From spencer at 5ninesolutions.com Tue Nov 1 00:07:23 2011 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Mon, 31 Oct 2011 14:07:23 -0700 Subject: [Freeswitch-users] Using Presence with Valet Parking In-Reply-To: References: <74536C50-3AF4-4456-8F19-E8DB58663BE3@5ninesolutions.com> Message-ID: <55DDF6EA-1226-4EF2-9EC5-B58404590C77@5ninesolutions.com> Absolutely. On Oct 31, 2011, at 1:17 PM, Michael Collins wrote: > Would you mind paying the wiki tax and writing up your scenario and posting a sample config? You can starting adding content on this page, since it needs some love anyway: > > http://wiki.freeswitch.org/wiki/Mod_valet_park > > Thanks, > MC > > On Mon, Oct 31, 2011 at 1:13 PM, Spencer Thomason wrote: > Hi Anthony, > Thanks for your help! This is working great very well now. > > Spencer > > On Oct 31, 2011, at 8:03 AM, Anthony Minessale wrote: > >> make sure DOMAIN is the default domain set in vars.xml otherwise name the lot with @domain.com right in it. >> Also make sure its the most recent rev of the mod. >> >> >> On Sun, Oct 30, 2011 at 9:44 PM, Spencer Thomason wrote: >> Hello, >> I noticed there have been some recent improvements to mod_valet_parking regarding presence. I was wondering if anyone had used this already and could shed some light on how to subscribe to the status of a particular slot. I'm testing this with a Cisco SPA-509G and can't seem to get it working. My dialplan looks like this: >> >> >> >> >> >> >> >> >> The phone is sending a SUBSCRIBE to park+71@ and gets a 202 Accepted response. >> >> I do have presence configured and working correctly for user extensions. When a call is transferred to 71, the parking works as expected but a NOTIFY is never sent to the phone. What am I missing? >> >> Thanks for any help! >> >> Spencer >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111031/9b839ca1/attachment.html From boris at tagnet.ru Tue Nov 1 05:39:28 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Tue, 01 Nov 2011 08:39:28 +0600 Subject: [Freeswitch-users] Linksys PAP2-T and echo In-Reply-To: <4EAEDDCA.4070907@coppice.org> References: <4EAEB77C.9060901@tagnet.ru> <4EAEBAB7.1030405@coppice.org> <3B1EABFB29D34F61BC96746169E2E2C4@DWP> <4EAECFFD.2090502@tagnet.ru> <4EAEDDCA.4070907@coppice.org> Message-ID: <4EAF5BE0.4050305@tagnet.ru> Hello! Steve, my configuration is: PAP2-T (A) ---- FreeSwitch --- PAP-2T (B) I tried Freeswitch in all modes: default, proxy_media, bypass_media. The echo is always present. User A hears its own echo, and user B hear its own. > Hi, > > The echo canceller in the SPA3102 is a disaster, but the ones in the > PAP2T and SPA2102 shouldn't give you problems. If you use the default > gains you shouldn't really notice echo. If you set some wild gains you > might bee poor results. Are you sure the echo originates from the PAP2T, > and not some other part of the signal chain? > > Steve > > > On 11/01/2011 12:42 AM, Boris Kovalenko wrote: >> Hello! >> >> Steve, Darcy, I've played ... no success :( Setting FXS to -15 do echo >> acceptable but I still can hear it. So there is no way to eleminate echo >> with Linksys devices? Only to buy more professional devices like Addpac? >>> on your pap2t web interface, go to voice, advanced, select the line, the >>> settings below will allow you to play with echo cancellation. However, you >>> also need to go the the regional tab in voice and play with the fxs input >>> and output gains, the voice gain on the pap2t or any linksys gateway, will >>> impact the echo on the line. This is most likely where your problem is. >>> You might not be able to cure the echo on every call, but you should be able >>> to make it acceptable. >>> >>> Darcy >>> >>> >>> Audio Configuration >>> Preferred Codec: Second Preferred Codec: >>> Third Preferred Codec: Use Pref Codec Only: >>> Silence Supp Enable: Silence Threshold: >>> G729a Enable: Echo Canc Enable: >>> G723 Enable: Echo Canc Adapt Enable: >>> G726-16 Enable: Echo Supp Enable: >>> G726-24 Enable: FAX CED Detect Enable: >>> G726-32 Enable: FAX CNG Detect Enable: >>> G726-40 Enable: FAX Passthru Codec: >>> DTMF Process INFO: FAX Codec Symmetric: >>> DTMF Process AVT: FAX Passthru Method: >>> DTMF Tx Method: DTMF Tx Mode: >>> DTMF Tx Strict Hold Off Time: FAX Process NSE: >>> Hook Flash Tx Method: FAX Disable ECAN: >>> Release Unused Codec: FAX Enable T38: >>> FAX T38 Redundancy: FAX Tone Detect Mode: >>> >>> -----Original Message----- >>> From: Steve Underwood >>> Sent: Monday, October 31, 2011 11:11 AM >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo >>> >>> On 10/31/2011 10:58 PM, Boris Kovalenko wrote: >>>> Hello! >>>> >>>> >>>> I'm wondering if it is possible to solve echo with Linksys PAP2-T >>>> devices. Does anybody here uses it? I read many articles and still can't >>>> understand how to eleminate echo :( I read >>>> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found >>>> that Asterisk has its own EC. Does Freeswitch has it too? I found this >>>> article http://docs.freeswitch.org/echo_can_page.html but can't >>>> understand - has Freeswitch EC or not? If yes - how to turn it on and off? >>> Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo >>> cancellation over IP is very problematic, and hardly ever attempted. The >>> PAP2T should be echo cancelling for itself, and they usually do a fairly >>> good job of this. I don't think it is configurable. Its always on. >>> >>> 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 >>> >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Regards, Boris From boris at tagnet.ru Tue Nov 1 05:41:47 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Tue, 01 Nov 2011 08:41:47 +0600 Subject: [Freeswitch-users] Linksys PAP2-T and echo In-Reply-To: References: <4EAEB77C.9060901@tagnet.ru> <4EAEBAB7.1030405@coppice.org><3B1EABFB29D34F61BC96746169E2E2C4@DWP> <4EAECFFD.2090502@tagnet.ru> Message-ID: <4EAF5C6B.5090006@tagnet.ru> Hello! Darcy, what is your firmware? 5.1.6? > Here are the settings from a gateway in the field, we have about 200 of them > installed with no echo compliants > > fax port gain, in and out -2 > set > echo canc enable yes > echo canc adapt enable yes > echo supp eanble yes > > Also, set silence supp enable to yes > and silence threshold to medium. > > Give that a try, I find it works quite well. These adapters are service by > a freeswitch with a 729 license. > > > -----Original Message----- > From: Boris Kovalenko > Sent: Monday, October 31, 2011 12:42 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo > > Hello! > > Steve, Darcy, I've played ... no success :( Setting FXS to -15 do echo > acceptable but I still can hear it. So there is no way to eleminate echo > with Linksys devices? Only to buy more professional devices like Addpac? >> on your pap2t web interface, go to voice, advanced, select the line, the >> settings below will allow you to play with echo cancellation. However, >> you >> also need to go the the regional tab in voice and play with the fxs input >> and output gains, the voice gain on the pap2t or any linksys gateway, will >> impact the echo on the line. This is most likely where your problem is. >> You might not be able to cure the echo on every call, but you should be >> able >> to make it acceptable. >> >> Darcy >> >> >> Audio Configuration >> Preferred Codec: Second Preferred Codec: >> Third Preferred Codec: Use Pref Codec Only: >> Silence Supp Enable: Silence Threshold: >> G729a Enable: Echo Canc Enable: >> G723 Enable: Echo Canc Adapt Enable: >> G726-16 Enable: Echo Supp Enable: >> G726-24 Enable: FAX CED Detect Enable: >> G726-32 Enable: FAX CNG Detect Enable: >> G726-40 Enable: FAX Passthru Codec: >> DTMF Process INFO: FAX Codec Symmetric: >> DTMF Process AVT: FAX Passthru Method: >> DTMF Tx Method: DTMF Tx Mode: >> DTMF Tx Strict Hold Off Time: FAX Process NSE: >> Hook Flash Tx Method: FAX Disable ECAN: >> Release Unused Codec: FAX Enable T38: >> FAX T38 Redundancy: FAX Tone Detect Mode: >> >> -----Original Message----- >> From: Steve Underwood >> Sent: Monday, October 31, 2011 11:11 AM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo >> >> On 10/31/2011 10:58 PM, Boris Kovalenko wrote: >>> Hello! >>> >>> >>> I'm wondering if it is possible to solve echo with Linksys PAP2-T >>> devices. Does anybody here uses it? I read many articles and still can't >>> understand how to eleminate echo :( I read >>> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found >>> that Asterisk has its own EC. Does Freeswitch has it too? I found this >>> article http://docs.freeswitch.org/echo_can_page.html but can't >>> understand - has Freeswitch EC or not? If yes - how to turn it on and >>> off? >> Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo >> cancellation over IP is very problematic, and hardly ever attempted. The >> PAP2T should be echo cancelling for itself, and they usually do a fairly >> good job of this. I don't think it is configurable. Its always on. >> >> 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 >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -- Regards, Boris From darcy at primrose.ws Tue Nov 1 06:15:28 2011 From: darcy at primrose.ws (Darcy) Date: Mon, 31 Oct 2011 23:15:28 -0400 Subject: [Freeswitch-users] Linksys PAP2-T and echo In-Reply-To: <4EAF5C6B.5090006@tagnet.ru> References: <4EAEB77C.9060901@tagnet.ru><4EAEBAB7.1030405@coppice.org><3B1EABFB29D34F61BC96746169E2E2C4@DWP><4EAECFFD.2090502@tagnet.ru> <4EAF5C6B.5090006@tagnet.ru> Message-ID: Yes, if it is older we always upgrade it to the 5.1.6. Also, when we do get an area where the echo is a little more difficult to cure, we use the spa2102, I do not know why but it seems to have more processing power, the fact that it will do t38 when the pap2t does not is probably an indication of the difference in two. Darc y -----Original Message----- From: Boris Kovalenko Sent: Monday, October 31, 2011 10:41 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo Hello! Darcy, what is your firmware? 