ID: <191c3a030911251510s6dad4526i7cc6e64925b8153a@mail.gmail.com> I added a patch to do it in more places On Wed, Nov 25, 2009 at 11:47 AM, Michael Jerris wrote: > It's possible it does not. I just added some code to set it on auto-adjust > so it might be there sometimes now. You might need to add some code in > mod_sofia to add it other times. Maybe it makes sense to move that var > setting down to switch_rtp.c. Patches for this would be welcome. > > Thanks > > Mike > > On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > > > Hi, > > > > In the case of proxy_media=true, does it gets set at all then? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/85b9b0aa/attachment-0002.html From john_platts at hotmail.com Wed Nov 25 15:21:51 2009 From: john_platts at hotmail.com (John Platts) Date: Wed, 25 Nov 2009 17:21:51 -0600 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: , Message-ID: How do I turn on dialplan processing of 302 responses? I can solve my problem if I can process 302 responses in my dialplan. ---------------------------------------- > From: mike at jerris.com > Date: Wed, 25 Nov 2009 12:45:50 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript > > In trunk there is a sofia profile setting to allow dialplan processing of 302 responses. This won't get you back into your same javascript, but you can probably do something clever from there. > > Mike > > On Nov 24, 2009, at 5:04 PM, John Platts wrote: > >> >> I have considered writing JavaScript code to bridge two calls together. However, I would like to perform custom handling of the 302 Moved Temporarily response. How do I handle the 302 Moved Temporarily response if I use JavaScript? >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search?q=restaurants&form=MFESRP&publ=WLHMTAG&crea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 From jmesquita at freeswitch.org Wed Nov 25 17:02:54 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 25 Nov 2009 23:02:54 -0200 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: <8595daf70911250929w26eeb3aboae0f95042f35393b@mail.gmail.com> References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> <8595daf70911250929w26eeb3aboae0f95042f35393b@mail.gmail.com> Message-ID: These guys can on E1, not T1. They are not compatible with FS just yet, but we are working on it. Let me know off-list if you are interested. JM On Wed, Nov 25, 2009 at 3:29 PM, Imthiyaz Ahmed wrote: > I mean to tap tx and rx of a PRI line using sangoma tap and record > the call information and actual calls without distrubing the existing > line . freeswitch will work in passive mode like trunk side call > recorder. > > Thanks > Imthiyaz > > > On Wed, Nov 25, 2009 at 10:43 PM, Anthony Minessale > wrote: > > What do you mean by passive encoding? > > > > On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed > > wrote: > >> > >> hi > >> is it possibe to enable passive recording in sangoma tdm interface > >> in feeswich. pls advice > >> Best Regards > >> G.Imthiyaz Ahmed > >> > >> > >> > >> -- > >> Best Regards > >> G.Imthiyaz Ahmed > >> PeopleTech systems (P) ltd > >> http://peopletech.co.in > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Best Regards > G.Imthiyaz Ahmed > PeopleTech systems (P) ltd > http://peopletech.co.in > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/13422ae4/attachment-0002.html From dujinfang at gmail.com Wed Nov 25 17:35:08 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 26 Nov 2009 09:35:08 +0800 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <191c3a030911251319g60cdd5a3t33a82a560faf7a2b@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <191c3a030911251319g60cdd5a3t33a82a560faf7a2b@mail.gmail.com> Message-ID: <23f91030911251735r3215a344h279a3f8589d5ff85@mail.gmail.com> Yeah, that's why I had to record to two files(read&write) and need to mix together by using sox. Do you only try to using PCMU to save CPU power matt? As Anthony said, the difference can be ignored. And you also need to take extra effort to make sure transcoding will not happen on a conversation. But it maybe useful for expensive codecs like g729, iLBC, speex etc for recording heavy scenarios. I'd like to take a look if there is a 5k bounty ;) 2009/11/26 Anthony Minessale > The processor power saved is negligible between PCMU and raw PCM and not > worth the fuss. > If you didn't decode the audio first you would not be able to mix the > stream to produce a single file. > So if we went to the trouble of making native media bugs to be able to do > that you could barely use them so it would not be worth the 5k or more > bounty to develop that functionality. > > > > On Sun, Nov 22, 2009 at 12:48 PM, Matthew Fong wrote: > >> I'm trying to conserve processor power by recording in native file format, >> PCMU in my case. It works great with the following line >> >> session:execute("record", >> "/tmp/my_recording."..session:getVariable("read_codec")); >> >> however it fails to work with >> >> session:execute("record_session", >> "/tmp/my_recording."..session:getVariable("read_codec")); >> or >> record = api:execute("sched_api", '+1 none uuid_record >> '..session:getVariable("uuid")..' start >> /tmp/my_recording.'..session:getVariable("read_codec")); >> >> Why is it that it works with record, but not with record_session or >> uuid_record? Is there something I'm over looking? In the latter two the >> consul reports >> >> 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File >> [/tmp/my_recording.PCMU] 8000hz >> >> as if it's recording, but /tmp/my_recording.PCMU never shows up. However >> if I change it to .wav instead of .PCMU it works. Any ideas? >> >> --matt >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/52f891bf/attachment-0002.html From josh at radianttiger.com Wed Nov 25 17:38:12 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 25 Nov 2009 17:38:12 -0800 Subject: [Freeswitch-users] ESL command completion Message-ID: Is there a way of determining if a call-command sent to a session via ESL has completed? Is there a return event which is always fired? Is there a identifier I can use to verify that the return event matches my command? Thanks, Josh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ccda899c/attachment-0002.html From mike at jerris.com Wed Nov 25 18:08:59 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 21:08:59 -0500 Subject: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript In-Reply-To: References: , Message-ID: <5AB26AE2-2ABC-4D46-B61A-675B1390553F@jerris.com> from http://svn.freeswitch.org/svn/freeswitch/trunk/conf/sip_profiles/internal.xml It appears this never made the wiki, could someone please get it on there. Thanks Mike On Nov 25, 2009, at 6:21 PM, John Platts wrote: > > How do I turn on dialplan processing of 302 responses? I can solve my problem if I can process 302 responses in my dialplan. > > ---------------------------------------- >> From: mike at jerris.com >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript >> >> In trunk there is a sofia profile setting to allow dialplan processing of 302 responses. This won't get you back into your same javascript, but you can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls together. However, I would like to perform custom handling of the 302 Moved Temporarily response. How do I handle the 302 Moved Temporarily response if I use JavaScript? >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________ > Bing brings you maps, menus, and reviews organized in one place. > http://www.bing.com/search?q=restaurants&form=MFESRP&publ=WLHMTAG&crea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From josh at radianttiger.com Wed Nov 25 18:10:16 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 25 Nov 2009 18:10:16 -0800 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> Message-ID: Carlos, Do you have any documentation or scripts for your builds? I'm interested in having a working automated build and installer build process, and I'm curious if there's any work you've done that can make my job easier. :) Thanks, Josh On Wed, Nov 4, 2009 at 6:51 AM, Carlos Talbot wrote: > > I usually try to update the svn file at least once a month. I have a new > version ready that was compiled last night but am ironing out login issues > with the FS dudes for upload access. Also, the SVN snapshot now includes > binaries for 32 and 64 bit. It no longer includes flite though as the > install file was approaching 80MB in size. I will revisit this later if > others feel it important to include flite. > >> >> You mentioned FreePBX V3. I had been fumbling around trying to work out >> what >> this is and from what I've read, it seems to provide a GUI Front End for >> configuring FreeSwitch ? >> > Yes, it's still in development phase and as such not ready for production > use. > >> >> I am guessing that while it has been installed with FreeSwitch, I then >> need >> to run the FreePBX Installer to update the FreePBX/FreeSwitch >> configuration >> on my hardware ? >> >> >> When I start FreeSwitch, it does not automatically load the WAMPServer. >> >> Freeswitch and WAMPServer are independant of each other. WAMPServer is > bundled in this install for the purpose of FreePBX as MySQL, Apache and PHP > are all required components of FreePBX. > > When I start WAMPServer manually, and open up localhost (127.0.0.1) in a >> web >> browser, I can see the WampServer logo and various tools such as phpinfo() >> and phpmyadmin. FreePBX is there under Your Projects. >> >> If you want to configure FreePBX you need to click on the FreePBX.url > shortcut that gets created on your desktop. > > >> When I opened this up the first time, it appeared to want to install >> FreePBX >> over FreeSwitch, I tried to abort this when it was going to overwrite some >> FreeSwitch conf files and I thought I'd better not go on until I had a >> better idea what was happening. I backed out of the FreePBX install and >> now >> I can't get the FreePBX or phpmyadmin pages up again (missing files) so it >> looks like I'm going to have to reinstall anyway. >> >> So, for next time,am I right in thinking that I should proceed with >> running >> the FreePBX install from the WAMPServer menu ? >> > > No, launch it from the shortcut as stated above. Unfortunately, at this > time there is very little user documentation on configuring FreePBX. Here is > the link to the developer's info: http://www.freepbx.org/v3 > > regards, > > Carlos > >> >> >> ----- Original Message ----- >> From: "Jeff Lenk" >> To: >> Sent: Tuesday, November 03, 2009 2:48 PM >> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >> >> >> > >> > Hi Dave, >> > >> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >> > >> > Just as you said freeswitch-1.0.4.exe is the tagged release and >> > freeswitch.exe is a newer svn snapshot. >> > >> > There should be no problems installing the new version allthough best to >> > just try and see! >> > >> > Not sure why the newest one is from October 7th. >> > >> > Jeff >> > >> > >> > Dave Stevenson wrote: >> >> >> >> Hi, >> >> >> >> I have read the Docs on the Wiki >> >> ( >> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >> >> but am still not sure of what the different Windows install files are. >> >> Currently, the Windows Installer directory contains :- >> >> >> >> LATEST_SVN_15106 - 6 Bytes >> >> >> >> freeswitch-1.0.4.exe - 42 Megabytes >> >> >> >> freeswitch.exe - 32 Megabytes >> >> >> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >> >> September. The freeswitch.exe file is dated 7th October and think that >> it >> >> contains the minor updates since 3rd September ? >> >> >> >> Could someone who knows FreeSwitch under windows help me understand the >> >> two files please ? >> >> >> >> I chickened out of running the later exe in case it did something to >> the >> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >> with >> >> the old one already installed ? >> >> What will it actually do ? >> >> >> >> regards >> >> Dave >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > -- >> > View this message in context: >> > >> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/f9ff8d51/attachment-0002.html From mike at jerris.com Wed Nov 25 18:11:31 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Nov 2009 21:11:31 -0500 Subject: [Freeswitch-users] ESL command completion In-Reply-To: References: Message-ID: <34EF9B51-9B0A-45A9-A269-17E49D597BA1@jerris.com> There are execute_complete events. I can't recall everything that is in them but they should always be fired. Mike On Nov 25, 2009, at 8:38 PM, Josh Rivers wrote: > Is there a way of determining if a call-command sent to a session via ESL has completed? Is there a return event which is always fired? Is there a identifier I can use to verify that the return event matches my command? > > Thanks, > Josh > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From carlos.talbot at gmail.com Wed Nov 25 20:37:30 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 25 Nov 2009 22:37:30 -0600 Subject: [Freeswitch-users] Precompiled Windows Binaries In-Reply-To: References: <95571858742E44F1A6B60B81A81673F0@bp1.ad.bp.com> <1257259714704-3938887.post@n2.nabble.com> <5800526b0911040651y7ca575efo2c43610967c27269@mail.gmail.com> Message-ID: <5800526b0911252037l5cf974cbie9542b40316ebf37@mail.gmail.com> I've just checked in the source files for the Inno Setup script I'm using to build the windows installer(svn 15681). That's about the extent of the documentation at this point. :) regards, Carlos On Wed, Nov 25, 2009 at 8:10 PM, Josh Rivers wrote: > Carlos, > > Do you have any documentation or scripts for your builds? I'm interested in > having a working automated build and installer build process, and I'm > curious if there's any work you've done that can make my job easier. :) > > Thanks, > Josh > > > On Wed, Nov 4, 2009 at 6:51 AM, Carlos Talbot wrote: > >> >> I usually try to update the svn file at least once a month. I have a new >> version ready that was compiled last night but am ironing out login issues >> with the FS dudes for upload access. Also, the SVN snapshot now includes >> binaries for 32 and 64 bit. It no longer includes flite though as the >> install file was approaching 80MB in size. I will revisit this later if >> others feel it important to include flite. >> >>> >>> You mentioned FreePBX V3. I had been fumbling around trying to work out >>> what >>> this is and from what I've read, it seems to provide a GUI Front End for >>> configuring FreeSwitch ? >>> >> Yes, it's still in development phase and as such not ready for production >> use. >> >>> >>> I am guessing that while it has been installed with FreeSwitch, I then >>> need >>> to run the FreePBX Installer to update the FreePBX/FreeSwitch >>> configuration >>> on my hardware ? >>> >>> >>> When I start FreeSwitch, it does not automatically load the WAMPServer. >>> >>> Freeswitch and WAMPServer are independant of each other. WAMPServer is >> bundled in this install for the purpose of FreePBX as MySQL, Apache and PHP >> are all required components of FreePBX. >> >> When I start WAMPServer manually, and open up localhost (127.0.0.1) in a >>> web >>> browser, I can see the WampServer logo and various tools such as >>> phpinfo() >>> and phpmyadmin. FreePBX is there under Your Projects. >>> >>> If you want to configure FreePBX you need to click on the FreePBX.url >> shortcut that gets created on your desktop. >> >> >>> When I opened this up the first time, it appeared to want to install >>> FreePBX >>> over FreeSwitch, I tried to abort this when it was going to overwrite >>> some >>> FreeSwitch conf files and I thought I'd better not go on until I had a >>> better idea what was happening. I backed out of the FreePBX install and >>> now >>> I can't get the FreePBX or phpmyadmin pages up again (missing files) so >>> it >>> looks like I'm going to have to reinstall anyway. >>> >>> So, for next time,am I right in thinking that I should proceed with >>> running >>> the FreePBX install from the WAMPServer menu ? >>> >> >> No, launch it from the shortcut as stated above. Unfortunately, at this >> time there is very little user documentation on configuring FreePBX. Here is >> the link to the developer's info: http://www.freepbx.org/v3 >> >> regards, >> >> Carlos >> >>> >>> >>> ----- Original Message ----- >>> From: "Jeff Lenk" >>> To: >>> Sent: Tuesday, November 03, 2009 2:48 PM >>> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries >>> >>> >>> > >>> > Hi Dave, >>> > >>> > These are supported by "Carlos Talbot" . They also include Freepbx v3 >>> > >>> > Just as you said freeswitch-1.0.4.exe is the tagged release and >>> > freeswitch.exe is a newer svn snapshot. >>> > >>> > There should be no problems installing the new version allthough best >>> to >>> > just try and see! >>> > >>> > Not sure why the newest one is from October 7th. >>> > >>> > Jeff >>> > >>> > >>> > Dave Stevenson wrote: >>> >> >>> >> Hi, >>> >> >>> >> I have read the Docs on the Wiki >>> >> ( >>> http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) >>> >> but am still not sure of what the different Windows install files are. >>> >> Currently, the Windows Installer directory contains :- >>> >> >>> >> LATEST_SVN_15106 - 6 Bytes >>> >> >>> >> freeswitch-1.0.4.exe - 42 Megabytes >>> >> >>> >> freeswitch.exe - 32 Megabytes >>> >> >>> >> I have installed the freeswitch-1.0.4.exe file which is dated 3rd >>> >> September. The freeswitch.exe file is dated 7th October and think that >>> it >>> >> contains the minor updates since 3rd September ? >>> >> >>> >> Could someone who knows FreeSwitch under windows help me understand >>> the >>> >> two files please ? >>> >> >>> >> I chickened out of running the later exe in case it did something to >>> the >>> >> running install of FreeSwitch 1.0.4, is it safe to run the newer exe >>> with >>> >> the old one already installed ? >>> >> What will it actually do ? >>> >> >>> >> regards >>> >> Dave >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> > >>> > -- >>> > View this message in context: >>> > >>> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html >>> > Sent from the freeswitch-users mailing list archive at Nabble.com. >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/fa70bb95/attachment-0002.html From edpimentl at gmail.com Wed Nov 25 20:49:10 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 25 Nov 2009 23:49:10 -0500 Subject: [Freeswitch-users] Fwd: passive recording In-Reply-To: References: <8595daf70911250742t3c8584bbp98e890693c088122@mail.gmail.com> <8595daf70911250900q19116f2y14d3b0528a01f8d3@mail.gmail.com> <191c3a030911250913l10cec804w16f62182883fc929@mail.gmail.com> <8595daf70911250929w26eeb3aboae0f95042f35393b@mail.gmail.com> Message-ID: <9dc4a1670911252049j1ba1a2a8g3fdc8c884f843951@mail.gmail.com> Are you wanting to provide "Lawfull Interecept" functionanility for CALEA Compliance? http://www.netequalizer.com/caleafaq.php -E Gpro.ws edpimentl [SKype ] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/a0c1e462/attachment-0002.html From JCasale at activenetwerx.com Wed Nov 25 21:42:28 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 26 Nov 2009 05:42:28 +0000 Subject: [Freeswitch-users] Faxing Advice Message-ID: I need to make faxing easy for some very computer illiterate folk. I am using an email service and going to use procmail to print anything incoming automatically but they cant get the hang of scanning to an email app, so I am going to buy a Linksys PAP2T as per the wiki. Since the setup will never receive inbound remote faxes, I just need to direct all fax's sent from the FXS port (that extension) to the email script in the wiki substituting the destination # as the alias portion of the email. So if I create a dialplan that catches the caller_id_number of the FXS port, does the $1 variable exist in the following scenario: as that's how our service requires fax's, the 10 digit # at their domain, fax.com. Is this a plausible setup? Lastly, I see in the interop list that Audiocodes Mediapack 114 is supported, but the 202 is not listed, is that simply because its new or is it known to not work? Given that its the same price as the Linksys, I would rather get it. Thanks! jlc From mctch at yahoo.com Wed Nov 25 23:33:06 2009 From: mctch at yahoo.com (Mark Crane) Date: Wed, 25 Nov 2009 23:33:06 -0800 (PST) Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: Message-ID: <221275.23339.qm@web56403.mail.re3.yahoo.com> "how about trying Fusionpbx.com? ( GUI)" -Ram I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was ready to release now but decided to do one more release candidate just to be sure. This should be the last release candidate before the release of version 1.0. The final release may be by the end of the week as long as no major issues are found. http://fusionpbx.com --- On Mon, 11/23/09, ram wrote: From: ram Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX To: freeswitch-users at lists.freeswitch.org Date: Monday, November 23, 2009, 10:54 PM On Mon, Nov 23, 2009 at 10:37 AM, Otis wrote: Thanks. I have to get a centos box I guess. Much appreciated Samuel 'Otis' ? how about trying Fusionpbx.com? ( GUI) ? Ram -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/c6afed1e/attachment-0002.html From lakindia89 at gmail.com Thu Nov 26 01:27:10 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Thu, 26 Nov 2009 14:57:10 +0530 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> Message-ID: <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> Hi, Any help or suggestion regarding my previous post. Especially "I also noted that, if I don't receive any events, especially "SERVER_DISCONNECTED", then the connection is in established state, but once I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it correct??" Here is the program by which I confirmed the above! require ESL; use IO::Socket::INET; my $ip = "192.168.1.222"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; my $con; my $type = "user/"; for(;;) { # wait for any client to connect, a new client will get connected when a new call comes in the dialplan. my $new_sock = $sock->accept(); # Do fork and let the parent to wait for more clients. my $pid = fork(); if ($pid) { close($new_sock); next; } # Extract the host of the client. my $host = $new_sock->sockhost(); # file descriptor for the socket. my $fd = fileno($new_sock); print "Host name is $host\n"; # Create object for the ESL connection package to access the ESL functions. $con = new ESL::ESLconnection($fd); # Gets the info about this channel. my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s to %s\n", $uuid, $info->getHeader("caller-caller-id-number"), $info->getHeader("caller-destination-number"); # Answer the channel. $con->execute("answer"); # Set the event lock to tell the FS to execute the instructions in the given order. $con->setEventLock("true"); # Play a file & Get the personal number from the user. $con->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); $con->execute("hangup"); while($con->connected()) { my $e=$con->recvEvent(); my $ename=$e->getHeader("Event-Name"); print $e->serialize(); print "$ename\n"; print "Connection exists\n"; sleep(1); } print "Bye\n------------------------------------------------------------------\n"; close($new_sock); } I've not registered for any events. In the above program I'm receiving the SERVER_DISCONNECTED event. Output when receiving event: Host name is 192.168.1.222 Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 Event-Name: SERVER_DISCONNECTED SERVER_DISCONNECTED Connection exists Bye When I comment the recvEvent line, I got the following output. Host name is 192.168.1.222 Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 Connection exists Connection exists Connection exists Connection exists Connection exists On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy wrote: > I've tried the following program as per the suggestion that you've told. > But it seems, no success. Once the connection is closed, I created a new > connection and I send originate to originate a new call. But it is not > working. > > require ESL; > use IO::Socket::INET; > use Data::Dumper; > > my $ip = "192.168.1.222"; > my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', > Proto => 'tcp', Listen => 2, Reuse => 1 ); > die "Could not create socket: $!\n" unless $sock; > > my $make_call; > my $con; > my $type = "user/"; > > for(;;) { > my $new_sock = $sock->accept(); > my $pid = fork(); > if ($pid) { > close($new_sock); > next; > } > my $host = $new_sock->sockhost(); > my $fd = fileno($new_sock); > $con = new ESL::ESLconnection($fd); > my $info = $con->getInfo(); > my $uuid = $info->getHeader("unique-id"); > printf "Connected call %s, from %s to %s\n", $uuid, > $info->getHeader("caller-caller-id-number"), > $info->getHeader("caller-destination-number"); > > $con->filter("Unique-Id", $uuid); > $con->events("plain", "all"); > $con->execute("answer"); > $con->setEventLock("true"); > my $number=$con->execute("read","2 4 > /usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav > accnt_number 5000 #"); > while($con->connected()) > { > my $e=$con->recvEvent(); > my $ename=$e->getHeader("Event-Name"); > my $app=$e->getHeader("Application"); > if($ename eq "CHANNEL_EXECUTE_COMPLETE" and $app eq "read") > { > my $num=$e->getHeader("variable_accnt_number"); > print "$num\n"; > $con->execute("hangup"); > } > } > if(!$con->connected()) > { > print "Connection not exists\n"; > $con = new ESL::ESLconnection($fd); > $con->api("originate","user/1000 &park()"); > print "Hai\n"; > } > print > "Bye\n------------------------------------------------------------------\n"; > close($new_sock); > } > Output: > Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097 > 1000 > Connection not exists > Hai > Bye > ------------------------------------------------------------------ > The freeswitch log is in > http://pastebin.freeswitch.org/11258 > > I also noted that, if I don't receive any events, especially > "SERVER_DISCONNECTED", then the connection is in established state, but once > I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it > correct?? > > > > > > On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> or open a new outbound connection at the end of your script so you can >> send your originate command. >> Since the channel hanging up will close your existing connection since >> it's only an outbound single session socket. >> >> >> On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins wrote: >> >>> >>> >>> On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < >>> lakindia89 at gmail.com> wrote: >>> >>>> Hi, >>>> I'm using perl ESL to control the call in freeswitch. >>>> I'm having the following scenario, but not able to get it right. >>>> >>>> Dialplan: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> 1. User A calls to an extention (1000). >>>> 2. My ESL program will be running, and it answers the call. >>>> 3. Then the program will get a number from the user. >>>> 4. It will hangup the call. >>>> 5. The program has to call to the number that was given by the user. >>>> >>>> In the above scenario, I was able to do until the 4th step. After hangup >>>> the call, if I say originate it is not working. >>>> Any ideas on how to do this in ESL. >>>> >>>> >>> I want to make sure I understand what the script is supposed to be doing. >>> The caller will key in a phone number to your script and your script will >>> collect those digits. The script will then hangup on the caller and >>> originate a completely new call? Perhaps you could use sched_api to schedule >>> a new originate command for a few seconds into the future and then hangup? >>> -MC >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/10741c63/attachment-0002.html From mike at jerris.com Thu Nov 26 01:53:57 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Nov 2009 04:53:57 -0500 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> Message-ID: Your using outbound socket and you hangup the call, so it tells you it is done with the server disconnected message and drops the connection. This is all as expected. I guess I don't understand what you think is the problem. This code is doing exactly what I would expect it to do. Mike On Nov 26, 2009, at 4:27 AM, lakshmanan ganapathy wrote: > Hi, Any help or suggestion regarding my previous post. Especially > > "I also noted that, if I don't receive any events, especially > "SERVER_DISCONNECTED", then the connection is in established state, > but once I receive the "SERVER_DISCONNECTED" event, the connection > is closed. Is it correct??" > Here is the program by which I confirmed the above! > > require ESL; > use IO::Socket::INET; > > my $ip = "192.168.1.222"; > my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => > '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); > die "Could not create socket: $!\n" unless $sock; > my $con; > my $type = "user/"; > > for(;;) { > # wait for any client to connect, a new client will get > connected when a new call comes in the dialplan. > my $new_sock = $sock->accept(); > # Do fork and let the parent to wait for more clients. > my $pid = fork(); > if ($pid) { > close($new_sock); > next; > } > # Extract the host of the client. > my $host = $new_sock->sockhost(); > # file descriptor for the socket. > my $fd = fileno($new_sock); > print "Host name is $host\n"; > # Create object for the ESL connection package to access the > ESL functions. > $con = new ESL::ESLconnection($fd); > # Gets the info about this channel. > my $info = $con->getInfo(); > my $uuid = $info->getHeader("unique-id"); > printf "Connected call %s, from %s to %s\n", $uuid, $info- > >getHeader("caller-caller-id-number"), $info->getHeader("caller- > destination-number"); > > # Answer the channel. > $con->execute("answer"); > # Set the event lock to tell the FS to execute the > instructions in the given order. > $con->setEventLock("true"); > # Play a file & Get the personal number from the user. > $con->execute("playback","/usr/local/freeswitch/sounds/en/us/ > callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); > $con->execute("hangup"); > while($con->connected()) > { > my $e=$con->recvEvent(); > my $ename=$e->getHeader("Event-Name"); > print $e->serialize(); > print "$ename\n"; > print "Connection exists\n"; > sleep(1); > } > print "Bye > \n------------------------------------------------------------------ > \n"; > close($new_sock); > } > I've not registered for any events. > In the above program I'm receiving the SERVER_DISCONNECTED event. > Output when receiving event: > Host name is 192.168.1.222 > Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 > to 9097 > Event-Name: SERVER_DISCONNECTED > > SERVER_DISCONNECTED > Connection exists > Bye > > When I comment the recvEvent line, I got the following output. > > Host name is 192.168.1.222 > Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 > to 9097 > Connection exists > Connection exists > Connection exists > Connection exists > Connection exists > > > On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy > wrote: > I've tried the following program as per the suggestion that you've > told. But it seems, no success. Once the connection is closed, I > created a new connection and I send originate to originate a new > call. But it is not working. > > require ESL; > use IO::Socket::INET; > use Data::Dumper; > > my $ip = "192.168.1.222"; > my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => > '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); > die "Could not create socket: $!\n" unless $sock; > > my $make_call; > my $con; > my $type = "user/"; > > for(;;) { > my $new_sock = $sock->accept(); > my $pid = fork(); > if ($pid) { > close($new_sock); > next; > } > my $host = $new_sock->sockhost(); > my $fd = fileno($new_sock); > $con = new ESL::ESLconnection($fd); > my $info = $con->getInfo(); > my $uuid = $info->getHeader("unique-id"); > printf "Connected call %s, from %s to %s\n", $uuid, $info- > >getHeader("caller-caller-id-number"), $info->getHeader("caller- > destination-number"); > > $con->filter("Unique-Id", $uuid); > $con->events("plain", "all"); > $con->execute("answer"); > $con->setEventLock("true"); > my $number=$con->execute("read","2 4 /usr/local/freeswitch/ > sounds/en/us/callie/conference/8000/conf-pin.wav accnt_number 5000 > #"); > while($con->connected()) > { > my $e=$con->recvEvent(); > my $ename=$e->getHeader("Event-Name"); > my $app=$e->getHeader("Application"); > if($ename eq "CHANNEL_EXECUTE_COMPLETE" and $app eq > "read") > { > my $num=$e->getHeader > ("variable_accnt_number"); > print "$num\n"; > $con->execute("hangup"); > } > } > if(!$con->connected()) > { > print "Connection not exists\n"; > $con = new ESL::ESLconnection($fd); > $con->api("originate","user/1000 &park()"); > print "Hai\n"; > } > print "Bye > \n------------------------------------------------------------------ > \n"; > close($new_sock); > } > Output: > Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097 > 1000 > Connection not exists > Hai > Bye > ------------------------------------------------------------------ > The freeswitch log is in > http://pastebin.freeswitch.org/11258 > > I also noted that, if I don't receive any events, especially > "SERVER_DISCONNECTED", then the connection is in established state, > but once I receive the "SERVER_DISCONNECTED" event, the connection > is closed. Is it correct?? > > > > > > On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale > wrote: > or open a new outbound connection at the end of your script so you > can send your originate command. > Since the channel hanging up will close your existing connection > since it's only an outbound single session socket. > > > On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins > wrote: > > > On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy > wrote: > Hi, > I'm using perl ESL to control the call in freeswitch. > I'm having the following scenario, but not able to get it right. > > Dialplan: > > > > > > > > > 1. User A calls to an extention (1000). > 2. My ESL program will be running, and it answers the call. > 3. Then the program will get a number from the user. > 4. It will hangup the call. > 5. The program has to call to the number that was given by the user. > > In the above scenario, I was able to do until the 4th step. After > hangup the call, if I say originate it is not working. > Any ideas on how to do this in ESL. > > > I want to make sure I understand what the script is supposed to be > doing. The caller will key in a phone number to your script and your > script will collect those digits. The script will then hangup on the > caller and originate a completely new call? Perhaps you could use > sched_api to schedule a new originate command for a few seconds into > the future and then hangup? > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/2439d860/attachment-0002.html From abeka at greatiam.com Thu Nov 26 02:53:23 2009 From: abeka at greatiam.com (Otis) Date: Thu, 26 Nov 2009 10:53:23 +0000 Subject: [Freeswitch-users] Requesting testing. In-Reply-To: References: Message-ID: <4B0E5E23.1060804@greatiam.com> Hi Checked out svn checkout y'day. I am in the UK. Installed . Installed on Fedora 11 i386 box. : bootstrap.sh configue --without-libcurl make make install On startup only errors: PMP I'm not behind a NAT so OK Stacksize registered as too high and advised to use the -waste switch. Other than the stack thing all quiet on the new front, Sir regards Michael Jerris wrote: > I have done quite a few changes to the build system and correcting build problems and other platform specific problems the last few days. Could everyone on the list please take a little time out of their day and do a clean fresh svn trunk checkout of FreeSWITCH and do a full build and report any errors you encounter (if not already reported) to http://jira.freeswitch.org. We have fixed things for many platforms including bsd, solaris, linux, and especially issues on OS X. Please try these out to make sure all works. > > Thanks > Mike > > > > From abeka at greatiam.com Thu Nov 26 02:59:00 2009 From: abeka at greatiam.com (Otis) Date: Thu, 26 Nov 2009 10:59:00 +0000 Subject: [Freeswitch-users] Help Freeswitch with Voipuser Gateway In-Reply-To: <4B0B2655.4010900@greatiam.com> References: <4B086689.6080804@greatiam.com> <4B097A89.2050400@greatiam.com> <4B0ABC4F.1010103@greatiam.com> <4B0B2655.4010900@greatiam.com> Message-ID: <4B0E5F74.10009@greatiam.com> This is resolved. I could someone to call my VOIPUSER number and call transferred to my designated extension. I could not get this to work from my network ie calling from one of my extensions and setting that the call be -rerouted to another extension. All OK now. Thanks folks Otis wrote: > Has anyone got any suggestion how I can set up a gateway to receive > incoming call on extension 1001 please. > Any generic conf file will do. my username with my gateway is s=say " > qwerty" and password "ytrewq" > > I have used the intruction from the link below without success. > > Thanks. > > > > > Otis wrote: >> Hello >> >> Could anyone point out what I have missed please ? >> At the moment I configured a gateway voipuser as described here >> : >> Any suggestion as to what path I can take will be highly welcome >> >> Thanks >> . >> >> >> >> >> Sam Abekah-Mensah wrote: >>>