5.1.6? > Here are the settings from a gateway in the field, we have about 200 of > them > installed with no echo compliants > > fax port gain, in and out -2 > set > echo canc enable yes > echo canc adapt enable yes > echo supp eanble yes > > Also, set silence supp enable to yes > and silence threshold to medium. > > Give that a try, I find it works quite well. These adapters are service > by > a freeswitch with a 729 license. > > > -----Original Message----- > From: Boris Kovalenko > Sent: Monday, October 31, 2011 12:42 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo > > Hello! > > Steve, Darcy, I've played ... no success :( Setting FXS to -15 do echo > acceptable but I still can hear it. So there is no way to eleminate echo > with Linksys devices? Only to buy more professional devices like Addpac? >> on your pap2t web interface, go to voice, advanced, select the line, the >> settings below will allow you to play with echo cancellation. However, >> you >> also need to go the the regional tab in voice and play with the fxs input >> and output gains, the voice gain on the pap2t or any linksys gateway, >> will >> impact the echo on the line. This is most likely where your problem is. >> You might not be able to cure the echo on every call, but you should be >> able >> to make it acceptable. >> >> Darcy >> >> >> Audio Configuration >> Preferred Codec: Second Preferred Codec: >> Third Preferred Codec: Use Pref Codec Only: >> Silence Supp Enable: Silence Threshold: >> G729a Enable: Echo Canc Enable: >> G723 Enable: Echo Canc Adapt Enable: >> G726-16 Enable: Echo Supp Enable: >> G726-24 Enable: FAX CED Detect Enable: >> G726-32 Enable: FAX CNG Detect Enable: >> G726-40 Enable: FAX Passthru Codec: >> DTMF Process INFO: FAX Codec Symmetric: >> DTMF Process AVT: FAX Passthru Method: >> DTMF Tx Method: DTMF Tx Mode: >> DTMF Tx Strict Hold Off Time: FAX Process NSE: >> Hook Flash Tx Method: FAX Disable ECAN: >> Release Unused Codec: FAX Enable T38: >> FAX T38 Redundancy: FAX Tone Detect Mode: >> >> -----Original Message----- >> From: Steve Underwood >> Sent: Monday, October 31, 2011 11:11 AM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo >> >> On 10/31/2011 10:58 PM, Boris Kovalenko wrote: >>> Hello! >>> >>> >>> I'm wondering if it is possible to solve echo with Linksys >>> PAP2-T >>> devices. Does anybody here uses it? I read many articles and still can't >>> understand how to eleminate echo :( I read >>> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found >>> that Asterisk has its own EC. Does Freeswitch has it too? I found this >>> article http://docs.freeswitch.org/echo_can_page.html but can't >>> understand - has Freeswitch EC or not? If yes - how to turn it on and >>> off? >> Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo >> cancellation over IP is very problematic, and hardly ever attempted. The >> PAP2T should be echo cancelling for itself, and they usually do a fairly >> good job of this. I don't think it is configurable. Its always on. >> >> 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 >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -- Regards, Boris FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From fieldpeak at gmail.com Tue Nov 1 11:49:50 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Tue, 1 Nov 2011 16:49:50 +0800 Subject: [Freeswitch-users] Log level of log file by fs_cli In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C59F2B77504@cooper> <76208DFD-604B-4C02-AD0B-4A6C36425134@gmail.com> Message-ID: Thanks MC, it works as you said, thanks. 2011/10/31 Michael Collins > Fieldpeak, > > Looks like you just confirmed what Steven A. said. Try his suggestion re: > "fsctl loglevel xxx" from within fs_cli and see what you get. > > -MC > > > On Fri, Oct 28, 2011 at 7:51 PM, fieldpeak wrote: > >> Hi Steven, >> >> i tried below cmd, and then make a call, inside the the log file, >> there is still 'DEBUG' level message logged. very strange... >> >> freeswitch at internal> fsctl loglevel alert >> +OK log level: ALERT [1] >> >> freeswitch at internal> >> >> inside freeswitch.log >> ... >> >> 2011-10-29 10:44:02.496776 [DEBUG] switch_core_state_machine.c:464 >> (sofia/internal/13580358068 at 124.193.106.104) Callstate Change HANGUP >> -> DOWN >> >> >> >> 2011/10/28, Steven Ayre : >> > Peter, that affects only the logging level of your fs_cli session - not >> the >> > log file. >> > >> > Fran, use the fsctl command from the cli. It's in mod_commands and >> > documented on the wiki. >> > >> > Eg: >> > fsctl loglevel >> > - shows current level >> > fsctl loglevel debug >> > - set to debug >> > >> > Steve on iPhone >> > >> > On 28 Oct 2011, at 11:09, Peter Olsson < >> peter.olsson at visionutveckling.se> >> > wrote: >> > >> >> Just enter ?/log [loglevel]? inside fs_cli. I think there is a startup >> >> parameter as well. >> >> >> >> /Peter >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r fieldpeak >> >> Skickat: den 28 oktober 2011 10:50 >> >> Till: FreeSWITCH-users >> >> ?mne: [Freeswitch-users] Log level of log file by fs_cli >> >> >> >> Hi friends, >> >> >> >> i know we can set the loglevel inside switch.conf.xml to change the log >> >> level of log files, >> >> >> >> however, i would like to change the log level of log files thru fs_cli >> on >> >> the fly(no need restart FS), is there any commands to support it, >> thanks. >> >> >> >> -- >> >> Regards, >> >> Charles >> >> >> >> !DSPAM:4eaa6d4932761466214554! >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> >> >> -- >> Regards, >> Charles >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/7005fddb/attachment-0001.html From john at whitesmiths.com Tue Nov 1 13:28:07 2011 From: john at whitesmiths.com (John O'Brien) Date: Tue, 1 Nov 2011 21:28:07 +1100 Subject: [Freeswitch-users] Access to sip_history_info from python Message-ID: Hi, I'm a freeswitch newbie. I have written a python script that processes calls that are diverted to it. Its sort of a messaging system that triggers some other python processes that run outside of mod_python. I want to from within the mod_python executed python code have access to the SIP header field History-Info. I can use this to help in determining how the call ended up been processed by my python code. The sofia variable sip_history_info is what I need access to but I'm at a loss to figure out how I can get to it from python. If I could get the entire SIP header I can probably parse that to extract the History-Info. Your help would be appreciated. Regards, John From lloyd.aloysius at gmail.com Tue Nov 1 15:48:48 2011 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 1 Nov 2011 08:48:48 -0400 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV Message-ID: Hi , I am testing the Polycom Phones [IP335,IP450 and IP6000] and DNS SRV records. All my DNS SRV records and other phone models [Cisico , Asstra, Yealink] working without any issue. But for Polycom Phones .. I need to specify the "Outbound Proxy" as IP address otherwise phone never register. * voIpProt.SIP.outboundProxy.address="IP ADDRESS"* I could not find why we need to specify "Outbound Proxy? How to avoid using IP address? Does Polycom phones fully support DNS SRV records? Any help is appreciated. Thanks and regards, Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/7e6d2884/attachment.html From yehavi.bourvine at gmail.com Tue Nov 1 16:08:41 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 1 Nov 2011 15:08:41 +0200 Subject: [Freeswitch-users] Oddities with Polycom's vesion 4.0.0 Message-ID: Hello, I've upgraded our Polycom phones from firmware 3.3.2 to 4.0.0 and have the following oddities: - On shared lines the call lists are "mixed": - No