Hi Michael >>> >>> Thanks >>> >>> I had set it to send incoming calls to extension 1001. This is in >>> the file abeka.xml in /usr/local/freeswitch/conf/dialplan/public >>> directory. >>> The contents are : >>> >>> >>> >>> >>> >>> >>> >>> >>> Is there >>> anything wrong with this please ? >>> >>> Thanks >>> >>> >>> >>> Michal Bielicki wrote: >>>> >>>> Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah: >>>> >>>>> >>>>> I need help as I cannot receive calls through VOIPUSER. This is a >>>>> learning setup Attached are my conf files. What is wrong with them >>>>> ? When I dial from a landline I get a continuous beep. >>>>> >>>>> Attached are my gateway and the conf file to transfer. Sopfia >>>>> Status is my screen message. I can see a FAIL and cannot make head >>>>> or tail of all that message. Hopefully anyone using voipuser or in >>>>> fact any of you clever folks can make sense of this. >>>>> >>>>> Thanks for your time. >>>>> >>>>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate >>>>> Buggy RFC2833 Mode! >>>>> 2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec >>>>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>>> 2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec >>>>> Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] >>>>> 2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec >>>>> sofia/external/nobody at 213.166.5.133 PCMA/8000 20 ms 160 samples >>>>> 2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf >>>>> payload to 101 >>>>> 2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 >>>>> (sofia/external/nobody at 213.166.5.133) State Change CS_NEW -> CS_INIT >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_INIT >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>>>> (sofia/external/nobody at 213.166.5.133) State INIT >>>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 >>>>> sofia/external/nobody at 213.166.5.133 SOFIA INIT >>>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 >>>>> (sofia/external/nobody at 213.166.5.133) State Change CS_INIT -> >>>>> CS_ROUTING >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 >>>>> (sofia/external/nobody at 213.166.5.133) State INIT going to sleep >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 >>>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_ROUTING >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 >>>>> (sofia/external/nobody at 213.166.5.133) State ROUTING >>>>> 2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 >>>>> sofia/external/nobody at 213.166.5.133 SOFIA ROUTING >>>>> 2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 >>>>> sofia/external/nobody at 213.166.5.133 Standard ROUTING >>>>> 2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 >>>>> Processing anonymous->abeka in context public >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->unloop] continue=false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (PASS) >>>>> [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->outside_call] continue=true >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Absolute Condition >>>>> [outside_call] >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Action >>>>> set(outside_call=true) >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->call_debug] continue=true >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->public_extensions] continue=false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/ >>>>> break=on-false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->public_did] continue=false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [public_did] destination_number(abeka) =~ /^(5551212)$/ >>>>> break=on-false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->sip at sip.voipuser.org] continue=false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [sip at sip.voipuser.org] destination_number(abeka) =~ /08715042951/ >>>>> break=on-false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 parsing >>>>> [public->Inbound-abeka at sip.voipuser.org]] continue=false >>>>> Dialplan: sofia/external/nobody at 213.166.5.133 Regex (FAIL) >>>>> [Inbound-abeka at sip.voipuser.org]] destination_number(abeka) =~ >>>>> /[08444846450]/ break=on-false >>>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 >>>>> (sofia/external/nobody at 213.166.5.133) State Change CS_ROUTING -> >>>>> CS_EXECUTE >>>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 >>>>> (sofia/external/nobody at 213.166.5.133) State ROUTING going to sleep >>>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 >>>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_EXECUTE >>>>> 2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 >>>>> (sofia/external/nobody at 213.166.5.133) State EXECUTE >>>>> 2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 >>>>> sofia/external/nobody at 213.166.5.133 SOFIA EXECUTE >>>>> 2009-11-21 22:07:15.706658 [DEBUG] switch_core_state_machine.c:151 >>>>> sofia/external/nobody at 213.166.5.133 Standard EXECUTE >>>>> EXECUTE sofia/external/nobody at 213.166.5.133 set(outside_call=true) >>>>> 2009-11-21 22:07:15.728613 [DEBUG] mod_dptools.c:748 >>>>> sofia/external/nobody at 213.166.5.133 SET [outside_call]=[true] >>>>> 2009-11-21 22:07:15.728613 [NOTICE] >>>>> switch_core_state_machine.c:179 Hangup >>>>> sofia/external/nobody at 213.166.5.133 [CS_EXECUTE] [NORMAL_CLEARING] >>>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_channel.c:1683 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [KILL] >>>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_session.c:932 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:491 >>>>> (sofia/external/nobody at 213.166.5.133) State EXECUTE going to sleep >>>>> 2009-11-21 22:07:15.728613 [DEBUG] switch_core_state_machine.c:398 >>>>> (sofia/external/nobody at 213.166.5.133) Running State Change CS_HANGUP >>>>> 2009-11-21 22:07:15.735830 [DEBUG] switch_core_state_machine.c:434 >>>>> (sofia/external/nobody at 213.166.5.133) State HANGUP >>>>> 2009-11-21 22:07:15.735830 [DEBUG] mod_sofia.c:338 Channel >>>>> sofia/external/nobody at 213.166.5.133 hanging up, cause: >>>>> NORMAL_CLEARING >>>>> 2009-11-21 22:07:15.737680 [DEBUG] mod_sofia.c:417 Responding to >>>>> INVITE with: 480 >>>>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:46 >>>>> sofia/external/nobody at 213.166.5.133 Standard HANGUP, cause: >>>>> NORMAL_CLEARING >>>>> 2009-11-21 22:07:15.741149 [DEBUG] switch_core_state_machine.c:434 >>>>> (sofia/external/nobody at 213.166.5.133) State HANGUP going to sleep >>>>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_state_machine.c:476 >>>>> (sofia/external/nobody at 213.166.5.133) State Change CS_HANGUP -> >>>>> CS_REPORTING >>>>> 2009-11-21 22:07:15.742930 [DEBUG] switch_core_session.c:932 Send >>>>> signal sofia/external/nobody at 213.166.5.133 [BREAK] >>>>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:398 >>>>> (sofia/external/nobody at 213.166.5.133) Running State Change >>>>> CS_REPORTING >>>>> 2009-11-21 22:07:15.744587 [DEBUG] switch_core_state_machine.c:612 >>>>> (sofia/external/nobody at 213.166.5.133) State REPORTING >>>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:53 >>>>> sofia/external/nobody at 213.166.5.133 Standard REPORTING, cause: >>>>> NORMAL_CLEARING >>>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:612 >>>>> (sofia/external/nobody at 213.166.5.133) State REPORTING going to sleep >>>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_state_machine.c:411 >>>>> (sofia/external/nobody at 213.166.5.133) State Change CS_REPORTING -> >>>>> CS_DESTROY >>>>> 2009-11-21 22:07:15.800497 [DEBUG] switch_core_session.c:1068 >>>>> Session 2 (sofia/external/nobody at 213.166.5.133) Locked, Waiting on >>>>> external entities >>>>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1086 >>>>> Session 2 (sofia/external/nobody at 213.166.5.133) Ended >>>>> 2009-11-21 22:07:15.800497 [NOTICE] switch_core_session.c:1088 >>>>> Close Channel sofia/external/nobody at 213.166.5.133 [CS_DESTROY] >>>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>>>> (sofia/external/nobody at 213.166.5.133) State DESTROY >>>>> 2009-11-21 22:07:15.802636 [DEBUG] mod_sofia.c:255 >>>>> sofia/external/nobody at 213.166.5.133 SOFIA DESTROY >>>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:60 >>>>> sofia/external/nobody at 213.166.5.133 Standard DESTROY >>>>> 2009-11-21 22:07:15.802636 [DEBUG] switch_core_state_machine.c:564 >>>>> (sofia/external/nobody at 213.166.5.133) State DESTROY going to sleep >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> : >>>> >>>> >>>> you seem to have not specified an extension where the call should >>>> go to >>>> my voipuser.org setup looks like: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> I am also surprised that your setup works with a from-domain of >>>> sip.voipuser.org >>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >>>>> http://www.freeswitch.org >>>> >>>> *Michal Bielicki* >>>> HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa >>>> t. +48228753290 | f. +48228753291 michal.bielicki at halokwadrat.pl >>>> | w. >>>> www.halokwadrat.pl >>>> >>>> >>>> >>>> *Knowledge & Low Prices. Guaranteed!* >>>> >>> >>> >>> >>>
>> >> > > From abeka at greatiam.com Thu Nov 26 03:02:12 2009 From: abeka at greatiam.com (Otis) Date: Thu, 26 Nov 2009 11:02:12 +0000 Subject: [Freeswitch-users] Re-routing calls to PSTN Message-ID: <4B0E6034.6050802@greatiam.com> Hi folks Can I get FS to re-route incoming-calls to PSTN. If this has been raised before could someone direct me to URL or link please Thanks. From jonas.gauffin at gmail.com Thu Nov 26 03:03:58 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Thu, 26 Nov 2009 12:03:58 +0100 Subject: [Freeswitch-users] Problems with nat Message-ID: I got a freeswitch that is behind nat and got three profiles. External (all calls are going through a proxy): Internal (phones on the same lan as FS) Wan (phones that are not in the same LAN, connecting from internet) The problem is that phones registered on the internal profile gets RECOVERY_ON_TIMER_EXPIRE error after 40-60 seconds. Audio works fine in all profiles. Log from a call: http://pastebin.freeswitch.org/11303 I'm running freeswitch with the -nonat option. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/e9edb679/attachment-0002.html From michaelt at voxcore.voxtelecom.co.za Thu Nov 26 04:08:03 2009 From: michaelt at voxcore.voxtelecom.co.za (Michael Toop) Date: Thu, 26 Nov 2009 14:08:03 +0200 Subject: [Freeswitch-users] B Leg Account Code on Fail Over dialing Message-ID: <330316f60911260408u408c8e2bq3e5cd311008d8a8@mail.gmail.com> Hi Everyone, How do I get the outbound sofia SIP route that the call took into a CDR when using fail over dialing with the 'bridge' application? ...I have tried numerous approaches with no luck, this last attempt pasted below did not work either: dialString = "{provider=providerB}sofia/gateway/sipB/%s|{provider=providerC}sofia/gateway/sipC/%s" % (numberToDial, numberToDial) session.execute("bridge",dialString) I am using mod_python and this line is in the Python module called in the dialplan. Thanks! Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/8d54eb4f/attachment-0002.html From abeka at greatiam.com Thu Nov 26 04:11:04 2009 From: abeka at greatiam.com (Otis) Date: Thu, 26 Nov 2009 12:11:04 +0000 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <221275.23339.qm@web56403.mail.re3.yahoo.com> References: <221275.23339.qm@web56403.mail.re3.yahoo.com> Message-ID: <4B0E7058.1010106@greatiam.com> Thank you so much. Regards Mark Crane wrote: > "how about trying Fusionpbx.com ( GUI)" -Ram > > I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was > ready to release now but decided to do one more release candidate just > to be sure. This should be the last release candidate before the > release of version 1.0. > > The final release may be by the end of the week as long as no major > issues are found. > > http://fusionpbx.com > > > > > --- On *Mon, 11/23/09, ram //* wrote: > > > From: ram > Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > To: freeswitch-users at lists.freeswitch.org > Date: Monday, November 23, 2009, 10:54 PM > > > > On Mon, Nov 23, 2009 at 10:37 AM, Otis > wrote: > > Thanks. > > I have to get a centos box I guess. > > Much appreciated > > Samuel 'Otis' > > > > how about trying Fusionpbx.com ( GUI) > > Ram > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From abeka at greatiam.com Thu Nov 26 04:24:36 2009 From: abeka at greatiam.com (Otis) Date: Thu, 26 Nov 2009 12:24:36 +0000 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <221275.23339.qm@web56403.mail.re3.yahoo.com> References: <221275.23339.qm@web56403.mail.re3.yahoo.com> Message-ID: <4B0E7384.5010809@greatiam.com> Thanks. I will try it . I am on Fedora 11 Mark Crane wrote: > "how about trying Fusionpbx.com ( GUI)" -Ram > > I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was > ready to release now but decided to do one more release candidate just > to be sure. This should be the last release candidate before the > release of version 1.0. > > The final release may be by the end of the week as long as no major > issues are found. > > http://fusionpbx.com > > > > > --- On *Mon, 11/23/09, ram //* wrote: > > > From: ram > Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > To: freeswitch-users at lists.freeswitch.org > Date: Monday, November 23, 2009, 10:54 PM > > > > On Mon, Nov 23, 2009 at 10:37 AM, Otis > wrote: > > Thanks. > > I have to get a centos box I guess. > > Much appreciated > > Samuel 'Otis' > > > > how about trying Fusionpbx.com ( GUI) > > Ram > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ovvenkatesan at gmail.com Thu Nov 26 04:38:12 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Thu, 26 Nov 2009 18:08:12 +0530 Subject: [Freeswitch-users] How to run IVR application In-Reply-To: <87f2f3b90911241955v4e726111ked993c8dbb556f99@mail.gmail.com> References: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> <87f2f3b90911241955v4e726111ked993c8dbb556f99@mail.gmail.com> Message-ID: <47d63d920911260438j29b56ee5w587bd6315eb64c42@mail.gmail.com> Thank you very much MC . Its working :) . I started loving "FS" ;) On Wed, Nov 25, 2009 at 9:25 AM, Michael Collins wrote: > > > On Tue, Nov 24, 2009 at 6:03 PM, Lei Tang wrote: > >> you can do this in follow steps: >> 1.edit default.xml diaplan config file in your fs config >> directory(FS/conf/dialplan/default.xml), and section >> >> >> >> >> >> 2. edit your ivr script, your can refer to >> http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in >> lua. >> 3. connect your sip phone to fs, and dial 114, this will launch your ivr >> application >> >> > > You can also do IVRs with static XML. I recommend you try out the demo IVR > by dialing 5000. Now go look at the two main files that we used to build > that IVR: > > conf/autoload_configs/ivr.conf.xml (menu structure) > conf/lang/en/demo/demo-ivr.xml (phrase macros) > > it's overwhelming at first, however once you get the hang of it you'll > appreciate how powerful it is. The wiki and the sample XML config files have > lots of information so be sure to read as much as you can and try things. > You can't break anything. :) > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/953db637/attachment-0002.html From rupa at rupa.com Thu Nov 26 05:50:42 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 26 Nov 2009 07:50:42 -0600 Subject: [Freeswitch-users] B Leg Account Code on Fail Over dialing In-Reply-To: <330316f60911260408u408c8e2bq3e5cd311008d8a8@mail.gmail.com> References: <330316f60911260408u408c8e2bq3e5cd311008d8a8@mail.gmail.com> Message-ID: You need to use brackets [] not braces {} for per-leg variables. On Thu, Nov 26, 2009 at 6:08 AM, Michael Toop < michaelt at voxcore.voxtelecom.co.za> wrote: > Hi Everyone, > > How do I get the outbound sofia SIP route that the call took into a CDR > when using fail over dialing with the 'bridge' application? > > ...I have tried numerous approaches with no luck, this last attempt pasted > below did not work either: > > dialString = > "{provider=providerB}sofia/gateway/sipB/%s|{provider=providerC}sofia/gateway/sipC/%s" > % (numberToDial, numberToDial) > session.execute("bridge",dialString) > > I am using mod_python and this line is in the Python module called in the > dialplan. > > Thanks! > > Michael > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/05dd1522/attachment-0002.html From freeswitch at servercorps.com Thu Nov 26 07:17:29 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Thu, 26 Nov 2009 09:17:29 -0600 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <4B0E7384.5010809@greatiam.com> References: <221275.23339.qm@web56403.mail.re3.yahoo.com> <4B0E7384.5010809@greatiam.com> Message-ID: <92e7d2090911260717j11ffad78kdd11b1c87dfd87be@mail.gmail.com> Fedora and Centos installation instructions are very similar. You should be able to compile on Fedora without any problems that I'm aware of. Regards, Nik On Thu, Nov 26, 2009 at 06:24, Otis wrote: > Thanks. I will try it . I am on Fedora 11 > > > > Mark Crane wrote: > > "how about trying Fusionpbx.com ( GUI)" -Ram > > > > I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was > > ready to release now but decided to do one more release candidate just > > to be sure. This should be the last release candidate before the > > release of version 1.0. > > > > The final release may be by the end of the week as long as no major > > issues are found. > > > > http://fusionpbx.com > > > > > > > > > > --- On *Mon, 11/23/09, ram //* wrote: > > > > > > From: ram > > Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > > To: freeswitch-users at lists.freeswitch.org > > Date: Monday, November 23, 2009, 10:54 PM > > > > > > > > On Mon, Nov 23, 2009 at 10:37 AM, Otis > > wrote: > > > > Thanks. > > > > I have to get a centos box I guess. > > > > Much appreciated > > > > Samuel 'Otis' > > > > > > > > how about trying Fusionpbx.com ( GUI) > > > > Ram > > > > -----Inline Attachment Follows----- > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/1f1d2c65/attachment-0002.html From brian at freeswitch.org Thu Nov 26 07:42:37 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Nov 2009 09:42:37 -0600 Subject: [Freeswitch-users] Problems with nat In-Reply-To: References: Message-ID: Are you doing this all on a linux box thats acting as your router too? If not you don't need two profiles... you also don't need to set the local-network-acl on ANY profile that isn't do anything with nat. /b On Nov 26, 2009, at 5:03 AM, Jonas Gauffin wrote: > I got a freeswitch that is behind nat and got three profiles. > > External (all calls are going through a proxy): > > > > > > > Internal (phones on the same lan as FS) > > > > > Wan (phones that are not in the same LAN, connecting from internet) > > > > > > > The problem is that phones registered on the internal profile gets > RECOVERY_ON_TIMER_EXPIRE error after 40-60 seconds. Audio works fine > in all profiles. > > Log from a call: > http://pastebin.freeswitch.org/11303 > > I'm running freeswitch with the -nonat option. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/e7541c6f/attachment-0002.html From orien at tx.rr.com Thu Nov 26 09:38:19 2009 From: orien at tx.rr.com (Orien Love) Date: Thu, 26 Nov 2009 11:38:19 -0600 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. Message-ID: <4B0EBD0B.7000905@tx.rr.com> Is there any way to build a dial plan so that when an extension calls itself the call is automatically put to that users voice mail? Example, extension 1001 calling 1001 and is sent to voice mail (to receive messages). I know that there is a * code to get to voice mail, I cannot recall which one right now but my phones want to dial their extension to get to voice mail.I can modify the voice mail button but this works only for the first line registered at that phone. Any help is appreciated. Orien From jonas.gauffin at gmail.com Thu Nov 26 10:14:34 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Thu, 26 Nov 2009 19:14:34 +0100 Subject: [Freeswitch-users] Problems with nat In-Reply-To: References: Message-ID: It's a windowsserver which is behind a router. Which profile should local-network-acl be specified on? When I bridge calls to the outside world, should I use sofia/internal/@gateway or sofia/external/@gateway? On Thu, Nov 26, 2009 at 4:42 PM, Brian West wrote: > Are you doing this all on a linux box thats acting as your router too? If > not you don't need two profiles... you also don't need to set the > local-network-acl on ANY profile that isn't do anything with nat. > > /b > > On Nov 26, 2009, at 5:03 AM, Jonas Gauffin wrote: > > I got a freeswitch that is behind nat and got three profiles. > > External (all calls are going through a proxy): > > > > > > > Internal (phones on the same lan as FS) > > > > > Wan (phones that are not in the same LAN, connecting from internet) > > > > > > > The problem is that phones registered on the internal profile > gets RECOVERY_ON_TIMER_EXPIRE error after 40-60 seconds. Audio works fine in > all profiles. > > Log from a call: > http://pastebin.freeswitch.org/11303 > > I'm running freeswitch with the -nonat option. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/592524e3/attachment-0002.html From frank at impactfax.com Thu Nov 26 11:22:52 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 26 Nov 2009 14:22:52 -0500 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS Message-ID: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> "GREAT SCOTT!!! Cannot execute batched statements! If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS" I realize a bit off of list topic. But I do have mysql 3.51.18 and higher but for the life of me , I cannot seem to get the DSN config setup so that the odbc connector seems to tell FS that it can do multi statements. Anyone have any insight on how and where to set this flag? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/118b0a06/attachment-0002.html From timuckun at gmail.com Thu Nov 26 12:22:44 2009 From: timuckun at gmail.com (Tim Uckun) Date: Fri, 27 Nov 2009 09:22:44 +1300 Subject: [Freeswitch-users] XML config file parsing In-Reply-To: References: <9e6fbacf0911190541m3d756507u27f9ecd944197bc6@mail.gmail.com> <691E4EF6-B22B-4FE2-8A3D-01A1D599A448@gmail.com> <855e4dcf0911221543o222bef63t1c3340b0a41d57c1@mail.gmail.com> <191c3a030911230838l103bc466p7582c1d05730f61a@mail.gmail.com> Message-ID: <855e4dcf0911261222x7c73593dn18e0c2997f09d633@mail.gmail.com> On Tue, Nov 24, 2009 at 5:48 AM, Eliot Gable wrote: > Or, you can use something like Smarty to cache your generated XML on > your web server and only invalidate those cached results when you > change something that will impact them. That sounds like a pretty sane way to go bout it. From mike at jerris.com Thu Nov 26 12:44:48 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Nov 2009 15:44:48 -0500 Subject: [Freeswitch-users] Problems with nat In-Reply-To: References: Message-ID: In this case you should not need 2 profiles either. On Nov 26, 2009, at 1:14 PM, Jonas Gauffin wrote: > It's a windowsserver which is behind a router. > > Which profile should local-network-acl be specified on? > > When I bridge calls to the outside world, should I use sofia/internal/@gateway or sofia/external/@gateway? > > > On Thu, Nov 26, 2009 at 4:42 PM, Brian West wrote: > Are you doing this all on a linux box thats acting as your router too? If not you don't need two profiles... you also don't need to set the local-network-acl on ANY profile that isn't do anything with nat. > > /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/afd5f554/attachment-0002.html From mike at jerris.com Thu Nov 26 12:47:00 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Nov 2009 15:47:00 -0500 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. In-Reply-To: <4B0EBD0B.7000905@tx.rr.com> References: <4B0EBD0B.7000905@tx.rr.com> Message-ID: <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> Of course. Please read through the default configs and the getting started guide and xml dialplan information on the wiki. Mike On Nov 26, 2009, at 12:38 PM, Orien Love wrote: > Is there any way to build a dial plan so that when an extension calls > itself the call is automatically put to that users voice mail? > > Example, extension 1001 calling 1001 and is sent to voice mail (to > receive messages). > I know that there is a * code to get to voice mail, I cannot recall > which one right now but my phones want to dial their extension to get to > voice mail.I can modify the voice mail button but this works only for > the first line registered at that phone. From mike at jerris.com Thu Nov 26 12:53:21 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Nov 2009 15:53:21 -0500 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> References: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> Message-ID: <0655757B-61CA-490D-BDB0-873263555575@jerris.com> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445) On Nov 26, 2009, at 2:22 PM, Frank @ Impact wrote: > ?GREAT SCOTT!!! Cannot execute batched statements! > If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS? > > I realize a bit off of list topic? > > But I do have mysql 3.51.18 and higher but for the life of me , I cannot seem to get the DSN config setup so that the odbc connector seems to tell FS that it can do multi statements. > > Anyone have any insight on how and where to set this flag? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/ec26c59d/attachment-0002.html From JCasale at activenetwerx.com Thu Nov 26 13:36:04 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 26 Nov 2009 21:36:04 +0000 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. In-Reply-To: <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> References: <4B0EBD0B.7000905@tx.rr.com> <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> Message-ID: >Of course. Please read through the default configs and the getting started guide and xml dialplan information on the wiki. > >Mike This is of interest to me as well, would that be something like this: Could anyone versed in xml and variables comment on this so it generically checked if the extension dialed was of your extension length, like ^(\d{3})$ then if it matched your caller_id_number go into the action so you could leave it as $1, not 100 in my case? That way you could only have one of these plans work for all extensions. From tculjaga at gmail.com Thu Nov 26 13:48:22 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 26 Nov 2009 22:48:22 +0100 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: <0655757B-61CA-490D-BDB0-873263555575@jerris.com> References: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> <0655757B-61CA-490D-BDB0-873263555575@jerris.com> Message-ID: <65d96fc80911261348p18c1d021of3b6500ff798f345@mail.gmail.com> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris wrote: > http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html > > MySQL Connector/ODBC now supports batched statements. In order to enable > cached statement support you must switch enable the batched > statement option (FLAG_MULTI_STATEMENTS, > 67108864, or Allow multiple statements > within a GUI configuration). Be aware that batched statements > create an increased chance of SQL injection attacks and you must > ensure that your application protects against this scenario. > (Bug#7445 ) > > > so, is this the right patch ? http://bugs.mysql.com/file.php?id=6994 T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/a512b1d8/attachment-0002.html From eman at chabotel.com Thu Nov 26 14:26:15 2009 From: eman at chabotel.com (freeswitch list) Date: Thu, 26 Nov 2009 17:26:15 -0500 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. In-Reply-To: References: <4B0EBD0B.7000905@tx.rr.com> <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> Message-ID: <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> On Thu, Nov 26, 2009 at 4:36 PM, Joseph L. Casale wrote: > >Of course. Please read through the default configs and the getting > started guide and xml dialplan information on the wiki. > > > >Mike > > This is of interest to me as well, would that be something like this: > > > > > > > > > > Could anyone versed in xml and variables comment on this so it generically > checked > if the extension dialed was of your extension length, like ^(\d{3})$ then > if it matched > your caller_id_number go into the action so you could leave it as $1, not > 100 in my case? > > That way you could only have one of these plans work for all extensions. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/d00c2357/attachment-0002.html From JCasale at activenetwerx.com Thu Nov 26 14:43:24 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 26 Nov 2009 22:43:24 +0000 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. In-Reply-To: <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> References: <4B0EBD0B.7000905@tx.rr.com> <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> Message-ID: >? Of course:) Thank you! jlc From Russell.Mosemann at cune.org Thu Nov 26 14:45:11 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Thu, 26 Nov 2009 16:45:11 -0600 Subject: [Freeswitch-users] dialplan rule to send the caller to voicemail when same extension is called. In-Reply-To: <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> References: <4B0EBD0B.7000905@tx.rr.com><2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> Message-ID: <33C8A289190544CA8CA66014B3153363@cune.pri> freeswitch list wrote: > I knew this day would come. After the accumulation of all of the knowledge from the list members, the list has finally achieved sentience and is now answering questions by itself. :-) -- Russell Mosemann From Prometheus001 at gmx.net Thu Nov 26 15:55:17 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 27 Nov 2009 00:55:17 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> Message-ID: <4B0F1565.6060909@gmx.net> I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded. Best regards Peter Brian West schrieb: > Yes an alias will be required for every domain you run on the profile > so it can find it. > > /b > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > >> Try an alias on the sip profile. >> >> Mike >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From rob4manhere at gmail.com Thu Nov 26 18:27:46 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 26 Nov 2009 20:27:46 -0600 Subject: [Freeswitch-users] dialplan rule to send the caller to voicemail when same extension is called. In-Reply-To: <33C8A289190544CA8CA66014B3153363@cune.pri> References: <4B0EBD0B.7000905@tx.rr.com><2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> <33C8A289190544CA8CA66014B3153363@cune.pri> Message-ID: <33089F3D-0DC0-481B-B1A7-E58B35A392A8@gmail.com> LOL thats funny. freeswitch, what is the meaning of life? On Nov 26, 2009, at 4:45 PM, Russell Mosemann wrote: > freeswitch list wrote: > >> > > I knew this day would come. After the accumulation of all of the > knowledge from the list members, the list has finally achieved > sentience and is now answering questions by itself. :-) > > -- > Russell Mosemann > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andrewkt at aktzero.com Thu Nov 26 20:08:21 2009 From: andrewkt at aktzero.com (Andrew Thompson) Date: Thu, 26 Nov 2009 23:08:21 -0500 Subject: [Freeswitch-users] Re-routing calls to PSTN In-Reply-To: <4B0E6034.6050802@greatiam.com> References: <4B0E6034.6050802@greatiam.com> Message-ID: <4B0F50B5.6030804@aktzero.com> On 11/26/2009 6:02 AM, Otis wrote: > Can I get FS to re-route incoming-calls to PSTN. If this has been > raised before could someone direct me to URL or link please Since I don't have a hard line, I do something like: -- Andrew Thompson From andrew at hijacked.us Thu Nov 26 20:13:26 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 26 Nov 2009 23:13:26 -0500 Subject: [Freeswitch-users] Business/holiday hours routing In-Reply-To: <016701ca6dfc$1a8e9ae0$4fabd0a0$@net> References: <00be01ca6ca5$31f64ff0$95e2efd0$@net> <20091124014808.GB3298@hijacked.us> <00e101ca6cab$c3525240$49f6f6c0$@net> <21CB5F92-98DE-4622-ADC5-013462A93BD2@freeswitch.org> <20091124064509.GA6360@hijacked.us> <016701ca6dfc$1a8e9ae0$4fabd0a0$@net> Message-ID: <20091127041326.GA2000@hijacked.us> On Wed, Nov 25, 2009 at 11:21:25AM -0700, Adam Ford wrote: > Awesome, thanks Andrew, I will have to keep an eye out for that patch. > Here's my patch in its (probably) final form. http://eagle.bsd.st/~andrew/mweek2.diff It includes a usage example that covers all but 2 of the US federal holidays (memorial day is a real toughie). I'm just waiting on Tony to green light it for commit. If the patch looks like a mess in your browser, blame the XML :) Andrew From eman at chabotel.com Thu Nov 26 20:26:26 2009 From: eman at chabotel.com (eman) Date: Thu, 26 Nov 2009 23:26:26 -0500 Subject: [Freeswitch-users] dialplan rule to send the caller to voicemail when same extension is called. In-Reply-To: <33089F3D-0DC0-481B-B1A7-E58B35A392A8@gmail.com> References: <4B0EBD0B.7000905@tx.rr.com> <2253294A-0247-4785-BA78-01DEB9D10E2D@jerris.com> <164a9ab00911261426p6bbd11cet9eb85d89b65aa12b@mail.gmail.com> <33C8A289190544CA8CA66014B3153363@cune.pri> <33089F3D-0DC0-481B-B1A7-E58B35A392A8@gmail.com> Message-ID: <164a9ab00911262026w65b65b1dw12846463af04170a@mail.gmail.com> Weird don't know how that got set to freeswitch list. On Thu, Nov 26, 2009 at 9:27 PM, Rob Forman wrote: > LOL thats funny. > > freeswitch, what is the meaning of life? > > > On Nov 26, 2009, at 4:45 PM, Russell Mosemann wrote: > > > freeswitch list wrote: > > > >> > > > > I knew this day would come. After the accumulation of all of the > > knowledge from the list members, the list has finally achieved > > sentience and is now answering questions by itself. :-) > > > > -- > > Russell Mosemann > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091126/28feed69/attachment-0002.html From orien at tx.rr.com Thu Nov 26 20:51:30 2009 From: orien at tx.rr.com (Orien Love) Date: Thu, 26 Nov 2009 22:51:30 -0600 Subject: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called.(Working) In-Reply-To: References: Message-ID: <4B0F5AD2.5040000@tx.rr.com> Thanks for all the help, here is what I put in the dialplan, I tested this and it is working for me. this was added just before the line Orien Love. Still learning, but getting there with help from all the great people on this list :) Subject: Re: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called. From: freeswitch list Date: Thu, 26 Nov 2009 17:26:15 -0500 To: freeswitch-users at lists.freeswitch.org > > On Thu, Nov 26, 2009 at 4:36 PM, Joseph L. Casale > > wrote: > > >Of course. Please read through the default configs and the > getting started guide and xml dialplan information on the wiki. > > > >Mike > > This is of interest to me as well, would that be something like this: > > > > > > > > > > Could anyone versed in xml and variables comment on this so it > generically checked > if the extension dialed was of your extension length, like > ^(\d{3})$ then if it matched > your caller_id_number go into the action so you could leave it as > $1, not 100 in my case? > > That way you could only have one of these plans work for all > extensions. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ovvenkatesan at gmail.com Thu Nov 26 21:59:11 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Fri, 27 Nov 2009 11:29:11 +0530 Subject: [Freeswitch-users] How to run IVR application In-Reply-To: <47d63d920911260438j29b56ee5w587bd6315eb64c42@mail.gmail.com> References: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> <87f2f3b90911241955v4e726111ked993c8dbb556f99@mail.gmail.com> <47d63d920911260438j29b56ee5w587bd6315eb64c42@mail.gmail.com> Message-ID: <47d63d920911262159s4c61e51dsb8c74d9530bc2d80@mail.gmail.com> Hi MC, I have created won sample application yesterday, It was working fine. Today, I checked that my local ip has changed. so, I changed the domain(IP) name in sip-account settings in my x-lite configuration. After that x-lite is not able to register with FS. I am getting error like "Registration error 405 : Method not Allowed ". Could you please tell me ,why its happening ? Thanks in advance, Venkat. On Thu, Nov 26, 2009 at 6:08 PM, ovvenkat wrote: > Thank you very much MC . Its working :) . I started loving "FS" ;) > > On Wed, Nov 25, 2009 at 9:25 AM, Michael Collins wrote: > >> >> >> On Tue, Nov 24, 2009 at 6:03 PM, Lei Tang wrote: >> >>> you can do this in follow steps: >>> 1.edit default.xml diaplan config file in your fs config >>> directory(FS/conf/dialplan/default.xml), and section >>> >>> >>> >>> >>> >>> 2. edit your ivr script, your can refer to >>> http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in >>> lua. >>> 3. connect your sip phone to fs, and dial 114, this will launch your ivr >>> application >>> >>> >> >> You can also do IVRs with static XML. I recommend you try out the demo IVR >> by dialing 5000. Now go look at the two main files that we used to build >> that IVR: >> >> conf/autoload_configs/ivr.conf.xml (menu structure) >> conf/lang/en/demo/demo-ivr.xml (phrase macros) >> >> it's overwhelming at first, however once you get the hang of it you'll >> appreciate how powerful it is. The wiki and the sample XML config files have >> lots of information so be sure to read as much as you can and try things. >> You can't break anything. :) >> >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > If you have come to help me, you are wasting your time. > If you have come to because your liberation is bound up in mine, we can > work together. > > > Regards > Venkatesan OV. > -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/e7bbf853/attachment-0002.html From jonas.gauffin at gmail.com Thu Nov 26 22:09:53 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Fri, 27 Nov 2009 07:09:53 +0100 Subject: [Freeswitch-users] Problems with nat In-Reply-To: References: Message-ID: Ok. I've been running this system since FS was a beta. It stopped working after a update. I'll switch to a single profile. What NAT settings should it have? I really want to get rid of the RECOVERY_ON_TIMER_EXPIRE error. On Thu, Nov 26, 2009 at 9:44 PM, Michael Jerris wrote: > In this case you should not need 2 profiles either. > > On Nov 26, 2009, at 1:14 PM, Jonas Gauffin wrote: > > It's a windowsserver which is behind a router. > > Which profile should local-network-acl be specified on? > > When I bridge calls to the outside world, should I use > sofia/internal/@gateway or > sofia/external/@gateway? > > > On Thu, Nov 26, 2009 at 4:42 PM, Brian West wrote: > >> Are you doing this all on a linux box thats acting as your router too? If >> not you don't need two profiles... you also don't need to set the >> local-network-acl on ANY profile that isn't do anything with nat. >> >> /b >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/b245108c/attachment-0002.html From christian.loeschenkohl at xpirio.com Fri Nov 27 00:22:19 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Fri, 27 Nov 2009 09:22:19 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> <4B0B005E.4080202@xpirio.com> <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> Message-ID: <4B0F8C3B.4000800@xpirio.com> hello sorry, for my late reply my core debugging was at info not at debug, now it's changed and i have the log needed i'm sorry but pastebin doesn't work (it seems that my trace was to big) http://pastebin.freeswitch.org/11305 says "Query failure: Got a packet bigger than 'max_allowed_packet' bytes" i'll send the logfile personal to you, hope you don't dislike this br On 2009-11-24 01:48, Anthony Minessale wrote: > You forgot to set freeswitch to debug loglevel > > You need both of the following: > > console loglevel debug > sofia profile internal siptrace on > > > > > 2009/11/23 Christian L?schenkohl > > > sorry about wasting your time (wasn't my intent) > > the log is at http://pastebin.freeswitch.org/11240 > i called 5214448370068 (also other calls are in the log) > > they now have changed 180 to 183 on the sonus, but makes no > difference here > > br > > On 2009-11-23 22:07, Anthony Minessale wrote: > > do you have the ringback variable set on the channel? > > if so it will cause 180 to attempt to play inband ringback indication > > > > I have nothing left to say because I asked for the whole log with the > > siptrace enables not just 5 lines of it. > > If you still want help, give me the log to examine and I will > tell you > > what your problem is. > > > > > > > > 2009/11/23 Christian L?schenkohl > > > >> > > > > thany ou for your answer > > > > we use g729 on all our other connections in passthrough mode > and it > > also doesn't work with alaw. > > so i don't think it's related to this. > > > > br > > > > > > On 2009-11-23 20:48, Brian West wrote: > > > Well its also G729 so I suspect you don't have G729 > > > > > > /b > > > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > > > >> hi > > >> > > >> our freeswitch server has to talk to a sonus ip-switch > > >> when we want to setup a call we do get a "100 Trying" and then a > > >> "180 Ringing" > > >> within the "180 Ringing" we get a sdp with "a=sendonly" then our > > >> freeswitch > > >> quits with a CANCEL message. > > >> i simply don't get why our freeswitch aborts the session - i think > > >> it would work > > >> if no "a=sendonly" would be present in the sdp. > > >> > > >> my technical contact doesn't want to switch 180 to 183 on the > sonus > > >> side - this would > > >> also work (i think). in fact he says that 180 ringing is vaild, he > > >> isn't that wrong in > > >> this case. > > >> > > >> our freeswitch works in proxy mode, we do use trunk 15396 > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > > >> > > >> 92.63.208.36 - freeswitch > > >> 38.105.229.100 - sonus > > >> > > >> br > > >> > > >> -- > > >> Ing. Christian L?schenkohl > > >> Technische Leitung, Forschung& Entwicklung VoIP > > >> > > >> xpirio > > >> Telekommunikation& Service GmbH > > >> Lakeside B04 > > >> 9020 Klagenfurt > > >> Austria > > >> > > >> T +43 (0) 5 77 11 - 1000 > > >> F +43 (0) 5 77 11 - 1002 > > >> E christian.loeschenkohl at xpirio.com > > > > > > >> > > >> _______________________________________________ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > >> users > > >> http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung & Entwicklung VoIP > > > > xpirio > > Telekommunikation & Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From abeka at greatiam.com Fri Nov 27 00:33:06 2009 From: abeka at greatiam.com (Otis) Date: Fri, 27 Nov 2009 08:33:06 +0000 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <92e7d2090911260717j11ffad78kdd11b1c87dfd87be@mail.gmail.com> References: <221275.23339.qm@web56403.mail.re3.yahoo.com> <4B0E7384.5010809@greatiam.com> <92e7d2090911260717j11ffad78kdd11b1c87dfd87be@mail.gmail.com> Message-ID: <4B0F8EC2.2080609@greatiam.com> Yes. I ventured to use that and got some error in connecting to the mysql database. Will try with the default sqlite before getting adventurous again. Thanks Addison Martin wrote: > Fedora and Centos installation instructions are very similar. You > should be able to compile on Fedora without any problems that I'm > aware of. > > Regards, > > Nik > > > > On Thu, Nov 26, 2009 at 06:24, Otis > wrote: > > Thanks. I will try it . I am on Fedora 11 > > > > Mark Crane wrote: > > "how about trying Fusionpbx.com ( GUI)" -Ram > > > > I'll second that! I released FusionPBX 1.0 RC5 today. I thought > it was > > ready to release now but decided to do one more release > candidate just > > to be sure. This should be the last release candidate before the > > release of version 1.0. > > > > The final release may be by the end of the week as long as no major > > issues are found. > > > > http://fusionpbx.com > > > > > > > > > > --- On *Mon, 11/23/09, ram / >/* wrote: > > > > > > From: ram > > > Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > > To: freeswitch-users at lists.freeswitch.org > > > Date: Monday, November 23, 2009, 10:54 PM > > > > > > > > On Mon, Nov 23, 2009 at 10:37 AM, Otis > > >> wrote: > > > > Thanks. > > > > I have to get a centos box I guess. > > > > Much appreciated > > > > Samuel 'Otis' > > > > > > > > how about trying Fusionpbx.com ( GUI) > > > > Ram > > > > -----Inline Attachment Follows----- > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From leon at scarlet-internet.nl Fri Nov 27 00:36:38 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Fri, 27 Nov 2009 09:36:38 +0100 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: <65d96fc80911261348p18c1d021of3b6500ff798f345@mail.gmail.com> References: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> <0655757B-61CA-490D-BDB0-873263555575@jerris.com> <65d96fc80911261348p18c1d021of3b6500ff798f345@mail.gmail.com> Message-ID: <1FCBD543-07E9-4BCA-B650-D93F0F96D6C4@scarlet-internet.nl> There's a little info here on how to enable it with odbc: http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 regards, Leon On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: > > On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris > wrote: > http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- > news-3-51-18.html > > MySQL Connector/ODBC now supports batched statements. In order to > enable > cached statement support you must switch enable the batched > statement option (FLAG_MULTI_STATEMENTS, > 67108864, or Allow multiple statements > within a GUI configuration). Be aware that batched statements > create an increased chance of SQL injection attacks and you > must > ensure that your application protects against this scenario. > (Bug#7445) > > > so, is this the right patch ? > > http://bugs.mysql.com/file.php?id=6994 > > > T. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/3b2fdda4/attachment-0002.html From abeka at greatiam.com Fri Nov 27 00:37:09 2009 From: abeka at greatiam.com (Otis) Date: Fri, 27 Nov 2009 08:37:09 +0000 Subject: [Freeswitch-users] Re-routing calls to PSTN In-Reply-To: <4B0F50B5.6030804@aktzero.com> References: <4B0E6034.6050802@greatiam.com> <4B0F50B5.6030804@aktzero.com> Message-ID: <4B0F8FB5.2030602@greatiam.com> Thank you very much . Please what are you calling a hard line ? Andrew Thompson wrote: >
On > 11/26/2009 6:02 AM, Otis wrote: >> Can I get FS to re-route incoming-calls to PSTN. If this has been >> raised before could someone direct me to URL or link please > > Since I don't have a hard line, I do something like: > > > > > data="sofia/gateway/YOURPROVIDER/18005551212"/> > > > > From zolotov at altron.ua Fri Nov 27 00:49:53 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Fri, 27 Nov 2009 10:49:53 +0200 Subject: [Freeswitch-users] Re-routing calls to PSTN In-Reply-To: <4B0F8FB5.2030602@greatiam.com> References: <4B0E6034.6050802@greatiam.com> <4B0F50B5.6030804@aktzero.com> <4B0F8FB5.2030602@greatiam.com> Message-ID: <4B0F92B1.2010601@altron.ua> Please try this: Otis ?????: > Thank you very much . Please what are you calling a hard line ? > > > > Andrew Thompson wrote: > >>
On >> 11/26/2009 6:02 AM, Otis wrote: >> >>> Can I get FS to re-route incoming-calls to PSTN. If this has been >>> raised before could someone direct me to URL or link please >>> >> Since I don't have a hard line, I do something like: >> >> >> >> >> > data="sofia/gateway/YOURPROVIDER/18005551212"/> >> >> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steve.kurzeja at gmail.com Fri Nov 27 02:00:49 2009 From: steve.kurzeja at gmail.com (Steve Kurzeja) Date: Fri, 27 Nov 2009 23:00:49 +1300 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: <191c3a030911251310h9f8bd1epf0d445c746e968a5@mail.gmail.com> References: <191c3a030911251310h9f8bd1epf0d445c746e968a5@mail.gmail.com> Message-ID: <5f7152000911270200o4cb73541ud05136bd02866447@mail.gmail.com> Is that USD ? :) On Thu, Nov 26, 2009 at 10:10 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I can spare you the pain and let you know outright that this sort of > functionality will cost somewhere in the range of 125,000.00 to 150,000.00 > to properly implement by assembling a team of consultants including members > of the development team from both FreeSWITCH and Sofia-SIP and even if you > have the money, finding the time to implement it would also be a factor as > it's a few thousand man-hours of work. > > > On Wed, Nov 25, 2009 at 3:01 PM, Mathieu Rene wrote: > >> You can read all about the sip library at >> http://sofia-sip.sourceforge.net/refdocs/ >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 25-Nov-09, at 3:58 PM, srinivasula reddy wrote: >> >> thanks for your reply mike, >> is there any api in freeswitch or any thing else to update lib >> programatically from pjsua. >> >> srinivas >> >> On Thu, Nov 26, 2009 at 2:05 AM, Michael Jerris wrote: >> >>> "something that is not available in that lib at this time." >>> >>> Mike >>> >>> On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote >>> >>> can please tell me how can i exchange session state into sip library. >>> >>> Thanks >>> srinivas >>> >>> On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: >>> >>>> For that you would need to fully exchange session state into the sip >>>> library, *something that is not available in that lib at this time.* >>>> >>>> >>>> On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: >>>> >>>> HI, >>>> thanks for your reply, my requirement is i am doing failover stuff with >>>> freeswitch. i dont want cut the calls when freeswitch dies, when failover >>>> happens mean one freeswitch dies we are going to start the second >>>> freeswitch, i dont want close call intiated by the first freeswtich, they >>>> are communicating with meida(bypass media). when one endpoing try to end the >>>> call at that time i want to close the call for the other end also. >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Srinivasula Reddy K >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Srinivasula Reddy K >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/f41f7967/attachment-0002.html From abeka at greatiam.com Fri Nov 27 03:15:08 2009 From: abeka at greatiam.com (Otis) Date: Fri, 27 Nov 2009 11:15:08 +0000 Subject: [Freeswitch-users] Connecting Multiple domains Message-ID: <4B0FB4BC.3090204@greatiam.com> Could someone please direct me to a link for connecting multiple say 2 domains each with their own FS server. Thanks From tculjaga at gmail.com Fri Nov 27 04:19:09 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 27 Nov 2009 13:19:09 +0100 Subject: [Freeswitch-users] Bypass_media and re_invite In-Reply-To: <5f7152000911270200o4cb73541ud05136bd02866447@mail.gmail.com> References: <191c3a030911251310h9f8bd1epf0d445c746e968a5@mail.gmail.com> <5f7152000911270200o4cb73541ud05136bd02866447@mail.gmail.com> Message-ID: <65d96fc80911270419x5935d50cpcae48ca8dfa1ee92@mail.gmail.com> On Fri, Nov 27, 2009 at 11:00 AM, Steve Kurzeja wrote: > Is that USD ? :) > > i believe these are not Turkish liras :P -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/fddb6411/attachment-0002.html From tculjaga at gmail.com Fri Nov 27 04:21:55 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 27 Nov 2009 13:21:55 +0100 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: <1FCBD543-07E9-4BCA-B650-D93F0F96D6C4@scarlet-internet.nl> References: <4EECDCF0EC9A4940933AF660F9F5587B@ws4> <0655757B-61CA-490D-BDB0-873263555575@jerris.com> <65d96fc80911261348p18c1d021of3b6500ff798f345@mail.gmail.com> <1FCBD543-07E9-4BCA-B650-D93F0F96D6C4@scarlet-internet.nl> Message-ID: <65d96fc80911270421y5a2c6cb6n90d1f48fccb23a4c@mail.gmail.com> On Fri, Nov 27, 2009 at 9:36 AM, Leon de Rooij wrote: > There's a little info here on how to enable it with odbc: > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 > > regards, > > Leon > > well i was centanly blind when i asked this :P [maxpowersoft_odbc] Driver = MySQL SERVER = localhost PORT = 3306 DATABASE = myDatabase *OPTIONS = 67108864* Socket = /var/lib/mysql/mysql.sock Thanks. T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/c326afdc/attachment-0002.html From frank at impactfax.com Fri Nov 27 06:41:44 2009 From: frank at impactfax.com (Frank @ Impact) Date: Fri, 27 Nov 2009 09:41:44 -0500 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: <1FCBD543-07E9-4BCA-B650-D93F0F96D6C4@scarlet-internet.nl> Message-ID: Thanks. But when I made these entries in /etc/odbc.ini and rebooted. [freeswitch] Driver = MySQL SERVER = 127.0.0.1 PORT = 4040 DATABASE = mydb OPTIONS = 67108864 .I still get FS complaining with this. Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched statements!#012If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS FreeSWITCH>version FreeSWITCH Version 1.0.trunk (15660) Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 EST 2009 x86_64 x86_64 x86_64 GNU/Linux >From /etc/odbcinst.ini DRIVER = /usr/lib64/libmyodbc5-5.1.5.so Setup = /usr/lib64/libodbcmyS.so Is this a FS issue ? or an issue with mysql odbc? Any insight would be great. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Leon de Rooij Sent: Friday, November 27, 2009 3:37 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS There's a little info here on how to enable it with odbc: http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 regards, Leon On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris wrote: http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445 ) so, is this the right patch ? http://bugs.mysql.com/file.php?id=6994 T. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/78f2b0c3/attachment-0002.html From leon at scarlet-internet.nl Fri Nov 27 07:19:03 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Fri, 27 Nov 2009 16:19:03 +0100 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: References: Message-ID: Are you using the myodbc 3.51.18 version or higher ? I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to upgrade from jaunty.. regards, Leon On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: > Thanks. But when I made these entries in /etc/odbc.ini and rebooted? > > [freeswitch] > Driver = MySQL > SERVER = 127.0.0.1 > PORT = 4040 > DATABASE = mydb > OPTIONS = 67108864 > > ?I still get FS complaining with this. > > Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 > [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched > statements!#012If you are using mysql, make sure you are using > MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS > > FreeSWITCH>version > FreeSWITCH Version 1.0.trunk (15660) > > Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 > EST 2009 x86_64 x86_64 x86_64 GNU/Linux > > From /etc/odbcinst.ini > DRIVER = /usr/lib64/libmyodbc5-5.1.5.so > Setup = /usr/lib64/libodbcmyS.so > > Is this a FS issue ? or an issue with mysql odbc? Any insight > would be great. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Leon de Rooij > Sent: Friday, November 27, 2009 3:37 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > > There's a little info here on how to enable it with odbc: > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 > > regards, > > Leon > > > On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: > > > > On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris > wrote: > http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- > news-3-51-18.html > > MySQL Connector/ODBC now supports batched statements. In order to > enable > cached statement support you must switch enable the batched > statement option (FLAG_MULTI_STATEMENTS, > 67108864, or Allow multiple statements > within a GUI configuration). Be aware that batched statements > create an increased chance of SQL injection attacks and you > must > ensure that your application protects against this scenario. > (Bug#7445) > > > so, is this the right patch ? > > http://bugs.mysql.com/file.php?id=6994 > > > T. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/9c86b324/attachment-0002.html From frank at impactfax.com Fri Nov 27 07:36:55 2009 From: frank at impactfax.com (Frank @ Impact) Date: Fri, 27 Nov 2009 10:36:55 -0500 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: Message-ID: Yes. I am using version 5.1 I am using Fedora 12. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Leon de Rooij Sent: Friday, November 27, 2009 10:19 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS Are you using the myodbc 3.51.18 version or higher ? I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to upgrade from jaunty.. regards, Leon On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: Thanks. But when I made these entries in /etc/odbc.ini and rebooted. [freeswitch] Driver = MySQL SERVER = 127.0.0.1 PORT = 4040 DATABASE = mydb OPTIONS = 67108864 .I still get FS complaining with this. Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched statements!#012If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS FreeSWITCH>version FreeSWITCH Version 1.0.trunk (15660) Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 EST 2009 x86_64 x86_64 x86_64 GNU/Linux >From /etc/odbcinst.ini DRIVER = /usr/lib64/libmyodbc5-5.1.5.so Setup = /usr/lib64/libodbcmyS.so Is this a FS issue ? or an issue with mysql odbc? Any insight would be great. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Leon de Rooij Sent: Friday, November 27, 2009 3:37 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS There's a little info here on how to enable it with odbc: http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 regards, Leon On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris wrote: http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445 ) so, is this the right patch ? http://bugs.mysql.com/file.php?id=6994 T. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/1bb30a5f/attachment-0002.html From andrewkt at aktzero.com Fri Nov 27 08:34:55 2009 From: andrewkt at aktzero.com (Andrew Thompson) Date: Fri, 27 Nov 2009 11:34:55 -0500 Subject: [Freeswitch-users] Re-routing calls to PSTN In-Reply-To: <4B0F8FB5.2030602@greatiam.com> References: <4B0E6034.6050802@greatiam.com> <4B0F50B5.6030804@aktzero.com> <4B0F8FB5.2030602@greatiam.com> Message-ID: <4B0FFFAF.6010700@aktzero.com> On 11/27/2009 3:37 AM, Otis wrote: > Thank you very much . Please what are you calling a hard line ? A real honest to goodness POTS line within 20 feet of and attached to my FS server. My calls come in and go out SIP, so If I was sending an inbound call back to the PSTN, I'd just route it back out through my SIP provider. -- Andrew Thompson From andrewkt at aktzero.com Fri Nov 27 08:44:51 2009 From: andrewkt at aktzero.com (Andrew Thompson) Date: Fri, 27 Nov 2009 11:44:51 -0500 Subject: [Freeswitch-users] FS missing in action (was: How to run IVR application) In-Reply-To: <47d63d920911262159s4c61e51dsb8c74d9530bc2d80@mail.gmail.com> References: <47d63d920911240449y2f4e0923q6b5186ef57434690@mail.gmail.com> <50c41b4e0911241803x561a7995m6536cfe1af51f68d@mail.gmail.com> <87f2f3b90911241955v4e726111ked993c8dbb556f99@mail.gmail.com> <47d63d920911260438j29b56ee5w587bd6315eb64c42@mail.gmail.com> <47d63d920911262159s4c61e51dsb8c74d9530bc2d80@mail.gmail.com> Message-ID: <4B100203.4090502@aktzero.com> On 11/27/2009 12:59 AM, ovvenkat wrote: > Hi MC, > > I have created won sample application yesterday, It was working fine. > Today, I checked that my local ip has changed. so, I changed the > domain(IP) name in sip-account settings in my x-lite configuration. > After that x-lite is not able to register with FS. I am getting error > like "Registration error 405 : Method not Allowed ". Could you please > tell me ,why its happening ? Wait, what? First, don't re-use an existing thread, messages have a tendancy to get ignored/lost that way. Did the IP of your FS change, or of your PC? I would expect "local ip" to mean your DHCP'ed address from your Internet connection. That should have no bearing on the IP of your FS. Go find your FS and make sure *IT* is still on the IP you expect it to be on. If your PC running x-lite is also your FS, you may have other issues with IP address changing that I don't know how to handle, as I've only used static IPs for FS. (Or, try connecting via localhost, 127.0.0.1 instead, until you're ready to really start using FS.) -- Andrew Thompson From jbarou at sqli.com Fri Nov 27 08:47:23 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Fri, 27 Nov 2009 17:47:23 +0100 Subject: [Freeswitch-users] Transfer Problem Message-ID: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> Hi everybody, I'm actually using the lastest version of Freeswitch, I have a problem. I have a trunk SIP with my PABX. There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) 2. one Alcatel IpTouch 4028 with number 987 (on PABX) 3. one Siemens Gigaset A580 IP with number 8401 (on Freeswitch) *The first test* is to say to the phone 2 to transfer all the call to number 8401. So when I dial 987 on the phone 1, all work perfectly, the phone 3 is ringing and it's work. I have that in the log : 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending early media 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP [sofia/internal/368 at 10.33.69.246] 10.33.169.92 port 23054 -> 10.33.69.246 port 32000 codec: 8 ms: 90 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 720 bytes per 90ms 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: v=0 o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 s=FreeSWITCH c=IN IP4 10.33.169.92 t=0 0 m=audio 23054 RTP/AVP 8 106 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:90 a=sendrecv 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer sofia/internal/368 at 10.33.69.246! 