calls are in outgoing calls. - answered incoming and outgoing calls are placed in incoming calls. - unanswered incoming and outgoing calls are placed in the answered calls. - When the destination is busy there is no busy sign on the screen - the call is disconnected immediately and the phone returns to an idle state. I've opened today a support call with Polycom, and will update when I get replies. Regards, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/19f61d08/attachment.html From brian at freeswitch.org Tue Nov 1 16:11:42 2011 From: brian at freeswitch.org (Brian West) Date: Tue, 1 Nov 2011 08:11:42 -0500 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: I'm going to guess you have not set voIpProt.server.x.transport="DNSnaptr" and you're missing NAPTR records too. /b On Nov 1, 2011, at 7:48 AM, Lloyd Aloysius wrote: > Hi , > > I am testing the Polycom Phones [IP335,IP450 and IP6000] and DNS SRV > records. All my DNS SRV records and other phone models [Cisico , Asstra, > Yealink] working without any issue. > > But for Polycom Phones .. I need to specify the "Outbound Proxy" as IP > address otherwise phone never register. > > * voIpProt.SIP.outboundProxy.address="IP ADDRESS"* > > I could not find why we need to specify "Outbound Proxy? How to avoid > using IP address? > > Does Polycom phones fully support DNS SRV records? > > Any help is appreciated. > > > > Thanks and regards, > > Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/ed2b3896/attachment.html From lloyd.aloysius at sunteltech.ca Tue Nov 1 16:22:45 2011 From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius) Date: Tue, 1 Nov 2011 09:22:45 -0400 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: Brian, Thank you for the reply. I set the voIpProt.SIP.transport="DNSnaptr" and you're missing NAPTR records too. > > /b > > On Nov 1, 2011, at 7:48 AM, Lloyd Aloysius wrote: > > Hi , > > I am testing the Polycom Phones [IP335,IP450 and IP6000] and DNS SRV > records. All my DNS SRV records and other phone models [Cisico , Asstra, > Yealink] working without any issue. > > But for Polycom Phones .. I need to specify the "Outbound Proxy" as IP > address otherwise phone never register. > > * voIpProt.SIP.outboundProxy.address="IP ADDRESS"* > > > I could not find why we need to specify "Outbound Proxy? How to avoid > using IP address? > > Does Polycom phones fully support DNS SRV records? > > Any help is appreciated. > > > > Thanks and regards, > > Lloyd > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/3324702d/attachment.html From brian at freeswitch.org Tue Nov 1 16:27:39 2011 From: brian at freeswitch.org (Brian West) Date: Tue, 1 Nov 2011 08:27:39 -0500 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: Ok you missed the part about the NAPTR records didn't you? Make sure you setup your NAPTR records correctly. I really don't get how people think they are optional. I know this says optional http://wiki.freeswitch.org/wiki/SIP_TLS#Step_5_-_DNS_NAPTR_.26_SRV_Records_.28optional.29 but they aren't. /b On Nov 1, 2011, at 8:22 AM, Lloyd Aloysius wrote: > But this fails to register. > > Thanks > Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/7bcb95d0/attachment.html From brian at freeswitch.org Tue Nov 1 16:28:13 2011 From: brian at freeswitch.org (Brian West) Date: Tue, 1 Nov 2011 08:28:13 -0500 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: <5DD37248-BC5B-4EBB-B1E4-E627AF0AB420@freeswitch.org> Also what do your SRV records look like? I suspect you have them slightly wrong and the polycom is rather strict about them. /b On Nov 1, 2011, at 8:22 AM, Lloyd Aloysius wrote: > But this fails to register. > > Thanks > Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/12b2d9b2/attachment-0001.html From boris at tagnet.ru Tue Nov 1 17:11:37 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Tue, 01 Nov 2011 20:11:37 +0600 Subject: [Freeswitch-users] Linksys PAP2-T and echo In-Reply-To: References: <4EAEB77C.9060901@tagnet.ru><4EAEBAB7.1030405@coppice.org><3B1EABFB29D34F61BC96746169E2E2C4@DWP><4EAECFFD.2090502@tagnet.ru> <4EAF5C6B.5090006@tagnet.ru> Message-ID: <4EAFFE19.4070908@tagnet.ru> Hello! Played various parameters and echo still here. In other ways but still here. With silence suppression the echo persist if there is doubletalk. If only one is speaking there is no echo. The only what helps - I set FXS output to -15 and echo is gone. But remote party is quietly. > Yes, if it is older we always upgrade it to the 5.1.6. Also, when we do > get an area where the echo is a little more difficult to cure, we use the > spa2102, I do not know why but it seems to have more processing power, the > fact that it will do t38 when the pap2t does not is probably an indication > of the difference in two. > > Darc y > > -----Original Message----- > From: Boris Kovalenko > Sent: Monday, October 31, 2011 10:41 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo > > Hello! > > Darcy, what is your firmware? 5.1.6? > >> Here are the settings from a gateway in the field, we have about 200 of >> them >> installed with no echo compliants >> >> fax port gain, in and out -2 >> set >> echo canc enable yes >> echo canc adapt enable yes >> echo supp eanble yes >> >> Also, set silence supp enable to yes >> and silence threshold to medium. >> >> Give that a try, I find it works quite well. These adapters are service >> by >> a freeswitch with a 729 license. >> >> >> -----Original Message----- >> From: Boris Kovalenko >> Sent: Monday, October 31, 2011 12:42 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo >> >> Hello! >> >> Steve, Darcy, I've played ... no success :( Setting FXS to -15 do echo >> acceptable but I still can hear it. So there is no way to eleminate echo >> with Linksys devices? Only to buy more professional devices like Addpac? >>> on your pap2t web interface, go to voice, advanced, select the line, the >>> settings below will allow you to play with echo cancellation. However, >>> you >>> also need to go the the regional tab in voice and play with the fxs input >>> and output gains, the voice gain on the pap2t or any linksys gateway, >>> will >>> impact the echo on the line. This is most likely where your problem is. >>> You might not be able to cure the echo on every call, but you should be >>> able >>> to make it acceptable. >>> >>> Darcy >>> >>> >>> Audio Configuration >>> Preferred Codec: Second Preferred Codec: >>> Third Preferred Codec: Use Pref Codec Only: >>> Silence Supp Enable: Silence Threshold: >>> G729a Enable: Echo Canc Enable: >>> G723 Enable: Echo Canc Adapt Enable: >>> G726-16 Enable: Echo Supp Enable: >>> G726-24 Enable: FAX CED Detect Enable: >>> G726-32 Enable: FAX CNG Detect Enable: >>> G726-40 Enable: FAX Passthru Codec: >>> DTMF Process INFO: FAX Codec Symmetric: >>> DTMF Process AVT: FAX Passthru Method: >>> DTMF Tx Method: DTMF Tx Mode: >>> DTMF Tx Strict Hold Off Time: FAX Process NSE: >>> Hook Flash Tx Method: FAX Disable ECAN: >>> Release Unused Codec: FAX Enable T38: >>> FAX T38 Redundancy: FAX Tone Detect Mode: >>> >>> -----Original Message----- >>> From: Steve Underwood >>> Sent: Monday, October 31, 2011 11:11 AM >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo >>> >>> On 10/31/2011 10:58 PM, Boris Kovalenko wrote: >>>> Hello! >>>> >>>> >>>> I'm wondering if it is possible to solve echo with Linksys >>>> PAP2-T >>>> devices. Does anybody here uses it? I read many articles and still can't >>>> understand how to eleminate echo :( I read >>>> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found >>>> that Asterisk has its own EC. Does Freeswitch has it too? I found this >>>> article http://docs.freeswitch.org/echo_can_page.html but can't >>>> understand - has Freeswitch EC or not? If yes - how to turn it on and >>>> off? >>> Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo >>> cancellation over IP is very problematic, and hardly ever attempted. The >>> PAP2T should be echo cancelling for itself, and they usually do a fairly >>> good job of this. I don't think it is configurable. Its always on. >>> >>> 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 >>> >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > -- ? ?????????, ????? ????????? ??? "??????" ???. +7 (3435) 230001 ???? +7 (3435) 230005 From lloyd.aloysius at sunteltech.ca Tue Nov 1 17:13:42 2011 From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius) Date: Tue, 1 Nov 2011 10:13:42 -0400 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: Thank you for the reply . I miss the NAPTR records. Let me setup the NAPTR records and test again. Thanks and regards, Lloyd * * On Tue, Nov 1, 2011 at 9:27 AM, Brian West wrote: > Ok you missed the part about the NAPTR records didn't you? Make sure you > setup your NAPTR records correctly. I really don't get how people think > they are optional. > > I know this says optional > http://wiki.freeswitch.org/wiki/SIP_TLS#Step_5_-_DNS_NAPTR_.26_SRV_Records_.28optional.29 > > but they aren't. > > /b > > On Nov 1, 2011, at 8:22 AM, Lloyd Aloysius wrote: > > But this fails to register. > > Thanks > Lloyd > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/9f3eeb45/attachment.html From vipkilla at gmail.com Tue Nov 1 17:46:41 2011 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 1 Nov 2011 10:46:41 -0400 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: I have Polycoms working with DNS SRV without NAPTR records It was a matter of polycom provisioning files. I attached the 2 important files and that last important file named polycom_0004f21a0e87.cfg looks like this: On Tue, Nov 1, 2011 at 10:13 AM, Lloyd Aloysius wrote: > Thank you for the reply . > I miss the NAPTR records. Let me setup the NAPTR records and test again. > Thanks and regards, > > Lloyd > > > > On Tue, Nov 1, 2011 at 9:27 AM, Brian West wrote: >> >> Ok you missed the part about the NAPTR records didn't you? ?Make sure you >> setup your NAPTR records correctly. ?I really don't get how people think >> they are optional. >> I know this says >> optional?http://wiki.freeswitch.org/wiki/SIP_TLS#Step_5_-_DNS_NAPTR_.26_SRV_Records_.28optional.29 >> but they aren't. >> /b >> On Nov 1, 2011, at 8:22 AM, Lloyd Aloysius wrote: >> >> But this fails to register. >> >> Thanks >> Lloyd >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.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: sip.cfg Type: application/octet-stream Size: 197255 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/733c4581/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: 0004f21a0e87.cfg Type: application/octet-stream Size: 1237 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/733c4581/attachment-0003.obj From johnrose at google.hm Tue Nov 1 18:10:36 2011 From: johnrose at google.hm (John Rose) Date: Tue, 1 Nov 2011 11:10:36 -0400 Subject: [Freeswitch-users] Feedback In-Reply-To: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM> References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM> Message-ID: <004501cc98a8$698763b0$3c962b10$@google.hm> Making a Windows only version of Freeswitch doesn't make sense, especially a .NET only version. Freeswitch already provides great Windows and .NET support running either on or off Windows which is something that most other open source telephony applications do not have. And having the freedom to build it on Windows or Linux is a huge advantage and convenience. Having said that it would be interesting to see what you have done source code wise. but I'm afraid that you are dismissing the importance of one of Freeswitch's best features and I cringe at the direction you are taking. Being a Windows developer 90% of the time myself I do get embarrassed by Windows developer's OS arrogance sometimes and I see why so many Linux people only produce Linux based open source software. John From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chad Vogel Subject: [Freeswitch-users] Feedback Hello, I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users. Here are some of the changes we have made: * We converted FS core to C++ and having it compiling to .NET 4.0 CLI * FS now runs inside its own namespace * We replace FS_CLI with a PowerShell shell app * Supports SQL Server support via the Native SQL Server Client API * Support for windows clustering (up to 32 node active/active cluster) * Ties more directly into the Win32 API and has less reliance on open source Libs * Replaces OpenSSL sockets with Windows encrypted sockets * Added windows performance monitoring * No longer need to use mod_managed for managed modules * Memory management relies on .Net garbage collection * Added support for Microsoft Speech * VoiceXML 3.0 and SCXML support * No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages) * Fixed RTP clock timing issues in virtualized environments * New configuration file format using .Net App Configuration files (kind of looks like an IIS config file) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/94a9d173/attachment.html From xyangni at gmail.com Tue Nov 1 18:17:46 2011 From: xyangni at gmail.com (Yihui Li) Date: Tue, 1 Nov 2011 23:17:46 +0800 Subject: [Freeswitch-users] No sound when 2 client in the same NAT Message-ID: Hi, I run FS on a VPS with public IP, and 2 x-lite client on 2 of my home pc which are in the same NAT. They works fine if each of them calls FS sample ivr or bridged to other external phone. But when they call each other, the call is completely silent. The sip trace is listed below. Can anyone please help. Thanks. recv 767 bytes from udp/[90.192.85.6]:6216 at 15:08:19.500457: ------------------------------------------------------------------------ INVITE sip:1015 at 178.79.188.153 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport Max-Forwards: 70 Contact: To: "1015" From: "1017";tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 235 v=0 o=- 8 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 43926 RTP/AVP 0 8 3 101 a=alt:1 1 : tAU3AHbD K56reERa 90.192.85.6 43928 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 379 bytes to udp/[90.192.85.6]:6216 at 15:08:19.501004: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Content-Length: 0 ------------------------------------------------------------------------ send 869 bytes to udp/[90.192.85.6]:6216 at 15:08:19.502160: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" ;tag=K2eK1a4UZ0vvp Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="178.79.188.153", nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 326 bytes from udp/[90.192.85.6]:6216 at 15:08:19.541053: ------------------------------------------------------------------------ ACK sip:1015 at 178.79.188.153 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport To: "1015" ;tag=K2eK1a4UZ0vvp From: "1017";tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 1031 bytes from udp/[90.192.85.6]:6216 at 15:08:19.552533: ------------------------------------------------------------------------ INVITE sip:1015 at 178.79.188.153 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport Max-Forwards: 70 Contact: To: "1015" From: "1017";tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="1017",realm="178.79.188.153 ",nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0",uri="sip:1015 at 178.79.188.153 ",response="603f04a293a7535b109d9fae13085db8",cnonce="fc27cb770d3b4701e15f7aadc4690a64",nc=00000001,qop=auth,algorithm=MD5 User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 235 v=0 o=- 8 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 43926 RTP/AVP 0 8 3 101 a=alt:1 1 : tAU3AHbD K56reERa 90.192.85.6 43928 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 379 bytes to udp/[90.192.85.6]:6216 at 15:08:19.552996: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:08:19.545486 [NOTICE] switch_channel.c:915 New Channel sofia/internal/1017 at 178.79.188.153 [d46f4a05-6572-48e9-bea9-828e70addb74] 2011-11-01 15:08:19.545486 [INFO] mod_dialplan_xml.c:336 Processing 1017 <1017>->1015 in context default 2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *1 execute_extension::dx XML features 2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1017.2011-11-01-15-08-19.wav 2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *3 execute_extension::cf XML features 2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *4 execute_extension::att_xfer XML features 2011-11-01 15:08:19.545486 [NOTICE] switch_channel.c:915 New Channel sofia/internal/sip:1015 at 90.192.85.6:63216[b6fc17cc-1585-4e43-803d-f431c90fc196] send 1309 bytes to udp/[90.192.85.6]:63216 at 15:08:19.565528: ------------------------------------------------------------------------ INVITE sip:1015 at 90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKFyaDU717BB1XS Max-Forwards: 69 From: "Extension 1017" ;tag=Nm144052Sj91D To: Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 317 X-FS-Support: update_display Remote-Party-ID: "Extension 1017" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1320132861 1320132862 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27238 RTP/AVP 0 98 99 9 8 3 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 438 bytes from udp/[90.192.85.6]:63216 at 15:08:19.707081: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: To: ;tag=054e4538 From: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:08:19.705449 [NOTICE] sofia.c:5375 Ring-Ready sofia/internal/ sip:1015 at 90.192.85.6:63216! 