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ 368 at 10.33.69.246 skipping state [early][183] 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal sofia/internal/368 at 10.33.69.246 [BREAK] 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec Activation Success L16 at 8000hz 1 channel 90ms 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ 368 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port confirmed. 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read Buffer at 1440 bytes vs 81 *The Second Tes*t is to say to the phone 1 to transfer all the call to number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just one time and after it hangup. I have that in the log : 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending early media 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP [sofia/internal/987 at 10.33.69.246] 10.33.169.92 port 27732 -> 10.33.69.144 port 32000 codec: 8 ms: 90 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 720 bytes per 90ms 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: v=0 o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 s=FreeSWITCH c=IN IP4 10.33.169.92 t=0 0 m=audio 27732 RTP/AVP 8 106 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:90 a=sendrecv 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer sofia/internal/987 at 10.33.69.246! 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ 987 at 10.33.69.246 skipping state [early][183] 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal sofia/internal/987 at 10.33.69.246 [BREAK] 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec Activation Success L16 at 8000hz 1 channel 90ms 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ 987 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ 987 at 10.33.69.246 entering state [terminated][487] 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ 987 at 10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/987 at 10.33.69.246 [KILL] 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/987 at 10.33.69.246 [BREAK] 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 thread mismatch skipping state handler. 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [KILL] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change CS_HANGUP 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ sip:8401 at 10.33.170.231:5060 Overriding SIP cause 487 with 487 from the other leg 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ sip:8401 at 10.33.170.231:5060 hanging up, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to sofia/internal/sip:8401 at 10.33.170.231:5060 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:8401 at 10.33.170.231:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_HANGUP -> CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:8401 at 10.33.170.231:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_REPORTING -> CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 (sofia/internal/sip:8401 at 10.33.170.231:5060) Locked, Waiting on external entities 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 thread mismatch skipping state handler. 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 48 (sofia/internal/sip:8401 at 10.33.170.231:5060) Ended 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close Channel sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_DESTROY] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ sip:8401 at 10.33.170.231:5060 SOFIA DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:8401 at 10.33.170.231:5060 Standard DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY going to sleep 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. Cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/987 at 10.33.69.246) State EXECUTE going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/987 at 10.33.69.246) Running State Change CS_HANGUP 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/987 at 10.33.69.246) State HANGUP 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ 987 at 10.33.69.246 Overriding SIP cause 487 with 487 from the other leg 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ 987 at 10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 sofia/internal/987 at 10.33.69.246 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/987 at 10.33.69.246) State HANGUP going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/987 at 10.33.69.246) State Change CS_HANGUP -> CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/987 at 10.33.69.246 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/987 at 10.33.69.246) Running State Change CS_REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/987 at 10.33.69.246) State REPORTING 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 sofia/internal/987 at 10.33.69.246 Standard REPORTING, cause: ORIGINATOR_CANCEL 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 (sofia/internal/987 at 10.33.69.246) State REPORTING going to sleep 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/987 at 10.33.69.246) State Change CS_REPORTING -> CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal sofia/internal/987 at 10.33.69.246 [BREAK] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 (sofia/internal/987 at 10.33.69.246) Locked, Waiting on external entities 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 47 (sofia/internal/987 at 10.33.69.246) Ended 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close Channel sofia/internal/987 at 10.33.69.246 [CS_DESTROY] 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/987 at 10.33.69.246) Running State Change CS_DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/987 at 10.33.69.246) State DESTROY 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ 987 at 10.33.69.246 SOFIA DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 sofia/internal/987 at 10.33.69.246 Standard DESTROY 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/987 at 10.33.69.246) State DESTROY going to sleep Finally when I tried to call the phone 3 with the phone 1 it's working, and not when I want to call the phone 3 with the phone 2, like just before, it's ringing just one time and hangup. Thanks you. Best Regards -- John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/316a1bfc/attachment-0002.html From samuelmukoti at gmail.com Fri Nov 27 08:45:51 2009 From: samuelmukoti at gmail.com (Samuel Mukoti) Date: Fri, 27 Nov 2009 18:45:51 +0200 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: References: Message-ID: Hi, Any recommendations for apps that can I use ontop of freeswitch as a GUI manager, to manage extensions, queues, ivr, and dialplans? Thanks Sam On 27 Nov,2009, at 5:19 PM, freeswitch-users-request at lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > Today's Topics: > > 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 27 Nov 2009 16:19:03 +0100 > From: Leon de Rooij > Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="windows-1252" > > Are you using the myodbc 3.51.18 version or higher ? > > I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to > upgrade from jaunty.. > > regards, > > Leon > > > On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: > >> Thanks. But when I made these entries in /etc/odbc.ini and rebooted? >> >> [freeswitch] >> Driver = MySQL >> SERVER = 127.0.0.1 >> PORT = 4040 >> DATABASE = mydb >> OPTIONS = 67108864 >> >> ?I still get FS complaining with this. >> >> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 >> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched >> statements!#012If you are using mysql, make sure you are using >> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS >> >> FreeSWITCH>version >> FreeSWITCH Version 1.0.trunk (15660) >> >> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 >> EST 2009 x86_64 x86_64 x86_64 GNU/Linux >> >> From /etc/odbcinst.ini >> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so >> Setup = /usr/lib64/libodbcmyS.so >> >> Is this a FS issue ? or an issue with mysql odbc? Any insight >> would be great. >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Leon de Rooij >> Sent: Friday, November 27, 2009 3:37 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >> >> There's a little info here on how to enable it with odbc: >> >> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 >> >> regards, >> >> Leon >> >> >> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: >> >> >> >> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris >> wrote: >> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- >> news-3-51-18.html >> >> MySQL Connector/ODBC now supports batched statements. In order to >> enable >> cached statement support you must switch enable the batched >> statement option (FLAG_MULTI_STATEMENTS, >> 67108864, or Allow multiple statements >> within a GUI configuration). Be aware that batched statements >> create an increased chance of SQL injection attacks and you >> must >> ensure that your application protects against this scenario. >> (Bug#7445) >> >> >> so, is this the right patch ? >> >> http://bugs.mysql.com/file.php?id=6994 >> >> >> T. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/9c86b324/attachment.html > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > End of FreeSWITCH-users Digest, Vol 41, Issue 209 > ************************************************* From anthony.minessale at gmail.com Fri Nov 27 08:57:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 10:57:06 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <4B0F8C3B.4000800@xpirio.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> <4B0B005E.4080202@xpirio.com> <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> <4B0F8C3B.4000800@xpirio.com> Message-ID: <191c3a030911270857j697738f9n99e9e5bb1b71c38@mail.gmail.com> or you can put it at a url on your web site and just post a link 2009/11/27 Christian L?schenkohl > hello > > sorry, for my late reply > my core debugging was at info not at debug, now it's changed and i have the > log needed > > i'm sorry but pastebin doesn't work (it seems that my trace was to big) > http://pastebin.freeswitch.org/11305 says "Query failure: Got a packet > bigger than 'max_allowed_packet' bytes" > > i'll send the logfile personal to you, hope you don't dislike this > > br > > On 2009-11-24 01:48, Anthony Minessale wrote: > > You forgot to set freeswitch to debug loglevel > > > > You need both of the following: > > > > console loglevel debug > > sofia profile internal siptrace on > > > > > > > > > > 2009/11/23 Christian L?schenkohl > > > > > > sorry about wasting your time (wasn't my intent) > > > > the log is at http://pastebin.freeswitch.org/11240 > > i called 5214448370068 (also other calls are in the log) > > > > they now have changed 180 to 183 on the sonus, but makes no > > difference here > > > > br > > > > On 2009-11-23 22:07, Anthony Minessale wrote: > > > do you have the ringback variable set on the channel? > > > if so it will cause 180 to attempt to play inband ringback > indication > > > > > > I have nothing left to say because I asked for the whole log with > the > > > siptrace enables not just 5 lines of it. > > > If you still want help, give me the log to examine and I will > > tell you > > > what your problem is. > > > > > > > > > > > > 2009/11/23 Christian L?schenkohl > > > > > > > >> > > > > > > thany ou for your answer > > > > > > we use g729 on all our other connections in passthrough mode > > and it > > > also doesn't work with alaw. > > > so i don't think it's related to this. > > > > > > br > > > > > > > > > On 2009-11-23 20:48, Brian West wrote: > > > > Well its also G729 so I suspect you don't have G729 > > > > > > > > /b > > > > > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > > > > > >> hi > > > >> > > > >> our freeswitch server has to talk to a sonus ip-switch > > > >> when we want to setup a call we do get a "100 Trying" and then > a > > > >> "180 Ringing" > > > >> within the "180 Ringing" we get a sdp with "a=sendonly" then > our > > > >> freeswitch > > > >> quits with a CANCEL message. > > > >> i simply don't get why our freeswitch aborts the session - i > think > > > >> it would work > > > >> if no "a=sendonly" would be present in the sdp. > > > >> > > > >> my technical contact doesn't want to switch 180 to 183 on the > > sonus > > > >> side - this would > > > >> also work (i think). in fact he says that 180 ringing is vaild, > he > > > >> isn't that wrong in > > > >> this case. > > > >> > > > >> our freeswitch works in proxy mode, we do use trunk 15396 > > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > > > >> > > > >> 92.63.208.36 - freeswitch > > > >> 38.105.229.100 - sonus > > > >> > > > >> br > > > >> > > > >> -- > > > >> Ing. Christian L?schenkohl > > > >> Technische Leitung, Forschung& Entwicklung VoIP > > > >> > > > >> xpirio > > > >> Telekommunikation& Service GmbH > > > >> Lakeside B04 > > > >> 9020 Klagenfurt > > > >> Austria > > > >> > > > >> T +43 (0) 5 77 11 - 1000 > > > >> F +43 (0) 5 77 11 - 1002 > > > >> E christian.loeschenkohl at xpirio.com > > > > > > > > > > >> > > > >> _______________________________________________ > > > >> FreeSWITCH-users mailing list > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > > >> users > > > >> http://www.freeswitch.org > > > > > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > -- > > > Ing. Christian L?schenkohl > > > Technische Leitung, Forschung & Entwicklung VoIP > > > > > > xpirio > > > Telekommunikation & Service GmbH > > > Lakeside B04 > > > 9020 Klagenfurt > > > Austria > > > > > > T +43 (0) 5 77 11 - 1000 > > > F +43 (0) 5 77 11 - 1002 > > > E christian.loeschenkohl at xpirio.com > > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > > > > > > > > >> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > > > > > >> > > > IRC: irc.freenode.net > > #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > > > > > > > > >> > > > iax:guest at conference.freeswitch.org/888 > > > > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > > > > > >> > > > pstn:213-799-1400 > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung & Entwicklung VoIP > > > > xpirio > > Telekommunikation & Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/f8497a2d/attachment-0002.html From anthony.minessale at gmail.com Fri Nov 27 09:03:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 11:03:10 -0600 Subject: [Freeswitch-users] Transfer Problem In-Reply-To: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> References: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> Message-ID: <191c3a030911270903i341d1f83pa15f67443422cb67@mail.gmail.com> by latest do you mean SVN trunk? Can you issue the command "sofia profile internal siptrace on" before capturing your trace and post the results to http://pastebin.freeswitch.org or open a jira http://jira.freeswitch.orgon the issue and attach the log after you create the issue ticket, don't include it in the mailing list. On Fri, Nov 27, 2009 at 10:47 AM, Jonathan Barou wrote: > Hi everybody, > > I'm actually using the lastest version of Freeswitch, I have a problem. I > have a trunk SIP with my PABX. > > There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) > 2. one Alcatel IpTouch 4028 with number 987 > (on PABX) > 3. one Siemens Gigaset A580 IP with number > 8401 (on Freeswitch) > > > *The first test* is to say to the phone 2 to transfer all the call to > number 8401. So when I dial 987 on the phone 1, all work perfectly, the > phone 3 is ringing and it's work. I have that in the log : > > 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending early > media > > 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP > [sofia/internal/368 at 10.33.69.246] 10.33.169.92 port 23054 -> 10.33.69.246 > port 32000 codec: 8 ms: 90 > > 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer [soft] > 720 bytes per 90ms > > 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: > > v=0 > > o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 > > s=FreeSWITCH > > c=IN IP4 10.33.169.92 > > t=0 0 > > m=audio 23054 RTP/AVP 8 106 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:106 telephone-event/8000 > > a=fmtp:106 0-16 > > a=silenceSupp:off - - - - > > a=ptime:90 > > a=sendrecv > > > 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer > sofia/internal/368 at 10.33.69.246! > > 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] > > 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ > sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] > > 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ > 368 at 10.33.69.246 skipping state [early][183] > > 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal > sofia/internal/368 at 10.33.69.246 [BREAK] > > 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec > Activation Success L16 at 8000hz 1 channel 90ms > > 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play > Ringback Tone [%(2000,4000,440.0,480.0)] > > 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ > 368 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] > > 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port > confirmed. > > 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read > Buffer at 1440 bytes vs 81 > > > > *The Second Tes*t is to say to the phone 1 to transfer all the call to > number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just > one time and after it hangup. I have that in the log : > > > 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending > early media > > 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP > [sofia/internal/987 at 10.33.69.246] 10.33.169.92 port 27732 -> 10.33.69.144 > port 32000 codec: 8 ms: 90 > > 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer [soft] > 720 bytes per 90ms > > 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: > > v=0 > > o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 > > s=FreeSWITCH > > c=IN IP4 10.33.169.92 > > t=0 0 > > m=audio 27732 RTP/AVP 8 106 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:106 telephone-event/8000 > > a=fmtp:106 0-16 > > a=silenceSupp:off - - - - > > a=ptime:90 > > a=sendrecv > > > 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer > sofia/internal/987 at 10.33.69.246! > > 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] > > 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ > sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] > > 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ > 987 at 10.33.69.246 skipping state [early][183] > > 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal > sofia/internal/987 at 10.33.69.246 [BREAK] > > 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec > Activation Success L16 at 8000hz 1 channel 90ms > > 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play > Ringback Tone [%(2000,4000,440.0,480.0)] > > 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ > 987 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] > > 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ > 987 at 10.33.69.246 entering state [terminated][487] > > 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ > 987 at 10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] > > 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal > sofia/internal/987 at 10.33.69.246 [KILL] > > 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/987 at 10.33.69.246 [BREAK] > > 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 thread > mismatch skipping state handler. > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore original > codec. > > 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup > sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [KILL] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change > CS_HANGUP > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ > sip:8401 at 10.33.170.231:5060 Overriding SIP cause 487 with 487 from the > other leg > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ > sip:8401 at 10.33.170.231:5060 hanging up, cause: ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to > sofia/internal/sip:8401 at 10.33.170.231:5060 > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/sip:8401 at 10.33.170.231:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP going to sleep > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_HANGUP -> > CS_REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change > CS_REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/sip:8401 at 10.33.170.231:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING going to > sleep > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_REPORTING -> > CS_DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 > (sofia/internal/sip:8401 at 10.33.170.231:5060) Locked, Waiting on external > entities > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 thread > mismatch skipping state handler. > > 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate > Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > > 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 48 > (sofia/internal/sip:8401 at 10.33.170.231:5060) Ended > > 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close > Channel sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_DESTROY] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change > CS_DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ > sip:8401 at 10.33.170.231:5060 SOFIA DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/sip:8401 at 10.33.170.231:5060 Standard DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY going to sleep > > 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot create > outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > > 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. > Cause: ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/987 at 10.33.69.246) State EXECUTE going to sleep > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/987 at 10.33.69.246) Running State Change CS_HANGUP > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 > (sofia/internal/987 at 10.33.69.246) State HANGUP > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ > 987 at 10.33.69.246 Overriding SIP cause 487 with 487 from the other leg > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ > 987 at 10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/987 at 10.33.69.246 Standard HANGUP, cause: ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 > (sofia/internal/987 at 10.33.69.246) State HANGUP going to sleep > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/987 at 10.33.69.246) State Change CS_HANGUP -> CS_REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/987 at 10.33.69.246 [BREAK] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/987 at 10.33.69.246) Running State Change CS_REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 > (sofia/internal/987 at 10.33.69.246) State REPORTING > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/987 at 10.33.69.246 Standard REPORTING, cause: > ORIGINATOR_CANCEL > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 > (sofia/internal/987 at 10.33.69.246) State REPORTING going to sleep > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/987 at 10.33.69.246) State Change CS_REPORTING -> CS_DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal > sofia/internal/987 at 10.33.69.246 [BREAK] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 > (sofia/internal/987 at 10.33.69.246) Locked, Waiting on external entities > > 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 47 > (sofia/internal/987 at 10.33.69.246) Ended > > 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close > Channel sofia/internal/987 at 10.33.69.246 [CS_DESTROY] > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/987 at 10.33.69.246) Running State Change CS_DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/987 at 10.33.69.246) State DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ > 987 at 10.33.69.246 SOFIA DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/987 at 10.33.69.246 Standard DESTROY > > 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/987 at 10.33.69.246) State DESTROY going to sleep > > Finally when I tried to call the phone 3 with the phone 1 it's working, and > not when I want to call the phone 3 with the phone 2, like just before, it's > ringing just one time and hangup. > > > Thanks you. > > > Best Regards > > -- > John > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/d11148c7/attachment-0002.html From abeka at greatiam.com Fri Nov 27 10:42:51 2009 From: abeka at greatiam.com (Otis) Date: Fri, 27 Nov 2009 18:42:51 +0000 Subject: [Freeswitch-users] Re-routing calls to PSTN In-Reply-To: <4B0FFFAF.6010700@aktzero.com> References: <4B0E6034.6050802@greatiam.com> <4B0F50B5.6030804@aktzero.com> <4B0F8FB5.2030602@greatiam.com> <4B0FFFAF.6010700@aktzero.com> Message-ID: <4B101DAB.4040500@greatiam.com> Ok. Thanks Andrew Thompson wrote: >
On > 11/27/2009 3:37 AM, Otis wrote: >> Thank you very much . Please what are you calling a hard line ? > A real honest to goodness POTS line within 20 feet of and attached to > my FS server. > > My calls come in and go out SIP, so If I was sending an inbound call > back to the PSTN, I'd just route it back out through my SIP provider. > From abeka at greatiam.com Fri Nov 27 10:48:35 2009 From: abeka at greatiam.com (Otis) Date: Fri, 27 Nov 2009 18:48:35 +0000 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: References: Message-ID: <4B101F03.5090802@greatiam.com> Hi I am no sure but read up on fusionpbx. I asked the same question and someone pointed me to that. check web site Regards Samuel Mukoti wrote: >
Hi, > > Any recommendations for apps that can I use ontop of freeswitch as a > GUI manager, to manage extensions, queues, ivr, and dialplans? > > Thanks > > Sam > > > On 27 Nov,2009, at 5:19 PM, > freeswitch-users-request at lists.freeswitch.org wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Fri, 27 Nov 2009 16:19:03 +0100 >> From: Leon de Rooij >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset="windows-1252" >> >> Are you using the myodbc 3.51.18 version or higher ? >> >> I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to >> upgrade from jaunty.. >> >> regards, >> >> Leon >> >> >> On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: >> >>> Thanks. But when I made these entries in /etc/odbc.ini and rebooted? >>> >>> [freeswitch] >>> Driver = MySQL >>> SERVER = 127.0.0.1 >>> PORT = 4040 >>> DATABASE = mydb >>> OPTIONS = 67108864 >>> >>> ?I still get FS complaining with this. >>> >>> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 >>> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched >>> statements!#012If you are using mysql, make sure you are using >>> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS >>> >>> FreeSWITCH>version >>> FreeSWITCH Version 1.0.trunk (15660) >>> >>> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 >>> EST 2009 x86_64 x86_64 x86_64 GNU/Linux >>> >>> From /etc/odbcinst.ini >>> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so >>> Setup = /usr/lib64/libodbcmyS.so >>> >>> Is this a FS issue ? or an issue with mysql odbc? Any insight >>> would be great. >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Leon de Rooij >>> Sent: Friday, November 27, 2009 3:37 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >>> >>> There's a little info here on how to enable it with odbc: >>> >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 >>> >>> regards, >>> >>> Leon >>> >>> >>> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: >>> >>> >>> >>> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris >>> wrote: >>> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- >>> news-3-51-18.html >>> >>> MySQL Connector/ODBC now supports batched statements. In order to >>> enable >>> cached statement support you must switch enable the batched >>> statement option (FLAG_MULTI_STATEMENTS, >>> 67108864, or Allow multiple statements >>> within a GUI configuration). Be aware that batched statements >>> create an increased chance of SQL injection attacks and you >>> must >>> ensure that your application protects against this scenario. >>> (Bug#7445) >>> >>> >>> so, is this the right patch ? >>> >>> http://bugs.mysql.com/file.php?id=6994 >>> >>> >>> T. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/9c86b324/attachment.html >> >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 209 >> ************************************************* > > >
> From anthony.minessale at gmail.com Fri Nov 27 11:03:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 13:03:48 -0600 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <191c3a030911270857j697738f9n99e9e5bb1b71c38@mail.gmail.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> <4B0B005E.4080202@xpirio.com> <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> <4B0F8C3B.4000800@xpirio.com> <191c3a030911270857j697738f9n99e9e5bb1b71c38@mail.gmail.com> Message-ID: <191c3a030911271103s462910acu96da189b4064e63e@mail.gmail.com> please update to latest trunk 15698 or greater and re-test. The 183 from the provider had a sendonly attr that tricked the proxy code into thinking it was a hold/unhold operation. On Fri, Nov 27, 2009 at 10:57 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > or you can put it at a url on your web site and just post a link > > > 2009/11/27 Christian L?schenkohl > > hello >> >> sorry, for my late reply >> my core debugging was at info not at debug, now it's changed and i have >> the log needed >> >> i'm sorry but pastebin doesn't work (it seems that my trace was to big) >> http://pastebin.freeswitch.org/11305 says "Query failure: Got a packet >> bigger than 'max_allowed_packet' bytes" >> >> i'll send the logfile personal to you, hope you don't dislike this >> >> br >> >> On 2009-11-24 01:48, Anthony Minessale wrote: >> > You forgot to set freeswitch to debug loglevel >> > >> > You need both of the following: >> > >> > console loglevel debug >> > sofia profile internal siptrace on >> > >> > >> > >> > >> > 2009/11/23 Christian L?schenkohl > > > >> > >> > sorry about wasting your time (wasn't my intent) >> > >> > the log is at http://pastebin.freeswitch.org/11240 >> > i called 5214448370068 (also other calls are in the log) >> > >> > they now have changed 180 to 183 on the sonus, but makes no >> > difference here >> > >> > br >> > >> > On 2009-11-23 22:07, Anthony Minessale wrote: >> > > do you have the ringback variable set on the channel? >> > > if so it will cause 180 to attempt to play inband ringback >> indication >> > > >> > > I have nothing left to say because I asked for the whole log with >> the >> > > siptrace enables not just 5 lines of it. >> > > If you still want help, give me the log to examine and I will >> > tell you >> > > what your problem is. >> > > >> > > >> > > >> > > 2009/11/23 Christian L?schenkohl >> > > > >> > > > > >> >> > > >> > > thany ou for your answer >> > > >> > > we use g729 on all our other connections in passthrough mode >> > and it >> > > also doesn't work with alaw. >> > > so i don't think it's related to this. >> > > >> > > br >> > > >> > > >> > > On 2009-11-23 20:48, Brian West wrote: >> > > > Well its also G729 so I suspect you don't have G729 >> > > > >> > > > /b >> > > > >> > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: >> > > > >> > > >> hi >> > > >> >> > > >> our freeswitch server has to talk to a sonus ip-switch >> > > >> when we want to setup a call we do get a "100 Trying" and then >> a >> > > >> "180 Ringing" >> > > >> within the "180 Ringing" we get a sdp with "a=sendonly" then >> our >> > > >> freeswitch >> > > >> quits with a CANCEL message. >> > > >> i simply don't get why our freeswitch aborts the session - i >> think >> > > >> it would work >> > > >> if no "a=sendonly" would be present in the sdp. >> > > >> >> > > >> my technical contact doesn't want to switch 180 to 183 on the >> > sonus >> > > >> side - this would >> > > >> also work (i think). in fact he says that 180 ringing is >> vaild, he >> > > >> isn't that wrong in >> > > >> this case. >> > > >> >> > > >> our freeswitch works in proxy mode, we do use trunk 15396 >> > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 >> > > >> >> > > >> 92.63.208.36 - freeswitch >> > > >> 38.105.229.100 - sonus >> > > >> >> > > >> br >> > > >> >> > > >> -- >> > > >> Ing. Christian L?schenkohl >> > > >> Technische Leitung, Forschung& Entwicklung VoIP >> > > >> >> > > >> xpirio >> > > >> Telekommunikation& Service GmbH >> > > >> Lakeside B04 >> > > >> 9020 Klagenfurt >> > > >> Austria >> > > >> >> > > >> T +43 (0) 5 77 11 - 1000 >> > > >> F +43 (0) 5 77 11 - 1002 >> > > >> E christian.loeschenkohl at xpirio.com >> > >> > > > > > >> > > >> >> > > >> _______________________________________________ >> > > >> FreeSWITCH-users mailing list >> > > >> FreeSWITCH-users at lists.freeswitch.org >> > >> > > > > > >> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > > >> users >> > > >> http://www.freeswitch.org >> > > > >> > > > >> > > > _______________________________________________ >> > > > FreeSWITCH-users mailing list >> > > > FreeSWITCH-users at lists.freeswitch.org >> > >> > > > > > >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > > >> > > >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > > http://www.freeswitch.org >> > > >> > > -- >> > > Ing. Christian L?schenkohl >> > > Technische Leitung, Forschung & Entwicklung VoIP >> > > >> > > xpirio >> > > Telekommunikation & Service GmbH >> > > Lakeside B04 >> > > 9020 Klagenfurt >> > > Austria >> > > >> > > T +43 (0) 5 77 11 - 1000 >> > > F +43 (0) 5 77 11 - 1002 >> > > E christian.loeschenkohl at xpirio.com >> > >> > > > > > >> > > >> > > _______________________________________________ >> > > FreeSWITCH-users mailing list >> > > FreeSWITCH-users at lists.freeswitch.org >> > >> > > > > > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > > >> > > >> > > >> > > >> > > -- >> > > Anthony Minessale II >> > > >> > > FreeSWITCH http://www.freeswitch.org/ >> > > ClueCon http://www.cluecon.com/ >> > > Twitter: http://twitter.com/FreeSWITCH_wire >> > > >> > > AIM: anthm >> > > MSN:anthony_minessale at hotmail.com >> > >> > >> > > >> > >> >> >> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > >> > >> > > >> > >> >> >> > > IRC: irc.freenode.net >> > #freeswitch >> > > >> > > FreeSWITCH Developer Conference >> > > sip:888 at conference.freeswitch.org >> > >> > >> > > >> > >> >> >> > > iax:guest at conference.freeswitch.org/888 >> > >> > > >> > > googletalk:conf+888 at conference.freeswitch.org >> > >> > >> > > >> > >> >> >> > > pstn:213-799-1400 >> > > >> > > >> > > >> > >> ------------------------------------------------------------------------ >> > > >> > > _______________________________________________ >> > > FreeSWITCH-users mailing list >> > > FreeSWITCH-users at lists.freeswitch.org >> > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > >> > -- >> > Ing. Christian L?schenkohl >> > Technische Leitung, Forschung & Entwicklung VoIP >> > >> > xpirio >> > Telekommunikation & Service GmbH >> > Lakeside B04 >> > 9020 Klagenfurt >> > Austria >> > >> > T +43 (0) 5 77 11 - 1000 >> > F +43 (0) 5 77 11 - 1002 >> > E christian.loeschenkohl at xpirio.com >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > >> > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > >> > >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> > googletalk:conf+888 at conference.freeswitch.org >> > >> > >> > pstn:213-799-1400 >> > >> > >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> -- >> Ing. Christian L?schenkohl >> Technische Leitung, Forschung & Entwicklung VoIP >> >> xpirio >> Telekommunikation & Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenkohl at xpirio.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/14f15f32/attachment-0002.html From anthony.minessale at gmail.com Fri Nov 27 11:19:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 13:19:31 -0600 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> Message-ID: <191c3a030911271119k3f38a343k8351b121275580b9@mail.gmail.com> I told you to make a new separate inbound connection back to the server from your script, do not use the same one thta was tethered to the call because its too late to use that one. Why do I have to answer you twice? On Thu, Nov 26, 2009 at 3:27 AM, lakshmanan ganapathy wrote: > Hi, Any help or suggestion regarding my previous post. Especially > > > "I also noted that, if I don't receive any events, especially > "SERVER_DISCONNECTED", then the connection is in established state, but once > I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it > correct??" > Here is the program by which I confirmed the above! > > > require ESL; > use IO::Socket::INET; > > my $ip = "192.168.1.222"; > my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', > Proto => 'tcp', Listen => 2, Reuse => 1 ); > die "Could not create socket: $!\n" unless $sock; > my $con; > my $type = "user/"; > > for(;;) { > # wait for any client to connect, a new client will get connected > when a new call comes in the dialplan. > > my $new_sock = $sock->accept(); > # Do fork and let the parent to wait for more clients. > > my $pid = fork(); > if ($pid) { > close($new_sock); > next; > } > # Extract the host of the client. > > my $host = $new_sock->sockhost(); > # file descriptor for the socket. > > my $fd = fileno($new_sock); > print "Host name is $host\n"; > # Create object for the ESL connection package to access the ESL > functions. > > $con = new ESL::ESLconnection($fd); > # Gets the info about this channel. > > my $info = $con->getInfo(); > my $uuid = $info->getHeader("unique-id"); > printf "Connected call %s, from %s to %s\n", $uuid, > $info->getHeader("caller-caller-id-number"), > $info->getHeader("caller-destination-number"); > > # Answer the channel. > $con->execute("answer"); > # Set the event lock to tell the FS to execute the instructions in > the given order. > $con->setEventLock("true"); > # Play a file & Get the personal number from the user. > > $con->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); > $con->execute("hangup"); > > while($con->connected()) > { > my $e=$con->recvEvent(); > my $ename=$e->getHeader("Event-Name"); > print $e->serialize(); > print "$ename\n"; > print "Connection exists\n"; > sleep(1); > > } > print > "Bye\n------------------------------------------------------------------\n"; > close($new_sock); > } > I've not registered for any events. > In the above program I'm receiving the SERVER_DISCONNECTED event. > Output when receiving event: > Host name is 192.168.1.222 > Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 > Event-Name: SERVER_DISCONNECTED > > SERVER_DISCONNECTED > Connection exists > Bye > > When I comment the recvEvent line, I got the following output. > > Host name is 192.168.1.222 > Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 > Connection exists > Connection exists > Connection exists > Connection exists > Connection exists > > > > On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> I've tried the following program as per the suggestion that you've told. >> But it seems, no success. Once the connection is closed, I created a new >> connection and I send originate to originate a new call. But it is not >> working. >> >> require ESL; >> use IO::Socket::INET; >> use Data::Dumper; >> >> my $ip = "192.168.1.222"; >> my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', >> Proto => 'tcp', Listen => 2, Reuse => 1 ); >> die "Could not create socket: $!\n" unless $sock; >> >> my $make_call; >> my $con; >> my $type = "user/"; >> >> for(;;) { >> my $new_sock = $sock->accept(); >> my $pid = fork(); >> if ($pid) { >> close($new_sock); >> next; >> } >> my $host = $new_sock->sockhost(); >> my $fd = fileno($new_sock); >> $con = new ESL::ESLconnection($fd); >> my $info = $con->getInfo(); >> my $uuid = $info->getHeader("unique-id"); >> printf "Connected call %s, from %s to %s\n", $uuid, >> $info->getHeader("caller-caller-id-number"), >> $info->getHeader("caller-destination-number"); >> >> $con->filter("Unique-Id", $uuid); >> $con->events("plain", "all"); >> $con->execute("answer"); >> $con->setEventLock("true"); >> my $number=$con->execute("read","2 4 >> /usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav >> accnt_number 5000 #"); >> while($con->connected()) >> { >> my $e=$con->recvEvent(); >> my $ename=$e->getHeader("Event-Name"); >> my $app=$e->getHeader("Application"); >> if($ename eq "CHANNEL_EXECUTE_COMPLETE" and $app eq >> "read") >> { >> my $num=$e->getHeader("variable_accnt_number"); >> print "$num\n"; >> $con->execute("hangup"); >> } >> } >> if(!$con->connected()) >> { >> print "Connection not exists\n"; >> $con = new ESL::ESLconnection($fd); >> $con->api("originate","user/1000 &park()"); >> print "Hai\n"; >> } >> print >> "Bye\n------------------------------------------------------------------\n"; >> close($new_sock); >> } >> Output: >> Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097 >> 1000 >> Connection not exists >> Hai >> Bye >> ------------------------------------------------------------------ >> The freeswitch log is in >> http://pastebin.freeswitch.org/11258 >> >> I also noted that, if I don't receive any events, especially >> "SERVER_DISCONNECTED", then the connection is in established state, but once >> I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it >> correct?? >> >> >> >> >> >> On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> or open a new outbound connection at the end of your script so you can >>> send your originate command. >>> Since the channel hanging up will close your existing connection since >>> it's only an outbound single session socket. >>> >>> >>> On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins wrote: >>> >>>> >>>> >>>> On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < >>>> lakindia89 at gmail.com> wrote: >>>> >>>>> Hi, >>>>> I'm using perl ESL to control the call in freeswitch. >>>>> I'm having the following scenario, but not able to get it right. >>>>> >>>>> Dialplan: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 1. User A calls to an extention (1000). >>>>> 2. My ESL program will be running, and it answers the call. >>>>> 3. Then the program will get a number from the user. >>>>> 4. It will hangup the call. >>>>> 5. The program has to call to the number that was given by the user. >>>>> >>>>> In the above scenario, I was able to do until the 4th step. After >>>>> hangup the call, if I say originate it is not working. >>>>> Any ideas on how to do this in ESL. >>>>> >>>>> >>>> I want to make sure I understand what the script is supposed to be >>>> doing. The caller will key in a phone number to your script and your script >>>> will collect those digits. The script will then hangup on the caller and >>>> originate a completely new call? Perhaps you could use sched_api to schedule >>>> a new originate command for a few seconds into the future and then hangup? >>>> -MC >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/360a2aad/attachment-0002.html From eman at chabotel.com Fri Nov 27 11:28:41 2009 From: eman at chabotel.com (eman) Date: Fri, 27 Nov 2009 14:28:41 -0500 Subject: [Freeswitch-users] Connecting Multiple domains In-Reply-To: <4B0FB4BC.3090204@greatiam.com> References: <4B0FB4BC.3090204@greatiam.com> Message-ID: <164a9ab00911271128y76ac22a5q63233475f17c4a94@mail.gmail.com> check out http://wiki.freeswitch.org/wiki/Multi-tenant On Fri, Nov 27, 2009 at 6:15 AM, Otis wrote: > > Could someone please direct me to a link for connecting multiple say 2 > domains each with their own FS server. > > Thanks > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/e0e86311/attachment-0002.html From eman at chabotel.com Fri Nov 27 11:31:27 2009 From: eman at chabotel.com (eman) Date: Fri, 27 Nov 2009 14:31:27 -0500 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> Message-ID: <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> Check out the Linksys SPA2102 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > The only ATA mentioned on the WIKI that supports TLS/SRTP is the > Grandstream HandyTone 503. But, again according to the wiki, that > doesn't seem to behave to well with TLS ... > > On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: > > Mark Campbell-Smith wrote: > >> Does the SPA3102 support TLS or only SRTP? > > > > I don't know, but supporting only SRTP would be ridiculous, since the > keys > > would then be transmitted in the clear and therefore amenable to > interception. > > SRTP requires the SIP channel to be encrypted by TLS in order to be > secure. > > ZRTP, on the other hand, doesn't have this limitation: it works entirely > in > > RTP. > > > > I would be rather surprised were a hardware manufacturer to implement > SRTP > > without TLS for the SIP traffic. On the other hand, we've seen often in > this > > forum that some manufacturers are really clueless... > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/5e00b363/attachment-0002.html From anthony.minessale at gmail.com Fri Nov 27 11:32:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 13:32:42 -0600 Subject: [Freeswitch-users] Recording with Native File PCMU In-Reply-To: <23f91030911251735r3215a344h279a3f8589d5ff85@mail.gmail.com> References: <4256bf830911221048u279a52d2h2aea595052ce48e9@mail.gmail.com> <191c3a030911251319g60cdd5a3t33a82a560faf7a2b@mail.gmail.com> <23f91030911251735r3215a344h279a3f8589d5ff85@mail.gmail.com> Message-ID: <191c3a030911271132w1dd8efe1x5fbda139d197fa99@mail.gmail.com> If you want to go to that much trouble consider http://www.orecx.com/web/ They snoop the RTP from an entirely different box and record your calls for you. On Wed, Nov 25, 2009 at 7:35 PM, Seven Du wrote: > Yeah, that's why I had to record to two files(read&write) and need to mix > together by using sox. Do you only try to using PCMU to save CPU power > matt? As Anthony said, the difference can be ignored. And you also need to > take extra effort to make sure transcoding will not happen on a > conversation. > > But it maybe useful for expensive codecs like g729, iLBC, speex etc for > recording heavy scenarios. I'd like to take a look if there is a 5k bounty > ;) > > 2009/11/26 Anthony Minessale > > The processor power saved is negligible between PCMU and raw PCM and not >> worth the fuss. >> If you didn't decode the audio first you would not be able to mix the >> stream to produce a single file. >> So if we went to the trouble of making native media bugs to be able to do >> that you could barely use them so it would not be worth the 5k or more >> bounty to develop that functionality. >> >> >> >> On Sun, Nov 22, 2009 at 12:48 PM, Matthew Fong wrote: >> >>> I'm trying to conserve processor power by recording in native file >>> format, PCMU in my case. It works great with the following line >>> >>> session:execute("record", >>> "/tmp/my_recording."..session:getVariable("read_codec")); >>> >>> however it fails to work with >>> >>> session:execute("record_session", >>> "/tmp/my_recording."..session:getVariable("read_codec")); >>> or >>> record = api:execute("sched_api", '+1 none uuid_record >>> '..session:getVariable("uuid")..' start >>> /tmp/my_recording.'..session:getVariable("read_codec")); >>> >>> Why is it that it works with record, but not with record_session or >>> uuid_record? Is there something I'm over looking? In the latter two the >>> consul reports >>> >>> 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File >>> [/tmp/my_recording.PCMU] 8000hz >>> >>> as if it's recording, but /tmp/my_recording.PCMU never shows up. However >>> if I change it to .wav instead of .PCMU it works. Any ideas? >>> >>> --matt >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/3ba0be37/attachment-0002.html From anthony.minessale at gmail.com Fri Nov 27 13:11:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 15:11:48 -0600 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B0F1565.6060909@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> Message-ID: <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> Did you check the 2 replies that told you you need aliases in your sofia profile to translate the domain found in your message_waiting to the right profile? Both Brian and Mike answered you. On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX wrote: > I tried now with phones directly attached to the freeswitch (without an > OpenSIPS in between). I also added the alias. But the behaviour is as > before: > No notify message from freeswitch, neither after register nor after a > voicemail is recorded. > > Best regards > Peter > Brian West schrieb: > > Yes an alias will be required for every domain you run on the profile > > so it can find it. > > > > /b > > > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > > > > >> Try an alias on the sip profile. > >> > >> Mike > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/afe2bc87/attachment-0002.html From mike at jerris.com Fri Nov 27 13:13:20 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Nov 2009 16:13:20 -0500 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B0F1565.6060909@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> Message-ID: <95C676DC-A619-4D00-B039-F59E3D74C059@jerris.com> Does the alias you added match the one that you saw in the event? The alias is 100% for sure the fix for this issue, please check again. Mike On Nov 26, 2009, at 6:55 PM, Peter P GMX wrote: > I tried now with phones directly attached to the freeswitch (without an > OpenSIPS in between). I also added the alias. But the behaviour is as > before: > No notify message from freeswitch, neither after register nor after a > voicemail is recorded. > > Best regards > Peter > Brian West schrieb: >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >> >>> Try an alias on the sip profile. >>> >>> Mike >>> From anthony.minessale at gmail.com Fri Nov 27 13:22:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Nov 2009 15:22:25 -0600 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> Message-ID: <191c3a030911271322tce11dcy991dc1d668179a76@mail.gmail.com> based on your example past sip1.mydomain.com is the domain in the packet and thus the profile should have an alias for this. Then the user must reside in your sip db with the user 200 and domain sip1.mydomain.com if you dont have this consider the force-register-domain and force-register-db-domain to normalize the host names. On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Did you check the 2 replies that told you you need aliases in your sofia > profile to translate the domain found in your message_waiting to the right > profile? Both Brian and Mike answered you. > > > > > > On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX wrote: > >> I tried now with phones directly attached to the freeswitch (without an >> OpenSIPS in between). I also added the alias. But the behaviour is as >> before: >> No notify message from freeswitch, neither after register nor after a >> voicemail is recorded. >> >> Best regards >> Peter >> Brian West schrieb: >> > Yes an alias will be required for every domain you run on the profile >> > so it can find it. >> > >> > /b >> > >> > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> > >> > >> >> Try an alias on the sip profile. >> >> >> >> Mike >> >> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/612afe7b/attachment-0002.html From lists at redbonez.net Fri Nov 27 17:10:33 2009 From: lists at redbonez.net (Adam Ford) Date: Fri, 27 Nov 2009 18:10:33 -0700 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: <4B101F03.5090802@greatiam.com> References: <4B101F03.5090802@greatiam.com> Message-ID: <01f601ca6fc7$975e11a0$c61a34e0$@net> FusionPBX, FreePBX v3, and wikiPBX are the three that I have found in the past. However they all seem to be in the early stages of development, and not 100% stable. I can say this for sure about FreePBX and FusionPBX, but I have not actually tried wikiPBX. -AF -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Otis Sent: Friday, November 27, 2009 11:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch admin GUI Hi I am no sure but read up on fusionpbx. I asked the same question and someone pointed me to that. check web site Regards Samuel Mukoti wrote: >
Hi, > > Any recommendations for apps that can I use ontop of freeswitch as a > GUI manager, to manage extensions, queues, ivr, and dialplans? > > Thanks > > Sam > > > On 27 Nov,2009, at 5:19 PM, > freeswitch-users-request at lists.freeswitch.org wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Fri, 27 Nov 2009 16:19:03 +0100 >> From: Leon de Rooij >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset="windows-1252" >> >> Are you using the myodbc 3.51.18 version or higher ? >> >> I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to >> upgrade from jaunty.. >> >> regards, >> >> Leon >> >> >> On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: >> >>> Thanks. But when I made these entries in /etc/odbc.ini and rebooted? >>> >>> [freeswitch] >>> Driver = MySQL >>> SERVER = 127.0.0.1 >>> PORT = 4040 >>> DATABASE = mydb >>> OPTIONS = 67108864 >>> >>> ?I still get FS complaining with this. >>> >>> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 >>> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched >>> statements!#012If you are using mysql, make sure you are using >>> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS >>> >>> FreeSWITCH>version >>> FreeSWITCH Version 1.0.trunk (15660) >>> >>> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 >>> EST 2009 x86_64 x86_64 x86_64 GNU/Linux >>> >>> From /etc/odbcinst.ini >>> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so >>> Setup = /usr/lib64/libodbcmyS.so >>> >>> Is this a FS issue ? or an issue with mysql odbc? Any insight >>> would be great. >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Leon de Rooij >>> Sent: Friday, November 27, 2009 3:37 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >>> >>> There's a little info here on how to enable it with odbc: >>> >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 >>> >>> regards, >>> >>> Leon >>> >>> >>> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: >>> >>> >>> >>> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris >>> wrote: >>> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- >>> news-3-51-18.html >>> >>> MySQL Connector/ODBC now supports batched statements. In order to >>> enable >>> cached statement support you must switch enable the batched >>> statement option (FLAG_MULTI_STATEMENTS, >>> 67108864, or Allow multiple statements >>> within a GUI configuration). Be aware that batched statements >>> create an increased chance of SQL injection attacks and you >>> must >>> ensure that your application protects against this scenario. >>> (Bug#7445) >>> >>> >>> so, is this the right patch ? >>> >>> http://bugs.mysql.com/file.php?id=6994 >>> >>> >>> T. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/ 9c86b324/attachment.html >> >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 209 >> ************************************************* > > >
> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From tleyden at branchcut.com Fri Nov 27 18:50:42 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Fri, 27 Nov 2009 18:50:42 -0800 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: <01f601ca6fc7$975e11a0$c61a34e0$@net> References: <4B101F03.5090802@greatiam.com> <01f601ca6fc7$975e11a0$c61a34e0$@net> Message-ID: Well now might be a good time to give wikipbx a spin, because about 15 minutes ago we just released the second major release .. version 0.8 Here are the official release notes: --- WikiPBX has been converted from being based on Twisted.web2, a somewhat exotic webserver, to running as a mod_wsgi app within Apache2. Actually it can run under any webserver that support mod_wsgi. Additionally, the multi-tenancy has been changed from sip profile based multi-tenancy, which is not really "the normal way" to do this .. to the standard approach of domain based multi-tenancy. Along with that comes the ability to manage sip profiles from the GUI. A lot of security enhancements have been added, it is possible to force sip profile wide authorization, as well as per-extension dialplan security -- checking a flag in the dialplan will make it public (for anyone to access) or private (only registered users). The dialplan security is overlayed on top of the sip profile security .. and sip profile security takes precedence. (sip profile security is done by freeswitch whereas per-extension dialplan security is done by wikipbx) Gateways can now be per-tenant or shared among all tenants, and can be assigned to any sip profile on the system. Mod_voicemail now works with wikipbx, and is easy to configure. No GUI support yet in terms of visual voicemail. A really simple XML export / import has been added for existing users using version 0.5 to upgrade to version 0.8. Documentation has been completely re-written to reflect all changes in this release. What's still missing? The configuration XML that is served up to FreeSWITCH is really, really old. The only upshot is that they are served from static template files, so you can hack stuff in without having to look at any code. We are planning to fix this by the 1.0 release, slated for November of 2018. See http://wikipbx.subwiki.com/release-notes-0-8 for more details on this release and how to install it. On Fri, Nov 27, 2009 at 5:10 PM, Adam Ford wrote: > FusionPBX, FreePBX v3, and wikiPBX are the three that I have found in the > past. However they all seem to be in the early stages of development, and > not 100% stable. I can say this for sure about FreePBX and FusionPBX, but I > have not actually tried wikiPBX. > > -AF > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Otis > Sent: Friday, November 27, 2009 11:49 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Freeswitch admin GUI > > Hi > > I am no sure but read up on fusionpbx. I asked the same question and > someone pointed me to that. > check web site > > Regards > > > > Samuel Mukoti wrote: > >
Hi, > > > > Any recommendations for apps that can I use ontop of freeswitch as a > > GUI manager, to manage extensions, queues, ivr, and dialplans? > > > > Thanks > > > > Sam > > > > > > On 27 Nov,2009, at 5:19 PM, > > freeswitch-users-request at lists.freeswitch.org wrote: > > > >> Send FreeSWITCH-users mailing list submissions to > >> freeswitch-users at lists.freeswitch.org > >> > >> To subscribe or unsubscribe via the World Wide Web, visit > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> or, via email, send a message with subject or body 'help' to > >> freeswitch-users-request at lists.freeswitch.org > >> > >> You can reach the person managing the list at > >> freeswitch-users-owner at lists.freeswitch.org > >> > >> When replying, please edit your Subject line so it is more specific > >> than "Re: Contents of FreeSWITCH-users digest..." > >> > >> > >> Today's Topics: > >> > >> 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) > >> > >> > >> ---------------------------------------------------------------------- > >> > >> Message: 1 > >> Date: Fri, 27 Nov 2009 16:19:03 +0100 > >> From: Leon de Rooij > >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > >> To: freeswitch-users at lists.freeswitch.org > >> Message-ID: > >> Content-Type: text/plain; charset="windows-1252" > >> > >> Are you using the myodbc 3.51.18 version or higher ? > >> > >> I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to > >> upgrade from jaunty.. > >> > >> regards, > >> > >> Leon > >> > >> > >> On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: > >> > >>> Thanks. But when I made these entries in /etc/odbc.ini and rebooted? > >>> > >>> [freeswitch] > >>> Driver = MySQL > >>> SERVER = 127.0.0.1 > >>> PORT = 4040 > >>> DATABASE = mydb > >>> OPTIONS = 67108864 > >>> > >>> ?I still get FS complaining with this. > >>> > >>> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 > >>> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched > >>> statements!#012If you are using mysql, make sure you are using > >>> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS > >>> > >>> FreeSWITCH>version > >>> FreeSWITCH Version 1.0.trunk (15660) > >>> > >>> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 > >>> EST 2009 x86_64 x86_64 x86_64 GNU/Linux > >>> > >>> From /etc/odbcinst.ini > >>> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so > >>> Setup = /usr/lib64/libodbcmyS.so > >>> > >>> Is this a FS issue ? or an issue with mysql odbc? Any insight > >>> would be great. > >>> > >>> -----Original Message----- > >>> From: freeswitch-users-bounces at lists.freeswitch.org > >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org > >>> ] On Behalf Of Leon de Rooij > >>> Sent: Friday, November 27, 2009 3:37 AM > >>> To: freeswitch-users at lists.freeswitch.org > >>> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > >>> > >>> There's a little info here on how to enable it with odbc: > >>> > >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 > >>> > >>> regards, > >>> > >>> Leon > >>> > >>> > >>> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: > >>> > >>> > >>> > >>> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris > >>> wrote: > >>> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- > >>> news-3-51-18.html > >>> > >>> MySQL Connector/ODBC now supports batched statements. In order to > >>> enable > >>> cached statement support you must switch enable the batched > >>> statement option (FLAG_MULTI_STATEMENTS, > >>> 67108864, or Allow multiple statements > >>> within a GUI configuration). Be aware that batched statements > >>> create an increased chance of SQL injection attacks and you > >>> must > >>> ensure that your application protects against this scenario. > >>> (Bug#7445) > >>> > >>> > >>> so, is this the right patch ? > >>> > >>> http://bugs.mysql.com/file.php?id=6994 > >>> > >>> > >>> T. > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> > >>> http://www.freeswitch.org > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> > >>> http://www.freeswitch.org > >> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > >> > > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/ > 9c86b324/attachment.html > >> > >> > >> ------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> End of FreeSWITCH-users Digest, Vol 41, Issue 209 > >> ************************************************* > > > > > >
> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/d8d05d00/attachment-0002.html From juanbackson at gmail.com Fri Nov 27 21:48:21 2009 From: juanbackson at gmail.com (Juan Backson) Date: Sat, 28 Nov 2009 13:48:21 +0800 Subject: [Freeswitch-users] custom call counter Message-ID: <27c25bc40911272148o10bbcb9fo2566c5e9b64fa261@mail.gmail.com> Hi, Instead of using "show calls count" to obtain the current call count stat, I am writing some C code to increment a counter during on_answer_hook and decrement the counter during on_hangup_hook. It looks like my counter result is very closed to "show calls count" when the traffic is low, like 50 -60. But when traffic is high, like 1000 calls, my counter is showing 30% less. When all calls are finished, my counter becomes 0 again, and that proves that it does not increment/decrement more than it should. Is this normal? Does anyone have any idea why there is such discrepancy? thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/21dbd2b4/attachment-0002.html From mike at jerris.com Fri Nov 27 23:01:47 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 28 Nov 2009 02:01:47 -0500 Subject: [Freeswitch-users] custom call counter In-Reply-To: <27c25bc40911272148o10bbcb9fo2566c5e9b64fa261@mail.gmail.com> References: <27c25bc40911272148o10bbcb9fo2566c5e9b64fa261@mail.gmail.com> Message-ID: <028E32A6-1B83-4B74-8ABD-6D4F317B7325@jerris.com> It depends on the timing of when your increment and decrement are vs when the sql calls to push the events into the tables that are used for show calls are. Also, the sql calls are batched and queued causing a little delay (less than a second). If your doing a lot of short lived calls there is sure to be timing discrepancy. I am sure there is even more discrepancy if you look at the output of status which shows the current number of sessions (those are individual call legs) as that information is a little more real time. Mike On Nov 28, 2009, at 12:48 AM, Juan Backson wrote: > Hi, > > Instead of using "show calls count" to obtain the current call count stat, I am writing some C code to increment a counter during on_answer_hook and decrement the counter during on_hangup_hook. > > It looks like my counter result is very closed to "show calls count" when the traffic is low, like 50 -60. But when traffic is high, like 1000 calls, my counter is showing 30% less. > > When all calls are finished, my counter becomes 0 again, and that proves that it does not increment/decrement more than it should. > > Is this normal? Does anyone have any idea why there is such discrepancy? > > thanks, > jb > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From talk2ram at gmail.com Sat Nov 28 00:51:51 2009 From: talk2ram at gmail.com (ram) Date: Sat, 28 Nov 2009 14:21:51 +0530 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: References: <4B101F03.5090802@greatiam.com> <01f601ca6fc7$975e11a0$c61a34e0$@net> Message-ID: On Sat, Nov 28, 2009 at 8:20 AM, Traun Leyden wrote: > > Well now might be a good time to give wikipbx a spin, because about 15 > minutes ago we just released the second major release .. version 0.8 > > Here are the official release notes: > > --- > > WikiPBX has been converted from being based on Twisted.web2, a somewhat > exotic webserver, to running as a mod_wsgi app within Apache2. Actually it > can run under any webserver that support mod_wsgi. > > Additionally, the multi-tenancy has been changed from sip profile based > multi-tenancy, which is not really "the normal way" to do this .. to the > standard approach of domain based multi-tenancy. Along with that comes the > ability to manage sip profiles from the GUI. > > A lot of security enhancements have been added, it is possible to force sip > profile wide authorization, as well as per-extension dialplan security -- > checking a flag in the dialplan will make it public (for anyone to access) > or private (only registered users). The dialplan security is overlayed on > top of the sip profile security .. and sip profile security takes > precedence. (sip profile security is done by freeswitch whereas > per-extension dialplan security is done by wikipbx) > > Gateways can now be per-tenant or shared among all tenants, and can be > assigned to any sip profile on the system. > > Mod_voicemail now works with wikipbx, and is easy to configure. No GUI > support yet in terms of visual voicemail. > > A really simple XML export / import has been added for existing users using > version 0.5 to upgrade to version 0.8. > > Documentation has been completely re-written to reflect all changes in this > release. > > What's still missing? The configuration XML that is served up to > FreeSWITCH is really, really old. The only upshot is that they are served > from static template files, so you can hack stuff in without having to look > at any code. We are planning to fix this by the 1.0 release, slated for > November of 2018. > > See http://wikipbx.subwiki.com/release-notes-0-8 for more details on this > release and how to install it. > Hi does wikipbx have any IRC channel Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/16091786/attachment-0002.html From talk2ram at gmail.com Sat Nov 28 00:57:17 2009 From: talk2ram at gmail.com (ram) Date: Sat, 28 Nov 2009 14:27:17 +0530 Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: <4B0F8EC2.2080609@greatiam.com> References: <221275.23339.qm@web56403.mail.re3.yahoo.com> <4B0E7384.5010809@greatiam.com> <92e7d2090911260717j11ffad78kdd11b1c87dfd87be@mail.gmail.com> <4B0F8EC2.2080609@greatiam.com> Message-ID: On Fri, Nov 27, 2009 at 2:03 PM, Otis wrote: > Yes. I ventured to use that and got some error in connecting to the > mysql database. Will try with the default sqlite before getting > adventurous again. > > Hi download latest RC5 it has install wizard automatically create database ( sqllite/mysql/pgsql) Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/5e2c247e/attachment-0002.html From grevenx at me.com Sat Nov 28 02:58:05 2009 From: grevenx at me.com (=?iso-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Sat, 28 Nov 2009 11:58:05 +0100 Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: References: <4B101F03.5090802@greatiam.com> <01f601ca6fc7$975e11a0$c61a34e0$@net> Message-ID: On 28. nov. 2009, at 03.50, Traun Leyden wrote: > What's still missing? The configuration XML that is served up to FreeSWITCH is really, really old. The only upshot is that they are served from static template files, so you can hack stuff in without having to look at any code. We are planning to fix this by the 1.0 release, slated for November of 2018. Oh boy, I really DO hope FreeSWITCH will still be alive and well in 2018!! Best regards, Even Andr? From simon.woodhead at me.com Sat Nov 28 09:47:04 2009 From: simon.woodhead at me.com (Simon Woodhead) Date: Sat, 28 Nov 2009 17:47:04 +0000 Subject: [Freeswitch-users] Accessing custom SIP headers Message-ID: <86b72a770911280947m143f40aah640ff8e56ed08950@mail.gmail.com> Hi folks, I'm hoping someone can help me get at custom headers in the dial-plan. I've read about X- headers being accessible but need to get at some X_ headers passed through from a proxy. Reading the info app docs, the X shouldn't actually matter but no matter which way I try I always seem to get a null result. An example header in an INVITE is: X_ACCOUNTCODE: XXXXXX. I've tried the following dial-plan structures hoping one might work but none do: Any help would be much appreciated. Thanks, Simon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/6600985b/attachment-0002.html From christian.loeschenkohl at xpirio.com Sat Nov 28 11:24:31 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Sat, 28 Nov 2009 20:24:31 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing In-Reply-To: <191c3a030911271103s462910acu96da189b4064e63e@mail.gmail.com> References: <4B0ADFE1.4070506@xpirio.com> <5D7CFF6E-4667-4097-BCE4-A500C87AD55D@freeswitch.org> <4B0AF6EF.8070507@xpirio.com> <191c3a030911231307w346544fdh8c970134f465e5d6@mail.gmail.com> <4B0B005E.4080202@xpirio.com> <191c3a030911231648q1540444cj1e0e7e1da6aba0a5@mail.gmail.com> <4B0F8C3B.4000800@xpirio.com> <191c3a030911270857j697738f9n99e9e5bb1b71c38@mail.gmail.com> <191c3a030911271103s462910acu96da189b4064e63e@mail.gmail.com> Message-ID: <4B1178EF.3060604@xpirio.com> works now thank you very much On 2009-11-27 20:03, Anthony Minessale wrote: > please update to latest trunk 15698 or greater and re-test. > The 183 from the provider had a sendonly attr that tricked the proxy > code into thinking it was a hold/unhold operation. > > > On Fri, Nov 27, 2009 at 10:57 AM, Anthony Minessale > > wrote: > > or you can put it at a url on your web site and just post a link > > > 2009/11/27 Christian L?schenkohl > > > hello > > sorry, for my late reply > my core debugging was at info not at debug, now it's changed and > i have the log needed > > i'm sorry but pastebin doesn't work (it seems that my trace was > to big) > http://pastebin.freeswitch.org/11305 says "Query failure: Got a > packet bigger than 'max_allowed_packet' bytes" > > i'll send the logfile personal to you, hope you don't dislike this > > br > > On 2009-11-24 01:48, Anthony Minessale wrote: > > You forgot to set freeswitch to debug loglevel > > > > You need both of the following: > > > > console loglevel debug > > sofia profile internal siptrace on > > > > > > > > > > 2009/11/23 Christian L?schenkohl > > > >> > > > > sorry about wasting your time (wasn't my intent) > > > > the log is at http://pastebin.freeswitch.org/11240 > > i called 5214448370068 (also other calls are in the log) > > > > they now have changed 180 to 183 on the sonus, but makes no > > difference here > > > > br > > > > On 2009-11-23 22:07, Anthony Minessale wrote: > > > do you have the ringback variable set on the channel? > > > if so it will cause 180 to attempt to play inband ringback > indication > > > > > > I have nothing left to say because I asked for the whole > log with the > > > siptrace enables not just 5 lines of it. > > > If you still want help, give me the log to examine and I will > > tell you > > > what your problem is. > > > > > > > > > > > > 2009/11/23 Christian L?schenkohl > > > > > > > > > > >>> > > > > > > thany ou for your answer > > > > > > we use g729 on all our other connections in passthrough > mode > > and it > > > also doesn't work with alaw. > > > so i don't think it's related to this. > > > > > > br > > > > > > > > > On 2009-11-23 20:48, Brian West wrote: > > > > Well its also G729 so I suspect you don't have G729 > > > > > > > > /b > > > > > > > > On Nov 23, 2009, at 1:17 PM, Christian L?schenkohl wrote: > > > > > > > >> hi > > > >> > > > >> our freeswitch server has to talk to a sonus ip-switch > > > >> when we want to setup a call we do get a "100 Trying" > and then a > > > >> "180 Ringing" > > > >> within the "180 Ringing" we get a sdp with "a=sendonly" > then our > > > >> freeswitch > > > >> quits with a CANCEL message. > > > >> i simply don't get why our freeswitch aborts the session > - i think > > > >> it would work > > > >> if no "a=sendonly" would be present in the sdp. > > > >> > > > >> my technical contact doesn't want to switch 180 to 183 > on the > > sonus > > > >> side - this would > > > >> also work (i think). in fact he says that 180 ringing is > vaild, he > > > >> isn't that wrong in > > > >> this case. > > > >> > > > >> our freeswitch works in proxy mode, we do use trunk 15396 > > > >> see a ngrep trace under http://pastebin.freeswitch.org/11235 > > > >> > > > >> 92.63.208.36 - freeswitch > > > >> 38.105.229.100 - sonus > > > >> > > > >> br > > > >> > > > >> -- > > > >> Ing. Christian L?schenkohl > > > >> Technische Leitung, Forschung& Entwicklung VoIP > > > >> > > > >> xpirio > > > >> Telekommunikation& Service GmbH > > > >> Lakeside B04 > > > >> 9020 Klagenfurt > > > >> Austria > > > >> > > > >> T +43 (0) 5 77 11 - 1000 > > > >> F +43 (0) 5 77 11 - 1002 > > > >> E christian.loeschenkohl at xpirio.com > > > > > > > > > >> > > > >> > > > >> _______________________________________________ > > > >> FreeSWITCH-users mailing list > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > >> > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > > >> users > > > >> http://www.freeswitch.org > > > > > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > >> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > -- > > > Ing. Christian L?schenkohl > > > Technische Leitung, Forschung & Entwicklung VoIP > > > > > > xpirio > > > Telekommunikation & Service GmbH > > > Lakeside B04 > > > 9020 Klagenfurt > > > Austria > > > > > > T +43 (0) 5 77 11 - 1000 > > > F +43 (0) 5 77 11 - 1002 > > > E christian.loeschenkohl at xpirio.com > > > > > > > > > >> > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > > > > > > > >> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > > > > >> > > > IRC: irc.freenode.net > > > #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > > > > > > > >> > > > iax:guest at conference.freeswitch.org/888 > > > > > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > > > > >> > > > pstn:213-799-1400 > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung & Entwicklung VoIP > > > > xpirio > > Telekommunikation & Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From mctch at yahoo.com Sat Nov 28 13:42:35 2009 From: mctch at yahoo.com (Mark Crane) Date: Sat, 28 Nov 2009 13:42:35 -0800 (PST) Subject: [Freeswitch-users] GUI for Freeswitch -- wikiPBX In-Reply-To: Message-ID: <150632.44098.qm@web56408.mail.re3.yahoo.com> During the install of FusionPBX if you try to connect to MySQL connection and use 'localhost' it will attempt to use a Unix Socket then throws an error. Instead use 127.0.0.1 then it will actually use TCP connection rather than the UnixSocket connection. This is not a bug in FusionPBX it seems to be just how PHP PDO MySQL handles the connection. Hope this helps. For the release version I will add a little wording suggesting 127.0.0.1 vs localhost for those that have a local MySQL install. Best Regards, Mark J Crane ? --- On Sat, 11/28/09, ram wrote: From: ram Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX To: freeswitch-users at lists.freeswitch.org Date: Saturday, November 28, 2009, 1:57 AM On Fri, Nov 27, 2009 at 2:03 PM, Otis wrote: Yes. I ventured to use that ?and got some error in connecting to the mysql database. Will try with the default sqlite before getting adventurous again. ? Hi ? download latest RC5 ? it has install wizard automatically create database ( sqllite/mysql/pgsql) ? Ram ? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/9d84a520/attachment-0002.html From mctch at yahoo.com Sat Nov 28 13:51:41 2009 From: mctch at yahoo.com (Mark Crane) Date: Sat, 28 Nov 2009 13:51:41 -0800 (PST) Subject: [Freeswitch-users] Freeswitch admin GUI In-Reply-To: <01f601ca6fc7$975e11a0$c61a34e0$@net> Message-ID: <90909.63088.qm@web56405.mail.re3.yahoo.com> FusionPBX is very close to a release. FusionPBX is on the last release candidate 5 before a 1.0 release. Most of the work in the past couple weeks has been to make the install easier. ISO versions will be available in the future. I have multiple businesses already running live on FusionPBX. The project will advance faster the more it is used and the more feedback that is given. Mark J Crane http://www.fusionpbx.com --- On Fri, 11/27/09, Adam Ford wrote: From: Adam Ford Subject: Re: [Freeswitch-users] Freeswitch admin GUI To: freeswitch-users at lists.freeswitch.org Date: Friday, November 27, 2009, 6:10 PM FusionPBX, FreePBX v3, and wikiPBX are the three that I have found in the past. However they all seem to be in the early stages of development, and not 100% stable. I can say this for sure about FreePBX and FusionPBX, but I have not actually tried wikiPBX. -AF -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Otis Sent: Friday, November 27, 2009 11:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch admin GUI Hi I am no sure but read up on fusionpbx. I asked the same question and someone pointed me to that. check web site Regards Samuel Mukoti wrote: >
Hi, > > Any recommendations for apps that can I use ontop of freeswitch as a > GUI manager, to manage extensions, queues, ivr, and dialplans? > > Thanks > > Sam > > > On 27 Nov,2009, at 5:19 PM, > freeswitch-users-request at lists.freeswitch.org wrote: > >> Send FreeSWITCH-users mailing list submissions to >>? ? freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >>? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >>? ? freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >>? ? freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >>???1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Fri, 27 Nov 2009 16:19:03 +0100 >> From: Leon de Rooij >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset="windows-1252" >> >> Are you using the myodbc 3.51.18 version or higher ? >> >> I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to >> upgrade from jaunty.. >> >> regards, >> >> Leon >> >> >> On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: >> >>> Thanks.? But when I made these entries in /etc/odbc.ini and rebooted? >>> >>> [freeswitch] >>> Driver? ? ? ? ? = MySQL >>> SERVER? ? ? ? ? = 127.0.0.1 >>> PORT? ? ? ? ? ? = 4040 >>> DATABASE? ? ? ? = mydb >>> OPTIONS? ? ? ???= 67108864 >>> >>> ?I still get FS complaining with this. >>> >>> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 >>> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched >>> statements!#012If you are using mysql, make sure you are using >>> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS >>> >>> FreeSWITCH>version >>> FreeSWITCH Version 1.0.trunk (15660) >>> >>> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 >>> EST 2009 x86_64 x86_64 x86_64 GNU/Linux >>> >>> From /etc/odbcinst.ini >>> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so >>> Setup = /usr/lib64/libodbcmyS.so >>> >>> Is this a FS issue ?? or an issue with mysql odbc?? Any insight >>> would be great. >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Leon de Rooij >>> Sent: Friday, November 27, 2009 3:37 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS >>> >>> There's a little info here on how to enable it with odbc: >>> >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 >>> >>> regards, >>> >>> Leon >>> >>> >>> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: >>> >>> >>> >>> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris >>> wrote: >>> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- >>> news-3-51-18.html >>> >>> MySQL Connector/ODBC now supports batched statements. In order to >>> enable >>>? ? ? ? cached statement support you must switch enable the batched >>>? ? ? ? statement option (FLAG_MULTI_STATEMENTS, >>>? ? ? ? 67108864, or Allow multiple statements >>>? ? ? ? within a GUI configuration). Be aware that batched statements >>>? ? ? ? create an increased chance of SQL injection attacks and you >>> must >>>? ? ? ? ensure that your application protects against this scenario. >>>? ? ???(Bug#7445) >>> >>> >>> so, is this the right patch ? >>> >>> http://bugs.mysql.com/file.php?id=6994 >>> >>> >>> T. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/ 9c86b324/attachment.html >> >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 209 >> ************************************************* > > >
> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/0178e815/attachment-0002.html From john_platts at hotmail.com Sat Nov 28 21:34:24 2009 From: john_platts at hotmail.com (John Platts) Date: Sat, 28 Nov 2009 23:34:24 -0600 Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 Message-ID: I have updated my FreeSWITCH installation to revision 15700. I am experiencing call transfer problems whenever proxy media or bypass media is enabled. When proxy media and bypass media are both disabled, the call transfer does not fail and there are no audio issues. When proxy media mode is enabled, the call stays up after the transfer occurs, but there is no audio flowing on either end of the call. When bypass media mode is enabled, there is no audio flowing on either end of the call, and the call actually gets disconnected. I have collected detailed traces using the TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch command. I have attached a ZIP file named freeswitch-rev15700-traces-112809-2210.zip, which includes the following traces: - freeswitch-rev15700-trace-112809-2210-proxyandbypassoff.txt - A trace with both media proxying and media bypass disabled. The call is being transferred without any problems in this scenario. - freeswitch-rev15700-trace-112809-2210-proxyonandbypassoff.txt - A trace with media proxying enabled and media bypass disabled. Media proxying is enabled for the call legs in this scenario. The call stays up in this scenario, but there is no audio flowing after the transfer completed. In this scenario, FreeSWITCH does not shutdown cleanly, and there is a segmentation violation when FreeSWITCH is terminated. - freeswitch-rev15700-trace-112809-2210-proxyandbypasson.txt - A trace with both media proxying and media bypass enabled. Media bypass is enabled for the call legs in this scenario. The call actually gets dropped and there is no audio after the transfer is completed in this scenario. I have looked over the SIP traces of the failing scenarios. I have caught the following problems in the failing scenarios: - The o= line in SDP descriptors coming from the IP phone contains the private IP address, but the c= line in the SDP descriptors coming from the IP phone contains the public IP address. I have noticed a problem in re-INVITEs being sent from in proxy media and bypass media modes. The c= line in the re-invites contains the private IP address instead of the public IP address. The c= line was modified by a SIP ALG to contain a public IP address, but FreeSWITCH is actually not handling this correctly when calls are transferred. - The wrong codec is being negotiated in re-INVITE to the transferred number in the scenario when media proxying is enabled but media bypass is disabled. - In the scenario where media bypass is used, the re-INVITE actually appears to contain the correct details, and we are receiving the correct responses from our IP to IP gateway, but FreeSWITCH is not handling the media streams properly. Example of SDP descriptor coming from IP phone (with SDP descriptor modified by SIP ALG): v=0 o=- 123576 123576 IN IP4 192.168.1.4 s=- c=IN IP4 173.57.44.212 t=0 0 m=audio 16406 RTP/AVP 18 0 8 2 9 104 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:104 L16/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv Notice that the c= line has the correct public IP address and the m= line containing the correct port. Example of incorrect SDP descriptor being sent by FreeSWITCH in re-INVITES: v=0 o=- 121397 121398 IN IP4 192.168.1.4 s=- c=IN IP4 192.168.1.4 t=0 0 m=audio 16404 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=ptime:20 Note that the c= line contains the wrong IP address, but the m= line contains the correct RTP port. Example of wrong re-INVITE message being sent to the number that the call was being transferred to: INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj Max-Forwards: 69 From: "John Platts" ;tag=c61Drt38KF72m To: ;tag=2B1339E0-1A2C Call-ID: 1c095553-5741-122d-33a8-00185167f91d CSeq: 123615824 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 183 X-FS-Support: update_display Remote-Party-ID: "John Platts" ;party=calling;screen=yes;privacy=off v=0 o=- 123576 123577 IN IP4 192.168.1.4 s=- c=IN IP4 168.75.202.212 t=0 0 m=audio 30186 RTP/AVP 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 Here is the correct re-INVITE for the call that was unsuccessfully transferred (after the transfer was completed): INVITE sip:19729555871 at 168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKgaDHFKZrc06vD Max-Forwards: 16 From: ;tag=BX8mpZj5p6ggS To: ;tag=2B12D184-BEC Call-ID: 15A1F95-DBD611DE-8C95D9DF-3419A306 at 168.75.202.246 CSeq: 123615820 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 222 X-FS-Support: update_display v=0 o=- 121397 121399 IN IP4 192.168.1.4 s=- c=IN IP4 168.75.202.212 t=0 0 m=audio 26106 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 _________________________________________________________________ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch-rev15700-traces-112809-2210.zip Type: application/zip Size: 75260 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091128/dd970d52/attachment-0002.zip From john_platts at hotmail.com Sun Nov 29 04:51:57 2009 From: john_platts at hotmail.com (John Platts) Date: Sun, 29 Nov 2009 06:51:57 -0600 Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 In-Reply-To: References: Message-ID: To clarify the problem, the invite message is incorrect because comfort noise is being negotiated in the re-invite instead of G.711 or G.729: INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj Max-Forwards: 69 From: "John Platts" ;tag=c61Drt38KF72m To: ;tag=2B1339E0-1A2C Call-ID: 1c095553-5741-122d-33a8-00185167f91d CSeq: 123615824 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 183 X-FS-Support: update_display Remote-Party-ID: "John Platts" ;party=calling;screen=yes;privacy=off v=0 o=- 123576 123577 IN IP4 192.168.1.4 s=- c=IN IP4 168.75.202.212 t=0 0 m=audio 30186 RTP/AVP 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 How do I get it to negotiate G.711, G.729, or other codec instead of comfort noise? Our IP phones, our FXS gateways, and our IP to IP gateways expect G.711, G.729, iLBC (if supported by the endpoints), G.722 (if supported by the endpoints), or G.726 (if supported by the endpoints) be negotiated. ---------------------------------------- > From: john_platts at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Sat, 28 Nov 2009 23:34:24 -0600 > Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 > > > I have updated my FreeSWITCH installation to revision 15700. I am experiencing call transfer problems whenever proxy media or bypass media is enabled. When proxy media and bypass media are both disabled, the call transfer does not fail and there are no audio issues. When proxy media mode is enabled, the call stays up after the transfer occurs, but there is no audio flowing on either end of the call. When bypass media mode is enabled, there is no audio flowing on either end of the call, and the call actually gets disconnected. > > I have collected detailed traces using the TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch command. I have attached a ZIP file named freeswitch-rev15700-traces-112809-2210.zip, which includes the following traces: > - freeswitch-rev15700-trace-112809-2210-proxyandbypassoff.txt - A trace with both media proxying and media bypass disabled. The call is being transferred without any problems in this scenario. > - freeswitch-rev15700-trace-112809-2210-proxyonandbypassoff.txt - A trace with media proxying enabled and media bypass disabled. Media proxying is enabled for the call legs in this scenario. The call stays up in this scenario, but there is no audio flowing after the transfer completed. In this scenario, FreeSWITCH does not shutdown cleanly, and there is a segmentation violation when FreeSWITCH is terminated. > - freeswitch-rev15700-trace-112809-2210-proxyandbypasson.txt - A trace with both media proxying and media bypass enabled. Media bypass is enabled for the call legs in this scenario. The call actually gets dropped and there is no audio after the transfer is completed in this scenario. > > I have looked over the SIP traces of the failing scenarios. > > I have caught the following problems in the failing scenarios: > - The o= line in SDP descriptors coming from the IP phone contains the private IP address, but the c= line in the SDP descriptors coming from the IP phone contains the public IP address. I have noticed a problem in re-INVITEs being sent from in proxy media and bypass media modes. The c= line in the re-invites contains the private IP address instead of the public IP address. The c= line was modified by a SIP ALG to contain a public IP address, but FreeSWITCH is actually not handling this correctly when calls are transferred. > - The wrong codec is being negotiated in re-INVITE to the transferred number in the scenario when media proxying is enabled but media bypass is disabled. > - In the scenario where media bypass is used, the re-INVITE actually appears to contain the correct details, and we are receiving the correct responses from our IP to IP gateway, but FreeSWITCH is not handling the media streams properly. > > Example of SDP descriptor coming from IP phone (with SDP descriptor modified by SIP ALG): > v=0 > o=- 123576 123576 IN IP4 192.168.1.4 > s=- > c=IN IP4 173.57.44.212 > t=0 0 > m=audio 16406 RTP/AVP 18 0 8 2 9 104 101 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:104 L16/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > > Notice that the c= line has the correct public IP address and the m= line containing the correct port. > > Example of incorrect SDP descriptor being sent by FreeSWITCH in re-INVITES: > v=0 > o=- 121397 121398 IN IP4 192.168.1.4 > s=- > c=IN IP4 192.168.1.4 > t=0 0 > m=audio 16404 RTP/AVP 18 0 8 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendonly > a=ptime:20 > > Note that the c= line contains the wrong IP address, but the m= line contains the correct RTP port. > > Example of wrong re-INVITE message being sent to the number that the call was being transferred to: > INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj > Max-Forwards: 69 > From: "John Platts" ;tag=c61Drt38KF72m > To: ;tag=2B1339E0-1A2C > Call-ID: 1c095553-5741-122d-33a8-00185167f91d > CSeq: 123615824 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 183 > X-FS-Support: update_display > Remote-Party-ID: "John Platts" ;party=calling;screen=yes;privacy=off > > v=0 > o=- 123576 123577 IN IP4 192.168.1.4 > s=- > c=IN IP4 168.75.202.212 > t=0 0 > m=audio 30186 RTP/AVP 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > > Here is the correct re-INVITE for the call that was unsuccessfully transferred (after the transfer was completed): > INVITE sip:19729555871 at 168.75.202.246:5060 SIP/2.0 > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKgaDHFKZrc06vD > Max-Forwards: 16 > From: ;tag=BX8mpZj5p6ggS > To: ;tag=2B12D184-BEC > Call-ID: 15A1F95-DBD611DE-8C95D9DF-3419A306 at 168.75.202.246 > CSeq: 123615820 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 222 > X-FS-Support: update_display > > v=0 > o=- 121397 121399 IN IP4 192.168.1.4 > s=- > c=IN IP4 168.75.202.212 > t=0 0 > m=audio 26106 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > _________________________________________________________________ > Windows 7: I wanted simpler, now it's simpler. I'm a rock star. > http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/ From yehavi.bourvine at gmail.com Sun Nov 29 07:47:39 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 29 Nov 2009 17:47:39 +0200 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch Message-ID: Hello, I am trying to set a Polycom 501 phone to do conferencing via the conference room on Freeswitch rather than on the phone (as on the phone it is limited to 3 participants only). Anyone had success with it? I have on the Freeswitch an extension named Conf.* which activates the conference application (it works with other brands). On the Polycom I tried to define voIpProt.SIP.*conference*.address=sip:Conf0000 at freeswitch-server. The phone continues to create the conference locally and add the above Conf0000 to it, without REFERing the parties to it. The first phone which called is left on hold... Anyone managed to make this feature work? We use firmware 3.1.3. Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091129/7a593bce/attachment-0002.html From abeka at greatiam.com Sun Nov 29 09:38:57 2009 From: abeka at greatiam.com (Otis) Date: Sun, 29 Nov 2009 17:38:57 +0000 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 41, Issue 219 In-Reply-To: References: Message-ID: <4B12B1B1.8060807@greatiam.com> Hello Mark Thank you so much. I will put the advise to work. Regards freeswitch-users-request at lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > ------------------------------------------------------------------------ > > Today's Topics: > > 1. Re: GUI for Freeswitch -- wikiPBX (Mark Crane) > 2. Re: Freeswitch admin GUI (Mark Crane) > 3. Call transfer fails in proxy media and bypass media modes in > FreeSWITCH revision 15700 (John Platts) > > > ------------------------------------------------------------------------ > > Subject: > Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > From: > Mark Crane > Date: > Sat, 28 Nov 2009 13:42:35 -0800 (PST) > To: > freeswitch-users at lists.freeswitch.org > > To: > freeswitch-users at lists.freeswitch.org > > > During the install of FusionPBX if you try to connect to MySQL > connection and use 'localhost' it will attempt to use a Unix Socket > then throws an error. > > Instead use 127.0.0.1 then it will actually use TCP connection rather > than the UnixSocket connection. > > This is not a bug in FusionPBX it seems to be just how PHP PDO MySQL > handles the connection. > > Hope this helps. For the release version I will add a little wording > suggesting 127.0.0.1 vs localhost for those that have a local MySQL > install. > > Best Regards, > > Mark J Crane > > > > > > --- On *Sat, 11/28/09, ram //* wrote: > > > From: ram > Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX > To: freeswitch-users at lists.freeswitch.org > Date: Saturday, November 28, 2009, 1:57 AM > > > > On Fri, Nov 27, 2009 at 2:03 PM, Otis > wrote: > > Yes. I ventured to use that and got some error in connecting > to the > mysql database. Will try with the default sqlite before getting > adventurous again. > > > Hi > > download latest RC5 > > it has install wizard automatically create database ( > sqllite/mysql/pgsql) > > Ram > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > > Subject: > Re: [Freeswitch-users] Freeswitch admin GUI > From: > Mark Crane > Date: > Sat, 28 Nov 2009 13:51:41 -0800 (PST) > To: > freeswitch-users at lists.freeswitch.org > > To: > freeswitch-users at lists.freeswitch.org > > > FusionPBX is very close to a release. FusionPBX is on the last release > candidate 5 before a 1.0 release. Most of the work in the past couple > weeks has been to make the install easier. ISO versions will be > available in the future. > > I have multiple businesses already running live on FusionPBX. > > The project will advance faster the more it is used and the more > feedback that is given. > > Mark J Crane > http://www.fusionpbx.com > > > > --- On *Fri, 11/27/09, Adam Ford //* wrote: > > > From: Adam Ford > Subject: Re: [Freeswitch-users] Freeswitch admin GUI > To: freeswitch-users at lists.freeswitch.org > Date: Friday, November 27, 2009, 6:10 PM > > FusionPBX, FreePBX v3, and wikiPBX are the three that I have found > in the > past. However they all seem to be in the early stages of > development, and > not 100% stable. I can say this for sure about FreePBX and > FusionPBX, but I > have not actually tried wikiPBX. > > -AF > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On > Behalf Of Otis > Sent: Friday, November 27, 2009 11:49 AM > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Freeswitch admin GUI > > Hi > > I am no sure but read up on fusionpbx. I asked the same question and > someone pointed me to that. > check web site > > Regards > > > > Samuel Mukoti wrote: > >