2011-11-01 15:08:19.705449 [INFO] switch_ivr_originate.c:1115 Sending early media 2011-11-01 15:08:19.705449 [NOTICE] mod_sofia.c:2475 Pre-Answer sofia/internal/1017 at 178.79.188.153! send 1188 bytes to udp/[90.192.85.6]:6216 at 15:08:19.713171: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" ;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 251 Remote-Party-ID: "1015" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1320132951 1320132952 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27148 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 ------------------------------------------------------------------------ recv 733 bytes from udp/[90.192.85.6]:63216 at 15:08:23.699243: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: To: ;tag=054e4538 From: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 1 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 44478 RTP/AVP 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 428 bytes to udp/[90.192.85.6]:63216 at 15:08:23.700969: ------------------------------------------------------------------------ ACK sip:1015 at 90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKg735v2jB9KQgN Max-Forwards: 70 From: "Extension 1017" ;tag=Nm144052Sj91D To: ;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:08:23.685455 [NOTICE] sofia.c:5983 Channel [sofia/internal/ sip:1015 at 90.192.85.6:63216] has been answered send 1158 bytes to udp/[90.192.85.6]:6216 at 15:08:23.706877: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" ;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 251 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1320132951 1320132952 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27148 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 ------------------------------------------------------------------------ 2011-11-01 15:08:23.705453 [NOTICE] switch_ivr_originate.c:3194 Channel [sofia/internal/1017 at 178.79.188.153] has been answered send 1158 bytes to udp/[90.192.85.6]:6216 at 15:08:24.207213: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" ;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 251 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1320132951 1320132952 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27148 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 ------------------------------------------------------------------------ recv 733 bytes from udp/[90.192.85.6]:63216 at 15:08:24.380014: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: To: ;tag=054e4538 From: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 1 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 44478 RTP/AVP 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 428 bytes to udp/[90.192.85.6]:63216 at 15:08:24.380297: ------------------------------------------------------------------------ ACK sip:1015 at 90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKg735v2jB9KQgN Max-Forwards: 70 From: "Extension 1017" ;tag=Nm144052Sj91D To: ;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ recv 733 bytes from udp/[90.192.85.6]:63216 at 15:08:25.063077: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: To: ;tag=054e4538 From: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 1 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 44478 RTP/AVP 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 428 bytes to udp/[90.192.85.6]:63216 at 15:08:25.063384: ------------------------------------------------------------------------ ACK sip:1015 at 90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKg735v2jB9KQgN Max-Forwards: 70 From: "Extension 1017" ;tag=Nm144052Sj91D To: ;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ recv 711 bytes from udp/[90.192.85.6]:6216 at 15:08:25.157427: ------------------------------------------------------------------------ ACK sip:1015 at 178.79.188.153:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-52077a3eda00e179-1---d8754z-;rport Max-Forwards: 70 Contact: To: "1015";tag=mB8B35mZv9jFj From: "1017";tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 ACK Proxy-Authorization: Digest username="1017",realm="178.79.188.153 ",nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0",uri="sip:1015 at 178.79.188.153 ",response="603f04a293a7535b109d9fae13085db8",cnonce="fc27cb770d3b4701e15f7aadc4690a64",nc=00000001,qop=auth,algorithm=MD5 User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ recv 711 bytes from udp/[90.192.85.6]:6216 at 15:08:25.585421: ------------------------------------------------------------------------ ACK sip:1015 at 178.79.188.153:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216 ;branch=z9hG4bK-d8754z-52077a3eda00e179-1---d8754z-;rport Max-Forwards: 70 Contact: To: "1015";tag=mB8B35mZv9jFj From: "1017";tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 ACK Proxy-Authorization: Digest username="1017",realm="178.79.188.153 ",nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0",uri="sip:1015 at 178.79.188.153 ",response="603f04a293a7535b109d9fae13085db8",cnonce="fc27cb770d3b4701e15f7aadc4690a64",nc=00000001,qop=auth,algorithm=MD5 User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ recv 533 bytes from udp/[90.192.85.6]:63216 at 15:09:24.414921: ------------------------------------------------------------------------ BYE sip:mod_sofia at 178.79.188.153:5060 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:63216 ;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Extension 1017";tag=Nm144052Sj91D From: ;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: eyeBeam release 1102q stamp 51814 Reason: SIP;description="User Hung Up" Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:09:24.425460 [NOTICE] sofia.c:573 Hangup sofia/internal/ sip:1015 at 90.192.85.6:63216 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] send 576 bytes to udp/[90.192.85.6]:63216 at 15:09:24.426280: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:63216 ;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport=63216 From: ;tag=054e4538 To: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:09:24.445471 [NOTICE] switch_ivr_bridge.c:1355 Hangup sofia/internal/1017 at 178.79.188.153 [CS_EXECUTE] [NORMAL_CLEARING] 2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1395 Session 2 (sofia/internal/sip:1015 at 90.192.85.6:63216) Ended 2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/sip:1015 at 90.192.85.6:63216 [CS_DESTROY] send 623 bytes to udp/[90.192.85.6]:6216 at 15:09:24.448813: ------------------------------------------------------------------------ BYE sip:1017 at 90.192.85.6:6216 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKHgXyyX3e6vD3g Max-Forwards: 70 From: "1015" ;tag=mB8B35mZv9jFj To: "1017" ;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1395 Session 1 (sofia/internal/1017 at 178.79.188.153) Ended 2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/1017 at 178.79.188.153 [CS_DESTROY] recv 533 bytes from udp/[90.192.85.6]:63216 at 15:09:25.132183: ------------------------------------------------------------------------ BYE sip:mod_sofia at 178.79.188.153:5060 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:63216 ;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Extension 1017";tag=Nm144052Sj91D From: ;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: eyeBeam release 1102q stamp 51814 Reason: SIP;description="User Hung Up" Content-Length: 0 ------------------------------------------------------------------------ send 576 bytes to udp/[90.192.85.6]:63216 at 15:09:25.132451: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:63216 ;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport=63216 From: ;tag=054e4538 To: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 623 bytes to udp/[90.192.85.6]:6216 at 15:09:25.449077: ------------------------------------------------------------------------ BYE sip:1017 at 90.192.85.6:6216 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKHgXyyX3e6vD3g Max-Forwards: 70 From: "1015" ;tag=mB8B35mZv9jFj To: "1017" ;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 376 bytes from udp/[90.192.85.6]:6216 at 15:09:25.483104: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKHgXyyX3e6vD3g Contact: To: "1017";tag=ab30a26d From: "1015";tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ recv 376 bytes from udp/[90.192.85.6]:6216 at 15:09:26.386027: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKHgXyyX3e6vD3g Contact: To: "1017";tag=ab30a26d From: "1015";tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/2673998b/attachment-0001.html From helmut.kuper at ewetel.de Tue Nov 1 20:11:18 2011 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 01 Nov 2011 18:11:18 +0100 Subject: [Freeswitch-users] Linksys PAP2-T and echo In-Reply-To: <4EAFFE19.4070908@tagnet.ru> References: <4EAEB77C.9060901@tagnet.ru><4EAEBAB7.1030405@coppice.org><3B1EABFB29D34F61BC96746169E2E2C4@DWP><4EAECFFD.2090502@tagnet.ru> <4EAF5C6B.5090006@tagnet.ru> <4EAFFE19.4070908@tagnet.ru> Message-ID: <4EB02836.4080306@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I use Fax on PAP2T with FS successfully