Hi, > > > > Any recommendations for apps that can I use ontop of freeswitch > as a > > GUI manager, to manage extensions, queues, ivr, and dialplans? > > > > Thanks > > > > Sam > > > > > > On 27 Nov,2009, at 5:19 PM, > > freeswitch-users-request at lists.freeswitch.org > wrote: > > > >> Send FreeSWITCH-users mailing list submissions to > >> freeswitch-users at lists.freeswitch.org > > >> > >> To subscribe or unsubscribe via the World Wide Web, visit > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> or, via email, send a message with subject or body 'help' to > >> freeswitch-users-request at lists.freeswitch.org > > >> > >> You can reach the person managing the list at > >> freeswitch-users-owner at lists.freeswitch.org > > >> > >> When replying, please edit your Subject line so it is more specific > >> than "Re: Contents of FreeSWITCH-users digest..." > >> > >> > >> Today's Topics: > >> > >> 1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij) > >> > >> > >> > ---------------------------------------------------------------------- > >> > >> Message: 1 > >> Date: Fri, 27 Nov 2009 16:19:03 +0100 > >> From: Leon de Rooij > > >> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > >> To: freeswitch-users at lists.freeswitch.org > > >> Message-ID: > > > >> Content-Type: text/plain; charset="windows-1252" > >> > >> Are you using the myodbc 3.51.18 version or higher ? > >> > >> I'm using 3.51.19 (ubuntu karmic) and it works properly. I also > had to > >> upgrade from jaunty.. > >> > >> regards, > >> > >> Leon > >> > >> > >> On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: > >> > >>> Thanks. But when I made these entries in /etc/odbc.ini and > rebooted? > >>> > >>> [freeswitch] > >>> Driver = MySQL > >>> SERVER = 127.0.0.1 > >>> PORT = 4040 > >>> DATABASE = mydb > >>> OPTIONS = 67108864 > >>> > >>> ?I still get FS complaining with this. > >>> > >>> Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 > >>> [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched > >>> statements!#012If you are using mysql, make sure you are using > >>> MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS > >>> > >>> FreeSWITCH>version > >>> FreeSWITCH Version 1.0.trunk (15660) > >>> > >>> Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 > >>> EST 2009 x86_64 x86_64 x86_64 GNU/Linux > >>> > >>> From /etc/odbcinst.ini > >>> DRIVER = /usr/lib64/libmyodbc5-5.1.5.so > >>> Setup = /usr/lib64/libodbcmyS.so > >>> > >>> Is this a FS issue ? or an issue with mysql odbc? Any insight > >>> would be great. > >>> > >>> -----Original Message----- > >>> From: freeswitch-users-bounces at lists.freeswitch.org > > >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org > > >>> ] On Behalf Of Leon de Rooij > >>> Sent: Friday, November 27, 2009 3:37 AM > >>> To: freeswitch-users at lists.freeswitch.org > > >>> Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > >>> > >>> There's a little info here on how to enable it with odbc: > >>> > >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 > >>> > >>> regards, > >>> > >>> Leon > >>> > >>> > >>> On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: > >>> > >>> > >>> > >>> On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris > > > >>> wrote: > >>> http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- > >>> news-3-51-18.html > >>> > >>> MySQL Connector/ODBC now supports batched statements. In order to > >>> enable > >>> cached statement support you must switch enable the batched > >>> statement option (FLAG_MULTI_STATEMENTS, > >>> 67108864, or Allow multiple statements > >>> within a GUI configuration). Be aware that batched > statements > >>> create an increased chance of SQL injection attacks and you > >>> must > >>> ensure that your application protects against this > scenario. > >>> (Bug#7445) > >>> > >>> > >>> so, is this the right patch ? > >>> > >>> http://bugs.mysql.com/file.php?id=6994 > >>> > >>> > >>> T. > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> > >>> http://www.freeswitch.org > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> > >>> http://www.freeswitch.org > >> > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: > >> > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/ > 9c86b324/attachment.html > >> > >> > >> ------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> End of FreeSWITCH-users Digest, Vol 41, Issue 209 > >> ************************************************* > > > > > >
> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > > Subject: > [Freeswitch-users] Call transfer fails in proxy media and bypass media > modes in FreeSWITCH revision 15700 > From: > John Platts > Date: > Sat, 28 Nov 2009 23:34:24 -0600 > To: > > > To: > > > > I have updated my FreeSWITCH installation to revision 15700. I am experiencing call transfer problems whenever proxy media or bypass media is enabled. When proxy media and bypass media are both disabled, the call transfer does not fail and there are no audio issues. When proxy media mode is enabled, the call stays up after the transfer occurs, but there is no audio flowing on either end of the call. When bypass media mode is enabled, there is no audio flowing on either end of the call, and the call actually gets disconnected. > > I have collected detailed traces using the TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch command. I have attached a ZIP file named freeswitch-rev15700-traces-112809-2210.zip, which includes the following traces: > - freeswitch-rev15700-trace-112809-2210-proxyandbypassoff.txt - A trace with both media proxying and media bypass disabled. The call is being transferred without any problems in this scenario. > - freeswitch-rev15700-trace-112809-2210-proxyonandbypassoff.txt - A trace with media proxying enabled and media bypass disabled. Media proxying is enabled for the call legs in this scenario. The call stays up in this scenario, but there is no audio flowing after the transfer completed. In this scenario, FreeSWITCH does not shutdown cleanly, and there is a segmentation violation when FreeSWITCH is terminated. > - freeswitch-rev15700-trace-112809-2210-proxyandbypasson.txt - A trace with both media proxying and media bypass enabled. Media bypass is enabled for the call legs in this scenario. The call actually gets dropped and there is no audio after the transfer is completed in this scenario. > > I have looked over the SIP traces of the failing scenarios. > > I have caught the following problems in the failing scenarios: > - The o= line in SDP descriptors coming from the IP phone contains the private IP address, but the c= line in the SDP descriptors coming from the IP phone contains the public IP address. I have noticed a problem in re-INVITEs being sent from in proxy media and bypass media modes. The c= line in the re-invites contains the private IP address instead of the public IP address. The c= line was modified by a SIP ALG to contain a public IP address, but FreeSWITCH is actually not handling this correctly when calls are transferred. > - The wrong codec is being negotiated in re-INVITE to the transferred number in the scenario when media proxying is enabled but media bypass is disabled. > - In the scenario where media bypass is used, the re-INVITE actually appears to contain the correct details, and we are receiving the correct responses from our IP to IP gateway, but FreeSWITCH is not handling the media streams properly. > > Example of SDP descriptor coming from IP phone (with SDP descriptor modified by SIP ALG): > v=0 > o=- 123576 123576 IN IP4 192.168.1.4 > s=- > c=IN IP4 173.57.44.212 > t=0 0 > m=audio 16406 RTP/AVP 18 0 8 2 9 104 101 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:104 L16/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > > Notice that the c= line has the correct public IP address and the m= line containing the correct port. > > Example of incorrect SDP descriptor being sent by FreeSWITCH in re-INVITES: > v=0 > o=- 121397 121398 IN IP4 192.168.1.4 > s=- > c=IN IP4 192.168.1.4 > t=0 0 > m=audio 16404 RTP/AVP 18 0 8 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendonly > a=ptime:20 > > Note that the c= line contains the wrong IP address, but the m= line contains the correct RTP port. > > Example of wrong re-INVITE message being sent to the number that the call was being transferred to: > INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj > Max-Forwards: 69 > From: "John Platts" ;tag=c61Drt38KF72m > To: ;tag=2B1339E0-1A2C > Call-ID: 1c095553-5741-122d-33a8-00185167f91d > CSeq: 123615824 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 183 > X-FS-Support: update_display > Remote-Party-ID: "John Platts" ;party=calling;screen=yes;privacy=off > > v=0 > o=- 123576 123577 IN IP4 192.168.1.4 > s=- > c=IN IP4 168.75.202.212 > t=0 0 > m=audio 30186 RTP/AVP 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > > Here is the correct re-INVITE for the call that was unsuccessfully transferred (after the transfer was completed): > INVITE sip:19729555871 at 168.75.202.246:5060 SIP/2.0 > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKgaDHFKZrc06vD > Max-Forwards: 16 > From: ;tag=BX8mpZj5p6ggS > To: ;tag=2B12D184-BEC > Call-ID: 15A1F95-DBD611DE-8C95D9DF-3419A306 at 168.75.202.246 > CSeq: 123615820 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 222 > X-FS-Support: update_display > > v=0 > o=- 121397 121399 IN IP4 192.168.1.4 > s=- > c=IN IP4 168.75.202.212 > t=0 0 > m=audio 26106 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > _________________________________________________________________ > Windows 7: I wanted simpler, now it's simpler. I'm a rock star. > http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ujjval at simplesignal.com Sun Nov 29 11:17:37 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Sun, 29 Nov 2009 11:17:37 -0800 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch In-Reply-To: References: Message-ID: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> Polycom Firmware matrix (Look at the polycom website) does not allow firmware higher than 2.3.2 (I think) to be loaded on the old 501 phones...So first confirm you are on a supported firmware release... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi Bourvine Sent: Sunday, November 29, 2009 8:48 AM To: freeswitch-users Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch Hello, I am trying to set a Polycom 501 phone to do conferencing via the conference room on Freeswitch rather than on the phone (as on the phone it is limited to 3 participants only). Anyone had success with it? I have on the Freeswitch an extension named Conf.* which activates the conference application (it works with other brands). On the Polycom I tried to define voIpProt.SIP.conference.address=sip:Conf0000 at freeswitch-server. The phone continues to create the conference locally and add the above Conf0000 to it, without REFERing the parties to it. The first phone which called is left on hold... Anyone managed to make this feature work? We use firmware 3.1.3. Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091129/cf802f04/attachment-0002.html From errotan at gmail.com Sun Nov 29 10:06:51 2009 From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=) Date: Sun, 29 Nov 2009 19:06:51 +0100 Subject: [Freeswitch-users] CDR records Message-ID: <200911291906.51520.errotan@gmail.com> Hi Guys! I'm using the latest svn (15711) with the default xml config. Only modified cdr_csv.conf.xml the line to Here is what i do: 1. 1000 calls 1001 (1001 answers the call) 2. 1001 do blind transfer to 1002 (using *1) 3. 1001 hangs up 4. 1002 answers the call 5. 1002 and 1000 hangs up 3 cdr records are generated (simplified): from,to,start,duration "1000" "1001" "2009-11-29 15:21:53" "53" "50" "1000" "1002" "2009-11-29 15:21:53" "79" "76" "1000" "1002" "2009-11-29 15:22:46" "26" "23" As you can see the second cdr is incorrect because 1000 doesn't speak with 1002 for 76 second. Is this a normal ? Is it possible to make only 2 record ? Thank you for any answer. From jbarou at sqli.com Mon Nov 30 00:33:13 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Mon, 30 Nov 2009 09:33:13 +0100 Subject: [Freeswitch-users] Transfer Problem In-Reply-To: <191c3a030911270903i341d1f83pa15f67443422cb67@mail.gmail.com> References: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> <191c3a030911270903i341d1f83pa15f67443422cb67@mail.gmail.com> Message-ID: <8048ff7f0911300033u45c7aa5cwca16581ef9a22c2b@mail.gmail.com> My version is FreeSWITCH Version 1.0.trunk (15691M) http://jira.freeswitch.org/browse/FSBUILD-213 Thanks you. 2009/11/27 Anthony Minessale > by latest do you mean SVN trunk? > > Can you issue the command "sofia profile internal siptrace on" before > capturing your trace and post the results > to http://pastebin.freeswitch.org or open a jira > http://jira.freeswitch.org on the issue and attach the log after you > create the issue ticket, don't include it in the mailing list. > > > On Fri, Nov 27, 2009 at 10:47 AM, Jonathan Barou wrote: > >> Hi everybody, >> >> I'm actually using the lastest version of Freeswitch, I have a problem. I >> have a trunk SIP with my PABX. >> >> There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) >> 2. one Alcatel IpTouch 4028 with number 987 >> (on PABX) >> 3. one Siemens Gigaset A580 IP with number >> 8401 (on Freeswitch) >> >> >> *The first test* is to say to the phone 2 to transfer all the call to >> number 8401. So when I dial 987 on the phone 1, all work perfectly, the >> phone 3 is ringing and it's work. I have that in the log : >> >> 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending >> early media >> >> 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP >> [sofia/internal/368 at 10.33.69.246] 10.33.169.92 port 23054 -> 10.33.69.246 >> port 32000 codec: 8 ms: 90 >> >> 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer [soft] >> 720 bytes per 90ms >> >> 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: >> >> v=0 >> >> o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 >> >> s=FreeSWITCH >> >> c=IN IP4 10.33.169.92 >> >> t=0 0 >> >> m=audio 23054 RTP/AVP 8 106 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:106 telephone-event/8000 >> >> a=fmtp:106 0-16 >> >> a=silenceSupp:off - - - - >> >> a=ptime:90 >> >> a=sendrecv >> >> >> 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer >> sofia/internal/368 at 10.33.69.246! >> >> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >> >> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ >> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >> >> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ >> 368 at 10.33.69.246 skipping state [early][183] >> >> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal >> sofia/internal/368 at 10.33.69.246 [BREAK] >> >> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >> Activation Success L16 at 8000hz 1 channel 90ms >> >> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play >> Ringback Tone [%(2000,4000,440.0,480.0)] >> >> 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ >> 368 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >> >> 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port >> confirmed. >> >> 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read >> Buffer at 1440 bytes vs 81 >> >> >> >> *The Second Tes*t is to say to the phone 1 to transfer all the call to >> number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just >> one time and after it hangup. I have that in the log : >> >> >> 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending >> early media >> >> 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP >> [sofia/internal/987 at 10.33.69.246] 10.33.169.92 port 27732 -> 10.33.69.144 >> port 32000 codec: 8 ms: 90 >> >> 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer [soft] >> 720 bytes per 90ms >> >> 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: >> >> v=0 >> >> o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 >> >> s=FreeSWITCH >> >> c=IN IP4 10.33.169.92 >> >> t=0 0 >> >> m=audio 27732 RTP/AVP 8 106 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:106 telephone-event/8000 >> >> a=fmtp:106 0-16 >> >> a=silenceSupp:off - - - - >> >> a=ptime:90 >> >> a=sendrecv >> >> >> 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer >> sofia/internal/987 at 10.33.69.246! >> >> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >> >> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ >> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >> >> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ >> 987 at 10.33.69.246 skipping state [early][183] >> >> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal >> sofia/internal/987 at 10.33.69.246 [BREAK] >> >> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >> Activation Success L16 at 8000hz 1 channel 90ms >> >> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play >> Ringback Tone [%(2000,4000,440.0,480.0)] >> >> 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ >> 987 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >> >> 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ >> 987 at 10.33.69.246 entering state [terminated][487] >> >> 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ >> 987 at 10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] >> >> 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal >> sofia/internal/987 at 10.33.69.246 [KILL] >> >> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/987 at 10.33.69.246 [BREAK] >> >> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 thread >> mismatch skipping state handler. >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore >> original codec. >> >> 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup >> sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_CONSUME_MEDIA] >> [ORIGINATOR_CANCEL] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [KILL] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >> CS_HANGUP >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >> sip:8401 at 10.33.170.231:5060 Overriding SIP cause 487 with 487 from the >> other leg >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ >> sip:8401 at 10.33.170.231:5060 hanging up, cause: ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to >> sofia/internal/sip:8401 at 10.33.170.231:5060 >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard HANGUP, cause: >> ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP going to sleep >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_HANGUP -> >> CS_REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >> CS_REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard REPORTING, cause: >> ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING going to >> sleep >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_REPORTING -> >> CS_DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) Locked, Waiting on external >> entities >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 thread >> mismatch skipping state handler. >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate >> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] >> >> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 48 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) Ended >> >> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >> Channel sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_DESTROY] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >> CS_DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >> sip:8401 at 10.33.170.231:5060 SOFIA DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY going to sleep >> >> 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot create >> outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate >> Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >> >> 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. >> Cause: ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/987 at 10.33.69.246) State EXECUTE going to sleep >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/987 at 10.33.69.246) Running State Change CS_HANGUP >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >> (sofia/internal/987 at 10.33.69.246) State HANGUP >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >> 987 at 10.33.69.246 Overriding SIP cause 487 with 487 from the other leg >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel sofia/internal/ >> 987 at 10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >> sofia/internal/987 at 10.33.69.246 Standard HANGUP, cause: ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >> (sofia/internal/987 at 10.33.69.246) State HANGUP going to sleep >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/987 at 10.33.69.246) State Change CS_HANGUP -> CS_REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/987 at 10.33.69.246 [BREAK] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/987 at 10.33.69.246) Running State Change CS_REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >> (sofia/internal/987 at 10.33.69.246) State REPORTING >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >> sofia/internal/987 at 10.33.69.246 Standard REPORTING, cause: >> ORIGINATOR_CANCEL >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >> (sofia/internal/987 at 10.33.69.246) State REPORTING going to sleep >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >> (sofia/internal/987 at 10.33.69.246) State Change CS_REPORTING -> CS_DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >> sofia/internal/987 at 10.33.69.246 [BREAK] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 >> (sofia/internal/987 at 10.33.69.246) Locked, Waiting on external entities >> >> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 47 >> (sofia/internal/987 at 10.33.69.246) Ended >> >> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >> Channel sofia/internal/987 at 10.33.69.246 [CS_DESTROY] >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >> (sofia/internal/987 at 10.33.69.246) Running State Change CS_DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >> (sofia/internal/987 at 10.33.69.246) State DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >> 987 at 10.33.69.246 SOFIA DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >> sofia/internal/987 at 10.33.69.246 Standard DESTROY >> >> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >> (sofia/internal/987 at 10.33.69.246) State DESTROY going to sleep >> >> Finally when I tried to call the phone 3 with the phone 1 it's working, >> and not when I want to call the phone 3 with the phone 2, like just before, >> it's ringing just one time and hangup. >> >> >> Thanks you. >> >> >> Best Regards >> >> -- >> John >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jonathan BAROU Groupe SQLI - CRCI 0472405368 jbarou at sqli.com 1, place Verrazzano 69258 LYON CEDEX 09 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/79f5ecb6/attachment-0002.html From lakindia89 at gmail.com Mon Nov 30 02:03:59 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 30 Nov 2009 15:33:59 +0530 Subject: [Freeswitch-users] Callback to the user in ESL In-Reply-To: <191c3a030911271119k3f38a343k8351b121275580b9@mail.gmail.com> References: <7d79b3930911230325p6480f68fvac3adfbcad532e78@mail.gmail.com> <87f2f3b90911230951u33d20a58pcf9c49fe9e262326@mail.gmail.com> <191c3a030911231140w3b759cd6g17a80e9e3f026c89@mail.gmail.com> <7d79b3930911240427x2a1d5a40j35894fde28275642@mail.gmail.com> <7d79b3930911260127g27153b16ndf247e9f62c27dbb@mail.gmail.com> <191c3a030911271119k3f38a343k8351b121275580b9@mail.gmail.com> Message-ID: <7d79b3930911300203n5879c24fte50dbcada4aa2309@mail.gmail.com> In the previous reply you told me to use new "OUTBOUND" connection. But in this post you mention "INBOUND" connection. That confusion only made me to ask the question once again. Pardon me if I made any mistake. Making a new inbound connection does the task. Thanks for that. On Sat, Nov 28, 2009 at 12:49 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I told you to make a new separate inbound connection back to the server > from your script, do not use the same one thta was tethered to the call > because its too late to use that one. > > Why do I have to answer you twice? > > > > On Thu, Nov 26, 2009 at 3:27 AM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> Hi, Any help or suggestion regarding my previous post. Especially >> >> >> "I also noted that, if I don't receive any events, especially >> "SERVER_DISCONNECTED", then the connection is in established state, but once >> I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it >> correct??" >> Here is the program by which I confirmed the above! >> >> >> require ESL; >> use IO::Socket::INET; >> >> my $ip = "192.168.1.222"; >> my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', >> Proto => 'tcp', Listen => 2, Reuse => 1 ); >> die "Could not create socket: $!\n" unless $sock; >> my $con; >> my $type = "user/"; >> >> for(;;) { >> # wait for any client to connect, a new client will get connected >> when a new call comes in the dialplan. >> >> my $new_sock = $sock->accept(); >> # Do fork and let the parent to wait for more clients. >> >> my $pid = fork(); >> if ($pid) { >> close($new_sock); >> next; >> } >> # Extract the host of the client. >> >> my $host = $new_sock->sockhost(); >> # file descriptor for the socket. >> >> my $fd = fileno($new_sock); >> print "Host name is $host\n"; >> # Create object for the ESL connection package to access the ESL >> functions. >> >> $con = new ESL::ESLconnection($fd); >> # Gets the info about this channel. >> >> my $info = $con->getInfo(); >> my $uuid = $info->getHeader("unique-id"); >> printf "Connected call %s, from %s to %s\n", $uuid, >> $info->getHeader("caller-caller-id-number"), >> $info->getHeader("caller-destination-number"); >> >> # Answer the channel. >> $con->execute("answer"); >> # Set the event lock to tell the FS to execute the instructions in >> the given order. >> $con->setEventLock("true"); >> # Play a file & Get the personal number from the user. >> >> $con->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); >> $con->execute("hangup"); >> >> while($con->connected()) >> { >> my $e=$con->recvEvent(); >> my $ename=$e->getHeader("Event-Name"); >> print $e->serialize(); >> print "$ename\n"; >> print "Connection exists\n"; >> sleep(1); >> >> } >> print >> "Bye\n------------------------------------------------------------------\n"; >> close($new_sock); >> } >> I've not registered for any events. >> In the above program I'm receiving the SERVER_DISCONNECTED event. >> Output when receiving event: >> Host name is 192.168.1.222 >> Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 >> Event-Name: SERVER_DISCONNECTED >> >> SERVER_DISCONNECTED >> Connection exists >> Bye >> >> When I comment the recvEvent line, I got the following output. >> >> Host name is 192.168.1.222 >> Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097 >> Connection exists >> Connection exists >> Connection exists >> Connection exists >> Connection exists >> >> >> >> On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy < >> lakindia89 at gmail.com> wrote: >> >>> I've tried the following program as per the suggestion that you've told. >>> But it seems, no success. Once the connection is closed, I created a new >>> connection and I send originate to originate a new call. But it is not >>> working. >>> >>> require ESL; >>> use IO::Socket::INET; >>> use Data::Dumper; >>> >>> my $ip = "192.168.1.222"; >>> my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => >>> '8447', Proto => 'tcp', Listen => 2, Reuse => 1 ); >>> die "Could not create socket: $!\n" unless $sock; >>> >>> my $make_call; >>> my $con; >>> my $type = "user/"; >>> >>> for(;;) { >>> my $new_sock = $sock->accept(); >>> my $pid = fork(); >>> if ($pid) { >>> close($new_sock); >>> next; >>> } >>> my $host = $new_sock->sockhost(); >>> my $fd = fileno($new_sock); >>> $con = new ESL::ESLconnection($fd); >>> my $info = $con->getInfo(); >>> my $uuid = $info->getHeader("unique-id"); >>> printf "Connected call %s, from %s to %s\n", $uuid, >>> $info->getHeader("caller-caller-id-number"), >>> $info->getHeader("caller-destination-number"); >>> >>> $con->filter("Unique-Id", $uuid); >>> $con->events("plain", "all"); >>> $con->execute("answer"); >>> $con->setEventLock("true"); >>> my $number=$con->execute("read","2 4 >>> /usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav >>> accnt_number 5000 #"); >>> while($con->connected()) >>> { >>> my $e=$con->recvEvent(); >>> my $ename=$e->getHeader("Event-Name"); >>> my $app=$e->getHeader("Application"); >>> if($ename eq "CHANNEL_EXECUTE_COMPLETE" and $app eq >>> "read") >>> { >>> my $num=$e->getHeader("variable_accnt_number"); >>> print "$num\n"; >>> $con->execute("hangup"); >>> } >>> } >>> if(!