for a while now. My Config is like this: Fax on LINE 1 Preferred Codec = PCMA aka G711a // Valid for Germany ;) Use Pref Codec Only: yes FAX process NSE: no FAX Disable ECAN: yes Silence Supp Enable: no Echo Canc Enable: no Echo Canc Adapt Enable: no FAX CED Detect enable: no FAX CNG Detect enable: no FAX Passthrough Codec: G711a FAX Passthrough method: None Call Waiting Serv: no Network Jitter: high Jitter Buffer Adjustment: up and down On the Regional page: All three Polarities: Forward FXS Port Input gain: -3 FXS Port Output Gain: -3 FXS Port power Limit: 3 FXS Port Impedance: 600 //Germany :) More Echo Suppression: no That works on my side. Am 01.11.2011 15:11, schrieb Boris Kovalenko: > Hello! > > Played various parameters and echo still here. In other ways but > still here. With silence suppression the echo persist if there is > doubletalk. If only one is speaking there is no echo. The only what > helps - I set FXS output to -15 and echo is gone. But remote party > is quietly. >> Yes, if it is older we always upgrade it to the 5.1.6. Also, >> when we do get an area where the echo is a little more difficult >> to cure, we use the spa2102, I do not know why but it seems to >> have more processing power, the fact that it will do t38 when the >> pap2t does not is probably an indication of the difference in >> two. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk6wKDYACgkQ4tZeNddg3dxL0ACcDn/1gXS6kL4yQ1F5Gsb8JMA0 +wIAnjZLeK1ZCF4PL5xsmy20dLiqe3jq =lLQk -----END PGP SIGNATURE----- From msc at freeswitch.org Tue Nov 1 20:46:31 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Nov 2011 10:46:31 -0700 Subject: [Freeswitch-users] bridging calls from different sofia profiles In-Reply-To: References: Message-ID: Get a console log with a sip trace. Connect to your server with fs_cli and then issue this command: sofia global siptrace on Then make a test call capturing all the output. Repeat for each soft phone. Put this info on pastebin.freeswitch.org and use "FreeSWITCH Log" as the syntax highlighting. Put the pastebin (pb) URL in this email thread and hopefully the gang here will be able to help you diagnose what's going on. In the meantime your homework is to read up on a few things: http://wiki.freeswitch.org/wiki/Reporting_Bugs http://wiki.freeswitch.org/wiki/Packet_Capture http://wiki.freeswitch.org/wiki/Nat The information in those wiki pages will serve you well in your FreeSWITCH hacking. :) -MC On Mon, Oct 31, 2011 at 11:44 AM, Julien Chavanton wrote: > > Hi, I have audio problems when bridging calls from different sofia > profiles. > I use 2 profiles in order to connect softphone through a VPN to avoid NAT > problem > One profile is sending/listening on a private IP (for softphone) and the > other profile is using a public IP (for external PSTN gateways) > > I have tested with X-lite and Ekiga and I face different one way audio > with both: > With X-lite the audio received from the private IP is not forwarded > While with Ekiga the audio from the public IP profile is not forwarded > > I have traced SIP and RTP and the signaling looks good. RTP is received by > FS on the correct socket but not forwarded. > > I understand I do not provide enough information to diagnostic, but maybe > I could get input on how to diagnostic RTP socket etc. in this scenario ? > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/f75e7ba3/attachment.html From bhickey at italk.ie Tue Nov 1 20:53:13 2011 From: bhickey at italk.ie (Brian Hickey) Date: Tue, 01 Nov 2011 17:53:13 +0000 Subject: [Freeswitch-users] Linksys PAP2-T and echo In-Reply-To: <4EAF5BE0.4050305@tagnet.ru> References: <4EAEB77C.9060901@tagnet.ru> <4EAEBAB7.1030405@coppice.org> <3B1EABFB29D34F61BC96746169E2E2C4@DWP> <4EAECFFD.2090502@tagnet.ru> <4EAEDDCA.4070907@coppice.org> <4EAF5BE0.4050305@tagnet.ru> Message-ID: Hi Boris, what kind of analogue handsets do you have attached to the PAP2-T's? I have found with other ATA's (mostly Grandstreams) that it is the impedance settings for the handset that contributes the most to echo. Different handsets can be seup for different countries and sometimes you have to adjust the ATA to match. Regards Brian On 01/11/2011 02:39, Boris Kovalenko wrote: > Hello! > > Steve, my configuration is: > > PAP2-T (A) ---- FreeSwitch --- PAP-2T (B) > I tried Freeswitch in all modes: default, proxy_media, bypass_media. The > echo is always present. User A hears its own echo, and user B hear its own. > >> Hi, >> >> The echo canceller in the SPA3102 is a disaster, but the ones in the >> PAP2T and SPA2102 shouldn't give you problems. If you use the default >> gains you shouldn't really notice echo. If you set some wild gains you >> might bee poor results. Are you sure the echo originates from the PAP2T, >> and not some other part of the signal chain? >> >> Steve >> >> >> On 11/01/2011 12:42 AM, Boris Kovalenko wrote: >>> Hello! >>> >>> Steve, Darcy, I've played ... no success :( Setting FXS to -15 do echo >>> acceptable but I still can hear it. So there is no way to eleminate echo >>> with Linksys devices? Only to buy more professional devices like Addpac? >>>> on your pap2t web interface, go to voice, advanced, select the line, the >>>> settings below will allow you to play with echo cancellation. However, you >>>> also need to go the the regional tab in voice and play with the fxs input >>>> and output gains, the voice gain on the pap2t or any linksys gateway, will >>>> impact the echo on the line. This is most likely where your problem is. >>>> You might not be able to cure the echo on every call, but you should be able >>>> to make it acceptable. >>>> >>>> Darcy >>>> >>>> >>>> Audio Configuration >>>> Preferred Codec: Second Preferred Codec: >>>> Third Preferred Codec: Use Pref Codec Only: >>>> Silence Supp Enable: Silence Threshold: >>>> G729a Enable: Echo Canc Enable: >>>> G723 Enable: Echo Canc Adapt Enable: >>>> G726-16 Enable: Echo Supp Enable: >>>> G726-24 Enable: FAX CED Detect Enable: >>>> G726-32 Enable: FAX CNG Detect Enable: >>>> G726-40 Enable: FAX Passthru Codec: >>>> DTMF Process INFO: FAX Codec Symmetric: >>>> DTMF Process AVT: FAX Passthru Method: >>>> DTMF Tx Method: DTMF Tx Mode: >>>> DTMF Tx Strict Hold Off Time: FAX Process NSE: >>>> Hook Flash Tx Method: FAX Disable ECAN: >>>> Release Unused Codec: FAX Enable T38: >>>> FAX T38 Redundancy: FAX Tone Detect Mode: >>>> >>>> -----Original Message----- >>>> From: Steve Underwood >>>> Sent: Monday, October 31, 2011 11:11 AM >>>> To: FreeSWITCH Users Help >>>> Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo >>>> >>>> On 10/31/2011 10:58 PM, Boris Kovalenko wrote: >>>>> Hello! >>>>> >>>>> >>>>> I'm wondering if it is possible to solve echo with Linksys PAP2-T >>>>> devices. Does anybody here uses it? I read many articles and still can't >>>>> understand how to eleminate echo :( I read >>>>> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found >>>>> that Asterisk has its own EC. Does Freeswitch has it too? I found this >>>>> article http://docs.freeswitch.org/echo_can_page.html but can't >>>>> understand - has Freeswitch EC or not? If yes - how to turn it on and off? >>>> Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo >>>> cancellation over IP is very problematic, and hardly ever attempted. The >>>> PAP2T should be echo cancelling for itself, and they usually do a fairly >>>> good job of this. I don't think it is configurable. Its always on. >>>> >>>> Steve From nasida at live.ru Tue Nov 1 21:14:32 2011 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 1 Nov 2011 22:14:32 +0400 Subject: [Freeswitch-users] double dtmf Message-ID: Hello community, I have some issue with delivery of dtmf.The call flow is:one sip provider - FS - second sip provider - endpoint which expects correct dtmf On FS I have simple custom lua application which implement logic of calling cards system. I have not issues when I call to lua application. I can enter destination number (by using dtmf apparently) and all works fine. But when I try to send dtmf second time for endpoint which expects correct dtmf I have issue with double dtmf. It is noted that when I call from sip device I have not ANY issues. But I have issue when I call from any cell phone or landline phone. Any ideas are welcomed. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/ce3c1a73/attachment.html From hynek.cihlar at gmail.com Tue Nov 1 21:39:15 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Tue, 1 Nov 2011 19:39:15 +0100 Subject: [Freeswitch-users] Call quality Message-ID: <-4241905887276648699@unknownmsgid> Would anybody know, if the call quality could be dependant on the number of media switches the call is routed through? Currently I'm testing a system with several freeswitches in the call way (up to 4) and I'm noticing very short gaps, not very disturbing but noticable. What's your experience? Sent from my mobile device From helmut.kuper at ewetel.de Tue Nov 1 21:54:26 2011 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 01 Nov 2011 19:54:26 +0100 Subject: [Freeswitch-users] Linksys PAP2-T and echo In-Reply-To: <4EB02836.4080306@ewetel.de> References: <4EAEB77C.9060901@tagnet.ru><4EAEBAB7.1030405@coppice.org><3B1EABFB29D34F61BC96746169E2E2C4@DWP><4EAECFFD.2090502@tagnet.ru> <4EAF5C6B.5090006@tagnet.ru> <4EAFFE19.4070908@tagnet.ru> <4EB02836.4080306@ewetel.de> Message-ID: <4EB04062.9060500@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ups, sorry for my reply. Somehow I thought u had a problem with Fax ... tststs Am 01.11.2011 18:11, schrieb Helmut Kuper: > Hi, > > I use Fax on PAP2T with FS successfully for a while now. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk6wQGIACgkQ4tZeNddg3dy83gCfa6UuAipzGnVvtSuudyZcj/S4 L94AnjxXl1omSoXr0O+p/Ur06yHJZ9h5 =UQiC -----END PGP SIGNATURE----- From avi at avimarcus.net Tue Nov 1 22:22:11 2011 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 1 Nov 2011 21:22:11 +0200 Subject: [Freeswitch-users] double dtmf In-Reply-To: References: Message-ID: "when I try to send dtmf second time for endpoint " - are you doing something explicitly to send the DTMF? I just ignore the dtmf, and when FS bridges it sends the call on properly. -Avi 2011/11/1 Yuriy Nasida > Hello community, > > I have some issue with delivery of dtmf. > The call flow is: > one sip provider - FS - second sip provider - endpoint which expects > correct dtmf > > On FS I have simple custom lua application which implement logic of > calling cards system. > > I have not issues when I call to lua application. I can enter destination > number (by using dtmf apparently) and all works fine. But when I try to > send dtmf second time for endpoint which expects correct dtmf I have issue > with double dtmf. > > It is noted that when I call from sip device I have not ANY issues. But I > have issue when I call from any cell phone or landline phone. > > Any ideas are welcomed. > > 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/20111101/a0a8cb0e/attachment-0001.html From rsaavedra at ecogizmos.com Tue Nov 1 22:43:49 2011 From: rsaavedra at ecogizmos.com (rsaavedra at ecogizmos.com) Date: Tue, 1 Nov 2011 14:43:49 -0500 Subject: [Freeswitch-users] mod_rtmp login In-Reply-To: <4EA86414.3040903@livecall.com> References: <0e6a9dcb6a5cb714cd6a130ec6497142.squirrel@emailmg.globat.com> <4EA77BD1.7080302@livecall.com> <1f1fc7140c13295f010812918da115bf.squirrel@emailmg.globat.com> <4EA86414.3040903@livecall.com> Message-ID: <5e0d82ec10c5ba170c566d22b449a71a.squirrel@emailmg.globat.com> Thank you, I'm using on the flex client the user 1002 at 1.2.3.4, the user 1000 at 1.2.3.4 on a sip hard phone (aastra 480). My problem in this moment is: the user 1002 can hear but the user 1000 can't hear. Best regards, Ricardo Saavedra > That would be correct if your FreeSwitch server IP address is 1.2.3.4 > > > On 10/26/2011 10:04 AM, rsaavedra at ecogizmos.com wrote: >> Thank you, >> >> so in the flex client I will write the user like 1000 at 1.2.3.4 >> >> Best regards, >> >> Ricardo >> >>> Ricardo, >>> If your 1000 is default it is probably set up with >>> >>> >>> >>> if you set the param in rtmp.conf.xml to match the context and then do >>> a reloadxml or restart freeswitch it may authorize for you. >>> >>> >>> Hope that helps! >>> Jack >>> >>> >>> >>> On 10/25/2011 7:16 PM, rsaavedra at ecogizmos.com wrote: >>>> Hello, >>>> >>>> I'm trying to use mod_rtmp but everytime I tried on the console I see >>>> User >>>> not authorized. >>>> I was trying with user:1000 and password:1234. >>>> >>>> In the file rtmp.conf.xml I have all by default: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Best regards, >>>> >>>> Ricardo Saavedra >>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nasida at live.ru Tue Nov 1 22:56:49 2011 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 1 Nov 2011 23:56:49 +0400 Subject: [Freeswitch-users] double dtmf In-Reply-To: References: , Message-ID: Thank you for your answer. "are you doing something explicitly to send the DTMF"No. I just press buttons in my cell phone (I mean startpoint). In my opinion it is strange. I make answer() in lua script and use session:getDigits(). All works fine.In the end of script I use execute_extension which implements bridge.session:execute("execute_extension", "".. recieved_digit .." XML ".. user_context .."") Thus the call was answered in lua script. Next the call go to endpoint. How dtmf must be transmitted in this case ? Why I have not issues when I call from sip devices? Any help are welcomed. Thanks. From: avi at avimarcus.net Date: Tue, 1 Nov 2011 21:22:11 +0200 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] double dtmf "when I try to send dtmf second time for endpoint " - are you doing something explicitly to send the DTMF? I just ignore the dtmf, and when FS bridges it sends the call on properly. -Avi 2011/11/1 Yuriy Nasida Hello community, I have some issue with delivery of dtmf.The call flow is: one sip provider - FS - second sip provider - endpoint which expects correct dtmf On FS I have simple custom lua application which implement logic of calling cards system. I have not issues when I call to lua application. I can enter destination number (by using dtmf apparently) and all works fine. But when I try to send dtmf second time for endpoint which expects correct dtmf I have issue with double dtmf. It is noted that when I call from sip device I have not ANY issues. But I have issue when I call from any cell phone or landline phone. Any ideas are welcomed. 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/20111101/219bc726/attachment.html From lloyd.aloysius at gmail.com Tue Nov 1 23:57:26 2011 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 1 Nov 2011 16:57:26 -0400 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: Vik, I have the same setting in my configuration files. But the phone never register. I will follow the Brian instructions [create NAPTR Records] shortly. Thanks Lloyd On Tue, Nov 1, 2011 at 10:46 AM, Vik Killa wrote: > I have Polycoms working with DNS SRV without NAPTR records > It was a matter of polycom provisioning files. > I attached the 2 important files and that last important file named > polycom_0004f21a0e87.cfg looks like this: > > > reg.1.displayName="Test UA" > reg.1.address="1001" > reg.1.label="Test UA" > reg.1.type="private" > reg.1.auth.userId="1001" > reg.1.auth.password="PASSWORD" > reg.1.server.1.address="mypbx.mydomain.com" > /> > > > > On Tue, Nov 1, 2011 at 10:13 AM, Lloyd Aloysius > wrote: > > Thank you for the reply . > > I miss the NAPTR records. Let me setup the NAPTR records and test again. > > Thanks and regards, > > > > Lloyd > > > > > > > > On Tue, Nov 1, 2011 at 9:27 AM, Brian West wrote: > >> > >> Ok you missed the part about the NAPTR records didn't you? Make sure > you > >> setup your NAPTR records correctly. I really don't get how people think > >> they are optional. > >> I know this says > >> optional > http://wiki.freeswitch.org/wiki/SIP_TLS#Step_5_-_DNS_NAPTR_.26_SRV_Records_.28optional.29 > >> but they aren't. > >> /b > >> On Nov 1, 2011, at 8:22 AM, Lloyd Aloysius wrote: > >> > >> But this fails to register. > >> > >> Thanks > >> Lloyd > >> > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111101/6033be0a/attachment.html From vellayappan.n at mobax.com Wed Nov 2 08:14:08 2011 From: vellayappan.n at mobax.com (Vellayappan.N) Date: Wed, 2 Nov 2011 10:44:08 +0530 Subject: [Freeswitch-users] double dtmf In-Reply-To: References: Message-ID: Hi, I too have the same issue. When calling from PSTN number, the freeswitch says Incoming call from "0000000000" instead of the actual number. Moreover it detects only few digits (say 2 or 3) not all the dtmf numbers. Please let me know if you have addressed this issue. 