$con->connected()) >>> { >>> print "Connection not exists\n"; >>> $con = new ESL::ESLconnection($fd); >>> $con->api("originate","user/1000 &park()"); >>> print "Hai\n"; >>> } >>> print >>> "Bye\n------------------------------------------------------------------\n"; >>> close($new_sock); >>> } >>> Output: >>> Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097 >>> 1000 >>> Connection not exists >>> Hai >>> Bye >>> ------------------------------------------------------------------ >>> The freeswitch log is in >>> http://pastebin.freeswitch.org/11258 >>> >>> I also noted that, if I don't receive any events, especially >>> "SERVER_DISCONNECTED", then the connection is in established state, but once >>> I receive the "SERVER_DISCONNECTED" event, the connection is closed. Is it >>> correct?? >>> >>> >>> >>> >>> >>> On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> or open a new outbound connection at the end of your script so you can >>>> send your originate command. >>>> Since the channel hanging up will close your existing connection since >>>> it's only an outbound single session socket. >>>> >>>> >>>> On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins wrote: >>>> >>>>> >>>>> >>>>> On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy < >>>>> lakindia89 at gmail.com> wrote: >>>>> >>>>>> Hi, >>>>>> I'm using perl ESL to control the call in freeswitch. >>>>>> I'm having the following scenario, but not able to get it right. >>>>>> >>>>>> Dialplan: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 1. User A calls to an extention (1000). >>>>>> 2. My ESL program will be running, and it answers the call. >>>>>> 3. Then the program will get a number from the user. >>>>>> 4. It will hangup the call. >>>>>> 5. The program has to call to the number that was given by the user. >>>>>> >>>>>> In the above scenario, I was able to do until the 4th step. After >>>>>> hangup the call, if I say originate it is not working. >>>>>> Any ideas on how to do this in ESL. >>>>>> >>>>>> >>>>> I want to make sure I understand what the script is supposed to be >>>>> doing. The caller will key in a phone number to your script and your script >>>>> will collect those digits. The script will then hangup on the caller and >>>>> originate a completely new call? Perhaps you could use sched_api to schedule >>>>> a new originate command for a few seconds into the future and then hangup? >>>>> -MC >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/5e3a1382/attachment-0002.html From neilp at cs.stanford.edu Mon Nov 30 01:49:43 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Mon, 30 Nov 2009 15:19:43 +0530 Subject: [Freeswitch-users] errors installing wanpipe drivers Message-ID: Hi All, I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: > cd wanpipe-3.5.6.5/ > make openzap ... make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make -C api/libstelephony clean make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make[1]: *** No rule to make target `clean'. Stop. make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make: *** [all_lib] Error 2 The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/debbdc85/attachment-0002.html From devel at thom.fr.eu.org Mon Nov 30 02:41:02 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Mon, 30 Nov 2009 11:41:02 +0100 Subject: [Freeswitch-users] errors installing wanpipe drivers In-Reply-To: References: Message-ID: I did manage to build these drivers, but maybe you're not doing it the right way. Sangoma document state that the drivers should be built by using their ./Setup script that does all that is required. I did use ./Setup install which builds the kernel modules, the wanrouter utilities and install all the required stuff. Then you can go back to freeswitch and build the mod_openzap/libopenzap. Fran?ois On Mon, 30 Nov 2009 15:19:43 +0530, Neil Patel wrote: Hi All, I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: > cd wanpipe-3.5.6.5/ > make openzap ... make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make -C api/libstelephony clean make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make[1]: *** No rule to make target `clean'. Stop. make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make: *** [all_lib] Error 2 The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/20f6c2b1/attachment-0002.html From mike at jerris.com Mon Nov 30 03:23:26 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 30 Nov 2009 06:23:26 -0500 Subject: [Freeswitch-users] errors installing wanpipe drivers In-Reply-To: References: Message-ID: make openzap is the correct way to build when using with openzap/freeswitch. If you are having issues with this you should check with sangoma support as to why that build of the drivers is not supporting it properly and what version you should be using. Mike On Nov 30, 2009, at 5:41 AM, Fran?ois Legal wrote: > I did manage to build these drivers, but maybe you're not doing it the right way. Sangoma document state that the drivers should be built by using their ./Setup script that does all that is required. > > I did use ./Setup install which builds the kernel modules, the wanrouter utilities and install all the required stuff. > > Then you can go back to freeswitch and build the mod_openzap/libopenzap. > > > Fran?ois > > > On Mon, 30 Nov 2009 15:19:43 +0530, Neil Patel wrote: > > Hi All, > > I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: > > > cd wanpipe-3.5.6.5/ > > make openzap > ... > make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' > make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' > make -C api/libstelephony clean > make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' > make[1]: *** No rule to make target `clean'. Stop. > make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' > make: *** [all_lib] Error 2 > > The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. > > Thanks, > Neil > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/f80061e3/attachment-0002.html From michaelt at voxcore.voxtelecom.co.za Mon Nov 30 04:12:18 2009 From: michaelt at voxcore.voxtelecom.co.za (Michael Toop) Date: Mon, 30 Nov 2009 14:12:18 +0200 Subject: [Freeswitch-users] DTMF Digits Lost when Under Load In-Reply-To: <191c3a030911160725k38ebcda8ta8c38c36eb80e627@mail.gmail.com> References: <330316f60911152307w2800f2e1r87c77d6dcd70be65@mail.gmail.com> <191c3a030911160725k38ebcda8ta8c38c36eb80e627@mail.gmail.com> Message-ID: <330316f60911300412k52e5bbd6h4236c696f9a45524@mail.gmail.com> Hi All, Thought I would share my solution to this DTMF problem: it turns out my ISP was capping my bandwidth & dropping packets to keep the connection & 1Mbps, so the experienced DTMF loss was actually packets being discarded. On my way to this discovery I tested Freeswitch & DTMF quite thoroughly & never actually found any problems even at hundreds of concurrent calls. Here is how I tested, who knows this might be useful to someone: - I used SIPp to generate calls & a Python script to log the received DTMF digits - SIPp command line: - sipp -sf dtmfSenario.xml -d 10000 -s 451 -l 96 -mp 5606 -i xxx.xxx.xxx.xxx - dtmfSenario.xml below - Dialplan: - - Python: - import sys from freeswitch import * def get_number(session,invalid,num=20): digits = session.getDigits(num, "", 15000) consoleLog("info","Got '%s' digits from user.\n" % digits) if digits == '': # Invalid call if invalid == 3: consoleLog("info","Three invalid attempts!!\n") session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav") session.hangup() sys.exit(0) else: session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav") get_number(session,invalid + 1) else: consoleLog("info","Got a valid number: %s, proceeding...\n" % digits) return digits def handler(session, args): session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav") numberToDial = get_number(session,2,num=10) consoleLog('info','Got 10 DTMF digits. Writing "1" to file...\n') fo = open('/tmp/dtmfData.csv','a') fo.write('"1"\n') fo.close() # Do some stuff & wait for SIPP to hangup session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav") session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav") return - DTMF senario file: - # cat dtmfSenario.xml ;tag=[call_number] To: sut Call-ID: [call_id] CSeq: 1 INVITE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[local_ip_type] [local_ip] t=0 0 m=audio [auto_media_port] RTP/AVP 18 100 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ]]> ;tag=[call_number] To: sut [peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> ;tag=[call_number] To: sut [peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> Cheers, Michael On Mon, Nov 16, 2009 at 5:25 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > That's a pretty small problem description to be so sure about something. > It would probably be better to capture some evidence of the exact problem > you are having since we are using computers and we need to see the computers > in action doing something specifically incorrect to diagnose any sort of > problem. Take the time to describe the origin and destination of your > calls, the call flow, the hardware in use on both ends of the call, detailed > console logs on debug level, (maybe even uncomment the 2833 debug ifded in > switch_rtp.c) and gather something to go on besides "I seem to be losing > dtmf) maybe a packect capture of the networking interface on both ends of > these calls. > > Also problems should be reported to http://jira.freeswitch.org not this > mailing list. > Save us a step if you report a jira and provide all the info above or we > will just have to ask for it again. > > > On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop < > michaelt at voxcore.voxtelecom.co.za> wrote: > >> Hi All, >> >> I have an issue that when my call volumes on my FS IVR box > 30 calls >> DTMF digits are lost (using RFC2833). It is definitely load related as it >> all works perfectly under 30 calls. >> >> Any pointers or a solution to the problem? >> >> Thanks, >> >> Michael >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/227f9d35/attachment-0002.html From woodydickson at gmail.com Mon Nov 30 06:49:32 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Mon, 30 Nov 2009 22:49:32 +0800 Subject: [Freeswitch-users] park on hook Message-ID: Hi, Is there anyway to detect when a channel is park in a way that is similar to hangup-hook or answer-hook? I would like to detect that inside a custom mod, without using the event mechanism? woody -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/9c6fba42/attachment-0002.html From devel at thom.fr.eu.org Mon Nov 30 07:36:48 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Mon, 30 Nov 2009 16:36:48 +0100 Subject: [Freeswitch-users] =?utf-8?q?CLIP_on_FXS_channels_with_mod=5Fopen?= =?utf-8?q?zap?= Message-ID: <567eba90a27903f327c037bcb6062b1a@thom.fr.eu.org> Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. Fran?ois -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/d258e10c/attachment-0002.html From moises.silva at gmail.com Mon Nov 30 07:39:17 2009 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 30 Nov 2009 10:39:17 -0500 Subject: [Freeswitch-users] errors installing wanpipe drivers In-Reply-To: References: Message-ID: On Mon, Nov 30, 2009 at 4:49 AM, Neil Patel wrote: > Hi All, > > I am currently installing a Sangoma A102 card to work with FS using wanpipe > drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related > modules to compile: > > > cd wanpipe-3.5.6.5/ > > make openzap > ... > make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' > make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' > make -C api/libstelephony clean > make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' > make[1]: *** No rule to make target `clean'. Stop. > make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' > make: *** [all_lib] Error 2 > > The libstelephony directory has no Makefile in it. Why is it missing? Is > there a version of wanpipe drivers that will work? I have been unsuccessful > with 3.4.4 and 3.5.6 in similar fashion. > > Hi Neil, Most likely the creation of the Makefile failed (since you mention you can't see a Makefile). Please be sure to have installed the pre-requisites listed at http://wiki.sangoma.com/Requirements Particularly in this case, libtool, autoconf and automake packages. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/6c28008f/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 30 08:48:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Nov 2009 10:48:26 -0600 Subject: [Freeswitch-users] CLIP on FXS channels with mod_openzap In-Reply-To: <567eba90a27903f327c037bcb6062b1a@thom.fr.eu.org> References: <567eba90a27903f327c037bcb6062b1a@thom.fr.eu.org> Message-ID: <191c3a030911300848m6990a978jff57a4b74dd2192d@mail.gmail.com> can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: > Hello, > > > > I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems > on the FXS ports. > > When I ring on FXS ports, the connected phone does not display > callerid/callerid-name. > > I tried turning the stuff of in openzap.conf.xml () but it did not help. > > > > As a side note, turning this on on the FXO ports drops the callerid > information on incoming calls. > > > > Running freeswitch 1.0.4 on linux 2.6.27. > > > > Fran?ois > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/fb01126e/attachment-0002.html From helmut.kuper at ewetel.de Mon Nov 30 08:52:17 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 30 Nov 2009 17:52:17 +0100 Subject: [Freeswitch-users] Sangoma RTP TAP Message-ID: <4B13F841.1080905@ewetel.de> Hello, has anyone of you tried the RTP TAP function of sangoma`s wanpipe driver? It is described here: http://wiki.sangoma.com/wanpipe-voice-rtp-tap On my side wanrouter log says that RTP TAB is configured and enabled, but I can't detect any udp packets received by the remote server (which is described by RTP_TAP_IP, RTP_TAP_MAC and RTP_TAP_PORT). I've latest driver and double checked the wanpipe.conf config. I tried to send some udp packets from wanpiping server to remote server, where the packets were shown up via tcpdump. So there is no FW problem involved. Each try to do some kind of printf debugging in wanpipe-driver doesn't succeed. Any ideas? From kristian.kielhofner at gmail.com Mon Nov 30 08:54:42 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 30 Nov 2009 11:54:42 -0500 Subject: [Freeswitch-users] Accessing custom SIP headers In-Reply-To: <86b72a770911280947m143f40aah640ff8e56ed08950@mail.gmail.com> References: <86b72a770911280947m143f40aah640ff8e56ed08950@mail.gmail.com> Message-ID: <2d9149cd0911300854j251f1481s6ad9405f0b2effb5@mail.gmail.com> The correct way to pass non-standard headers is X- not X_ . On Sat, Nov 28, 2009 at 12:47 PM, Simon Woodhead wrote: > Hi folks, > I'm hoping someone can help me get at custom headers in the dial-plan. I've > read about X- headers being accessible but need to get at some X_ headers > passed through from a proxy. Reading the info app docs, the X shouldn't > actually matter but no matter which way I try I always seem to get a null > result. > An example header in an INVITE is: > X_ACCOUNTCODE: XXXXXX. > I've tried the following dial-plan structures hoping one might work but none > do: > data="accountcodea=${variable_sip_h_X_ACCOUNTCODE}" /> > /> > > > Any help would be much appreciated. > Thanks, > Simon > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From afritzlists at fritztech.com Mon Nov 30 08:54:52 2009 From: afritzlists at fritztech.com (Andrew Fritz) Date: Mon, 30 Nov 2009 10:54:52 -0600 Subject: [Freeswitch-users] Polycom Phones and Domains Message-ID: <4B13F8DC.9050108@fritztech.com> I'm attempting to configure several varieties of polycom (SoundPoint IP 550, SoundPoint IP 601) phones to connect to a freeswitch instance using a domain other than default (i.e. the ip address). Everything works wonderfully as long as the domain is named exactly the same thing as the server host provided to the phone (whether that is the server's ip address, or a hostname resolved via dns). As long as those match everything is fine. What I'm trying to sort out is, is it possible to convince the phone to use something other than the server's hostname/ip as the second part of the user name (i.e. user at host)? Or should I just resign myself to making the domain name some host name that can resolve via DNS? I've tried including @somedomain in the authid field of the line of the phone, but freeswitch reports that the user someuser at somedomain@serverip couldn't be authenticated. It appears that the phone always appends the server name... Does anyone know of a way that I configure a polycom to connect as user1 at d1 and another to connect as user2 at d2 where d1 and d2 are NOT in DNS or are polycoms just going to always use the server host as the @ part of the username? Andrew From anthony.minessale at gmail.com Mon Nov 30 09:22:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Nov 2009 11:22:36 -0600 Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 In-Reply-To: References: Message-ID: <191c3a030911300922y794a23bawf66e203ffc76af89@mail.gmail.com> I don't quite understand what you are talking about? So you have bypass_media=true and you attempt to make an attended xfer as soon as you complete the transfer according to your trace FS does re-invites to convert the call to be exchanging media with FS. The o= lines you don't like are being set by the anonymous device in your callflow and should not impact anything at all. Are you saying something that used to work suddenly has caused you problems or is this the first time you are trying this because we have tested this scenario many times. Are you getting packet captures also and checking where the media is going after those re-invites? if you are intentionally using an ALG you might try without it because 100% of ALG we have ever seen have been badly broken when working with something like FS. On Sun, Nov 29, 2009 at 6:51 AM, John Platts wrote: > > To clarify the problem, the invite message is incorrect because comfort > noise is being negotiated in the re-invite instead of G.711 or G.729: > INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj > Max-Forwards: 69 > From: "John Platts" > >;tag=c61Drt38KF72m > To: > >;tag=2B1339E0-1A2C > Call-ID: 1c095553-5741-122d-33a8-00185167f91d > CSeq: 123615824 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 183 > X-FS-Support: update_display > Remote-Party-ID: "John Platts" > >;party=calling;screen=yes;privacy=off > > v=0 > o=- 123576 123577 IN IP4 192.168.1.4 > s=- > c=IN IP4 168.75.202.212 > t=0 0 > m=audio 30186 RTP/AVP 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > > How do I get it to negotiate G.711, G.729, or other codec instead of > comfort noise? Our IP phones, our FXS gateways, and our IP to IP gateways > expect G.711, G.729, iLBC (if supported by the endpoints), G.722 (if > supported by the endpoints), or G.726 (if supported by the endpoints) be > negotiated. > > ---------------------------------------- > > From: john_platts at hotmail.com > > To: freeswitch-users at lists.freeswitch.org > > Date: Sat, 28 Nov 2009 23:34:24 -0600 > > Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass > media modes in FreeSWITCH revision 15700 > > > > > > I have updated my FreeSWITCH installation to revision 15700. I am > experiencing call transfer problems whenever proxy media or bypass media is > enabled. When proxy media and bypass media are both disabled, the call > transfer does not fail and there are no audio issues. When proxy media mode > is enabled, the call stays up after the transfer occurs, but there is no > audio flowing on either end of the call. When bypass media mode is enabled, > there is no audio flowing on either end of the call, and the call actually > gets disconnected. > > > > I have collected detailed traces using the TPORT_LOG=1 > /usr/local/freeswitch/bin/freeswitch command. I have attached a ZIP file > named freeswitch-rev15700-traces-112809-2210.zip, which includes the > following traces: > > - freeswitch-rev15700-trace-112809-2210-proxyandbypassoff.txt - A trace > with both media proxying and media bypass disabled. The call is being > transferred without any problems in this scenario. > > - freeswitch-rev15700-trace-112809-2210-proxyonandbypassoff.txt - A trace > with media proxying enabled and media bypass disabled. Media proxying is > enabled for the call legs in this scenario. The call stays up in this > scenario, but there is no audio flowing after the transfer completed. In > this scenario, FreeSWITCH does not shutdown cleanly, and there is a > segmentation violation when FreeSWITCH is terminated. > > - freeswitch-rev15700-trace-112809-2210-proxyandbypasson.txt - A trace > with both media proxying and media bypass enabled. Media bypass is enabled > for the call legs in this scenario. The call actually gets dropped and there > is no audio after the transfer is completed in this scenario. > > > > I have looked over the SIP traces of the failing scenarios. > > > > I have caught the following problems in the failing scenarios: > > - The o= line in SDP descriptors coming from the IP phone contains the > private IP address, but the c= line in the SDP descriptors coming from the > IP phone contains the public IP address. I have noticed a problem in > re-INVITEs being sent from in proxy media and bypass media modes. The c= > line in the re-invites contains the private IP address instead of the public > IP address. The c= line was modified by a SIP ALG to contain a public IP > address, but FreeSWITCH is actually not handling this correctly when calls > are transferred. > > - The wrong codec is being negotiated in re-INVITE to the transferred > number in the scenario when media proxying is enabled but media bypass is > disabled. > > - In the scenario where media bypass is used, the re-INVITE actually > appears to contain the correct details, and we are receiving the correct > responses from our IP to IP gateway, but FreeSWITCH is not handling the > media streams properly. > > > > Example of SDP descriptor coming from IP phone (with SDP descriptor > modified by SIP ALG): > > v=0 > > o=- 123576 123576 IN IP4 192.168.1.4 > > s=- > > c=IN IP4 173.57.44.212 > > t=0 0 > > m=audio 16406 RTP/AVP 18 0 8 2 9 104 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:2 G726-32/8000 > > a=rtpmap:9 G722/8000 > > a=rtpmap:104 L16/16000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > a=sendrecv > > > > Notice that the c= line has the correct public IP address and the m= line > containing the correct port. > > > > Example of incorrect SDP descriptor being sent by FreeSWITCH in > re-INVITES: > > v=0 > > o=- 121397 121398 IN IP4 192.168.1.4 > > s=- > > c=IN IP4 192.168.1.4 > > t=0 0 > > m=audio 16404 RTP/AVP 18 0 8 101 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=sendonly > > a=ptime:20 > > > > Note that the c= line contains the wrong IP address, but the m= line > contains the correct RTP port. > > > > Example of wrong re-INVITE message being sent to the number that the call > was being transferred to: > > INVITE sip:19729831777 at 168.75.202.246:5060 SIP/2.0 > > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj > > Max-Forwards: 69 > > From: "John Platts" ;tag=c61Drt38KF72m > > To: ;tag=2B1339E0-1A2C > > Call-ID: 1c095553-5741-122d-33a8-00185167f91d > > CSeq: 123615824 INVITE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 183 > > X-FS-Support: update_display > > Remote-Party-ID: "John Platts" ;party=calling;screen=yes;privacy=off > > > > v=0 > > o=- 123576 123577 IN IP4 192.168.1.4 > > s=- > > c=IN IP4 168.75.202.212 > > t=0 0 > > m=audio 30186 RTP/AVP 101 13 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:13 CN/8000 > > > > Here is the correct re-INVITE for the call that was unsuccessfully > transferred (after the transfer was completed): > > INVITE sip:19729555871 at 168.75.202.246:5060 SIP/2.0 > > Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKgaDHFKZrc06vD > > Max-Forwards: 16 > > From: ;tag=BX8mpZj5p6ggS > > To: ;tag=2B12D184-BEC > > Call-ID: 15A1F95-DBD611DE-8C95D9DF-3419A306 at 168.75.202.246 > > CSeq: 123615820 INVITE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 222 > > X-FS-Support: update_display > > > > v=0 > > o=- 121397 121399 IN IP4 192.168.1.4 > > s=- > > c=IN IP4 168.75.202.212 > > t=0 0 > > m=audio 26106 RTP/AVP 0 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > > > _________________________________________________________________ > > Windows 7: I wanted simpler, now it's simpler. I'm a rock star. > > > http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009 > > _________________________________________________________________ > Hotmail: Trusted email with powerful SPAM protection. > http://clk.atdmt.com/GBL/go/177141665/direct/01/ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/127d1dd5/attachment-0002.html From moises.silva at gmail.com Mon Nov 30 10:05:21 2009 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 30 Nov 2009 13:05:21 -0500 Subject: [Freeswitch-users] Sangoma RTP TAP In-Reply-To: <4B13F841.1080905@ewetel.de> References: <4B13F841.1080905@ewetel.de> Message-ID: Hello Helmut, On Mon, Nov 30, 2009 at 11:52 AM, Helmut Kuper wrote: > Each try to do some kind of printf debugging in wanpipe-driver doesn't > succeed. > > Any ideas? > The way the rtp tapping works right now is kinda hackish and pretty much Asterisk/Zaptel-based. We depend on the application (either Asterisk or FreeSWITCH) to enable/disable echo cancellation via zaptel commands. When echo cancellation is enabled we assume a call started and enable the tapping, when echo cancellation is disabled we stop the tapping. This behavior has yet to be implemented for FreeSWITCH. An easy way to do it is just to have the wanpipe card to work in zaptel mode and then add a call to zap_channel_command(tech_pvt->zchan, ZAP_COMMAND_ENABLE_ECHOCANCEL) on call start and ZAP_COMMAND_DISABLE_ECHOCANCEL on call stop in mod_openzap.c. The right way to do it is via new API in libsangoma to start tapping and stop tapping. I will add the new libsangoma API to my todo list, hopefully will be done sometime this month. If you want to test the first quick approach send me an off-line message with ssh connection information to get into your box to do these changes so you can test them. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/fa0ef0de/attachment-0002.html From info at daccii.it Mon Nov 30 11:46:54 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Mon, 30 Nov 2009 20:46:54 +0100 Subject: [Freeswitch-users] Questions on ISDN support for Freeswitch Message-ID: <4B14212E.7060003@daccii.it> Hi to all, shortly i'll make a pbx for a customer that uses a couple of isdn bri lines and, looking for hardware, i've seen that not too much expensive isdn cards that works well are the ones that uses hfc-4/8s controller (specifically i'll use a OpenVox B200P that has 2 ISDN ports and use an HFC-4S controller). I've seen that FreeSwitch doesn't support mISDN but uses openzap (trough ozmod_isdn.so). I got some serious troubles using OpenZAP on analogical lines (bad dtmf recognition on fxs ports (like press 4 and get 44), annoying noise, busy/hangup tone unrecognized [tones, in italy, differs by cadency and not frequency], and more). Using tone_detect i bypassed the tone recognition problem and the noise was however acceptable: the blocking problem was the bad dtmf recognition. In the end (hope god forgive me) i put asterisk as (*1*)(*2*)sip proxy between zaptel and freeswich: i need to fix in config hangup cause detection but it seems to works fine. So my questions are: - Do freeswitch supports mISDN? - If it doesn't support mISDN, tone and dtmf recognition will be done by the isdn card, the kernel module or will be done by openzap? - There are alternative ways to use (*3*) mISDN with freeswitch, apart put asterisk as proxy? Thank for your support! Best Regards, Daniele --- (*1*) At beginning i tried using IAX but freeswitch segfaults when it try to answer the call and when i reload the module (trought reload mod_iax) shutdown routing didn't get called(the used iax library, smartly, start using another port without saying anything), however i need to do more testing: hope to open some tickets on jira in short. (*2*) I've seen that mod_iax config file support a "context" variable, but it isn't used so i wrote a small fix to use it if context isn't specified in the iax request (*3*) I need to use mISDN, rather other things, because i should use octasis soft echo cancellation and them supports only mISDN and Zaptel modules -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/e485b4ff/attachment-0002.vcf From andrew at hijacked.us Mon Nov 30 11:51:53 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 30 Nov 2009 14:51:53 -0500 Subject: [Freeswitch-users] Holiday routing examples Message-ID: <20091130195153.GD8574@hijacked.us> Tony committed my patch for doing 'week of month' conditions in the XML dialplan along with some holiday routing examples to the default dialplan. Now you can detect all the major US holidays in pure dialplan XML without having to do any nasty math or anything (I did it all for you). I've also added a page to the wiki describing how to use it for other dates (like non-US holidays): http://wiki.freeswitch.org/wiki/Holiday_Routing Hope this helps some people. Andrew From pjintheusa at gmail.com Mon Nov 30 14:03:12 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 30 Nov 2009 17:03:12 -0500 Subject: [Freeswitch-users] Holiday routing examples In-Reply-To: <20091130195153.GD8574@hijacked.us> References: <20091130195153.GD8574@hijacked.us> Message-ID: <367751820911301403m118bdd46g6c40d3d3492db5e3@mail.gmail.com> Thanks for this goodness. I am sure to use it so it is appreciated. On Mon, Nov 30, 2009 at 2:51 PM, Andrew Thompson wrote: > Tony committed my patch for doing 'week of month' conditions in the XML > dialplan along with some holiday routing examples to the default > dialplan. Now you can detect all the major US holidays in pure dialplan > XML without having to do any nasty math or anything (I did it all for > you). > > I've also added a page to the wiki describing how to use it for other > dates (like non-US holidays): > > http://wiki.freeswitch.org/wiki/Holiday_Routing > > Hope this helps some people. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/68c78249/attachment-0002.html From yehavi.bourvine at gmail.com Mon Nov 30 21:59:59 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 1 Dec 2009 07:59:59 +0200 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller Message-ID: Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing & OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/e0bc2364/attachment-0002.html From anthony.minessale at gmail.com Mon Nov 30 22:15:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 00:15:54 -0600 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> Message-ID: <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> Just set the variables effective_callee_id_name and effective_callee_id_number in your dp before you answer the call On Dec 1, 2009 12:08 AM, "Yehavi Bourvine" wrote: Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing & OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/5f707482/attachment-0002.html From yehavi.bourvine at gmail.com Mon Nov 30 22:42:09 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 1 Dec 2009 08:42:09 +0200 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> Message-ID: Hello Anthony, I think I did not explain myself correctly: The destination sends the Remote-Party-ID in the Ringing and OK replies, but they are not relayed to the original caller. Thanks! __Yehavi: 2009/12/1 Anthony Minessale > Just set the variables effective_callee_id_name and > effective_callee_id_number in your dp before you answer the call > > On Dec 1, 2009 12:08 AM, "Yehavi Bourvine" > wrote: > > Hello, > > I would like Freeswitch to pass the Remote-Party-ID field of the called > party (sent in the Ringing & OK when answering the call) back to the > originator's phone. How can I do that? > > The drive for this is: Our Freeswitch is connected via a Cisco gateway and > PRI to the university's phone exchange. When we call some university's > extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK > which includes the called party's name. I would like Freeswitch to relay > this to the caller so he/she can see the name of the one who they called. > > Thanks! __Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/6fc00f1b/attachment-0002.html From mrene_lists at avgs.ca Mon Nov 30 22:49:30 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 1 Dec 2009 01:49:30 -0500 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> Message-ID: Are you on SVN trunk? As far as I recall the callee_id_number/name stuff isnt in 1.0.4. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 1-Dec-09, at 1:42 AM, Yehavi Bourvine wrote: > Hello Anthony, > > I think I did not explain myself correctly: The destination sends > the Remote-Party-ID in the Ringing and OK replies, but they are not > relayed to the original caller. > > Thanks! __Yehavi: > > 2009/12/1 Anthony Minessale > Just set the variables effective_callee_id_name and > effective_callee_id_number in your dp before you answer the call > > >> On Dec 1, 2009 12:08 AM, "Yehavi Bourvine" >> wrote: >> >> Hello, >> >> I would like Freeswitch to pass the Remote-Party-ID field of the >> called party (sent in the Ringing & OK when answering the call) >> back to the originator's phone. How can I do that? >> >> The drive for this is: Our Freeswitch is connected via a Cisco >> gateway and PRI to the university's phone exchange. When we call >> some university's extension the Cisco gateway adds Remote-Party-ID >> field to the Ringing and OK which includes the called party's name. >> I would like Freeswitch to relay this to the caller so he/she can >> see the name of the one who they called. >> >> Thanks! __Yehavi: >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/8807a857/attachment-0002.html From yehavi.bourvine at gmail.com Mon Nov 30 23:04:07 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 1 Dec 2009 09:04:07 +0200 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> Message-ID: > Are you on SVN trunk? As far as I recall the callee_id_number/name stuff isnt in 1.0.4. No, because the SVN has problems with Emailing the voicemail... We use 1.0.4 and set sip_callee_id_number/name which works. I would like to not set it and get it from the other side... Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/4e160994/attachment-0002.html