2011/11/2 Yuriy Nasida > Thank you for your answer. > > "are you doing something explicitly to send the DTMF" > No. I just press buttons in my cell phone (I mean startpoint). > > In my opinion it is strange. I make answer() in lua script and use > session:getDigits(). All works fine. > In the end of script I use execute_extension which implements bridge. > session:execute("execute_extension", "".. recieved_digit .." XML ".. > user_context .."") > > Thus the call was answered in lua script. Next the call go to endpoint. > How dtmf must be transmitted in this case ? > > Why I have not issues when I call from sip devices? > > Any help are welcomed. > > Thanks. > > ------------------------------ > From: avi at avimarcus.net > Date: Tue, 1 Nov 2011 21:22:11 +0200 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] double dtmf > > > "when I try to send dtmf second time for endpoint " - are you doing > something explicitly to send the DTMF? > I just ignore the dtmf, and when FS bridges it sends the call on properly. > > -Avi > > > 2011/11/1 Yuriy Nasida > > Hello community, > > I have some issue with delivery of dtmf. > The call flow is: > one sip provider - FS - second sip provider - endpoint which expects > correct dtmf > > On FS I have simple custom lua application which implement logic of > calling cards system. > > I have not issues when I call to lua application. I can enter destination > number (by using dtmf apparently) and all works fine. But when I try to > send dtmf second time for endpoint which expects correct dtmf I have issue > with double dtmf. > > It is noted that when I call from sip device I have not ANY issues. But I > have issue when I call from any cell phone or landline phone. > > Any ideas are welcomed. > > 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 > > -- Regards Vellayappan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/e9b2614a/attachment-0001.html From govoiper at gmail.com Wed Nov 2 08:42:57 2011 From: govoiper at gmail.com (Sammy Govind) Date: Wed, 2 Nov 2011 10:42:57 +0500 Subject: [Freeswitch-users] Call quality In-Reply-To: <-4241905887276648699@unknownmsgid> References: <-4241905887276648699@unknownmsgid> Message-ID: Network latency, transcoding processing, jitter handling, call session establishment on each Media-Server/switches etc all add up in overall call quality factor. On Tue, Nov 1, 2011 at 11:39 PM, Hynek Cihlar wrote: > Would anybody know, if the call quality could be dependant on the > number of media switches the call is routed through? > > Currently I'm testing a system with several freeswitches in the call > way (up to 4) and I'm noticing very short gaps, not very disturbing > but noticable. > > What's your experience? > > Sent from my mobile device > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/3d8e68f1/attachment.html From hynek.cihlar at gmail.com Wed Nov 2 15:09:58 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Wed, 2 Nov 2011 13:09:58 +0100 Subject: [Freeswitch-users] Call quality In-Reply-To: References: <-4241905887276648699@unknownmsgid> Message-ID: In this case, it's the "ideal" world. Testing environment, all media servers on one OS, one native server box, CPU idle, no load, no transcoding. One media switch is connected to a VOIP provider though through SIP. But the provider sits in the same telehouse and again no transcoding takes place. Any idea where to direct the investigation? Hynek On Wed, Nov 2, 2011 at 6:42 AM, Sammy Govind wrote: > Network latency, transcoding processing, jitter handling, call session > establishment on each Media-Server/switches etc all add up in overall call > quality factor. > > On Tue, Nov 1, 2011 at 11:39 PM, Hynek Cihlar wrote: > >> Would anybody know, if the call quality could be dependant on the >> number of media switches the call is routed through? >> >> Currently I'm testing a system with several freeswitches in the call >> way (up to 4) and I'm noticing very short gaps, not very disturbing >> but noticable. >> >> What's your experience? >> >> Sent from my mobile device >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/b046b90f/attachment.html From hynek.cihlar at gmail.com Wed Nov 2 15:39:41 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Wed, 2 Nov 2011 13:39:41 +0100 Subject: [Freeswitch-users] Public IP in Contact header Message-ID: Can anybody please point me to settings or a freeswitch state that makes sofia put public IP into the Contact header? Currently I have SIP clients behind NAT registering to freeswitch. Freeswitch registers these clients with their respective private IPs. Bellow is the profile status. ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 178.238.X.Y,178.238.X.Y Dialplan XML Context default Challenge Realm auto_from RTP-IP 178.238.X.Y Ext-RTP-IP 178.238.X.Y SIP-IP 178.238.X.Y Ext-SIP-IP 178.238.X.Y URL sip:mod_sofia at 178.238.X.Y:5060 BIND-URL sip:mod_sofia at 178.238.X.Y:5060;maddr=178.238.X.Y HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU,GSM,G7221 at 32000h,G7221 at 16000h,G722 CODECS OUT PCMA,PCMU,GSM,G7221 at 32000h,G7221 at 16000h,G722 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: 3454998XYZ at 192_168_59_102 User: 1 at ribtjuhd Contact: "383372XYZ" Agent: A580 IP/022270000000 Status: Registered(UDP)(unknown) EXP(2011-11-02 13:30:43) EXPSECS(100) Host: eva IP: 81.90.X.Y Port: 22055 Auth-User: 1 Auth-Realm: ribtjuhd MWI-Account: 1 at ribtjuhd Thanks! Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/81144c04/attachment.html From govoiper at gmail.com Wed Nov 2 15:40:48 2011 From: govoiper at gmail.com (Sammy Govind) Date: Wed, 2 Nov 2011 17:40:48 +0500 Subject: [Freeswitch-users] Call quality In-Reply-To: References: <-4241905887276648699@unknownmsgid> Message-ID: Please explain your query bit more, keeping in mind the ideal world scenario as your's, what exactly you require to know. On Wed, Nov 2, 2011 at 5:09 PM, Hynek Cihlar wrote: > In this case, it's the "ideal" world. Testing environment, all media > servers on one OS, one native server box, CPU idle, no load, no > transcoding. One media switch is connected to a VOIP provider though > through SIP. But the provider sits in the same telehouse and again no > transcoding takes place. > > Any idea where to direct the investigation? > > Hynek > > > > > On Wed, Nov 2, 2011 at 6:42 AM, Sammy Govind wrote: > >> Network latency, transcoding processing, jitter handling, call session >> establishment on each Media-Server/switches etc all add up in overall call >> quality factor. >> >> On Tue, Nov 1, 2011 at 11:39 PM, Hynek Cihlar wrote: >> >>> Would anybody know, if the call quality could be dependant on the >>> number of media switches the call is routed through? >>> >>> Currently I'm testing a system with several freeswitches in the call >>> way (up to 4) and I'm noticing very short gaps, not very disturbing >>> but noticable. >>> >>> What's your experience? >>> >>> Sent from my mobile device >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/b1a50c6a/attachment-0001.html From hynek.cihlar at gmail.com Wed Nov 2 15:56:50 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Wed, 2 Nov 2011 13:56:50 +0100 Subject: [Freeswitch-users] Call quality In-Reply-To: References: <-4241905887276648699@unknownmsgid> Message-ID: The motivation is to fix the audio distortions. The architecture is as follows. One freeswitch acting as a "gateway" with configured sip gateway connected to a VOIP provider. Second freeswitch acting as a "controller" conects to the "gateway". Incoming call passes the "gateway" to the "controller" where it is handled with ESL. Routing is decided in an ESL application and in its simplest case is forwarded back to the "gateway" through the VOIP provider to another PSTN number. The forwarding is implemented in the "controller" such that uuid_originate is issued and then bridged with the incoming call. No transocding should take place, neither between the"gateway" and VOIP provider nor between the "controller" and the "gateway". Although no transcoding takes place and the system is idle, still there are little distortions in the audio. Distortions that don't exist if I connect to the VOIP directly with a SIP phone. I'm looking for a way to find out, where the distortion is introduced. Hynek On Wed, Nov 2, 2011 at 1:40 PM, Sammy Govind wrote: > Please explain your query bit more, keeping in mind the ideal world > scenario as your's, what exactly you require to know. > > > On Wed, Nov 2, 2011 at 5:09 PM, Hynek Cihlar wrote: > >> In this case, it's the "ideal" world. Testing environment, all media >> servers on one OS, one native server box, CPU idle, no load, no >> transcoding. One media switch is connected to a VOIP provider though >> through SIP. But the provider sits in the same telehouse and again no >> transcoding takes place. >> >> Any idea where to direct the investigation? >> >> Hynek >> >> >> >> >> On Wed, Nov 2, 2011 at 6:42 AM, Sammy Govind wrote: >> >>> Network latency, transcoding processing, jitter handling, call session >>> establishment on each Media-Server/switches etc all add up in overall call >>> quality factor. >>> >>> On Tue, Nov 1, 2011 at 11:39 PM, Hynek Cihlar wrote: >>> >>>> Would anybody know, if the call quality could be dependant on the >>>> number of media switches the call is routed through? >>>> >>>> Currently I'm testing a system with several freeswitches in the call >>>> way (up to 4) and I'm noticing very short gaps, not very disturbing >>>> but noticable. >>>> >>>> What's your experience? >>>> >>>> Sent from my mobile device >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/2caf1c58/attachment.html From govoiper at gmail.com Wed Nov 2 16:15:03 2011 From: govoiper at gmail.com (Sammy Govind) Date: Wed, 2 Nov 2011 18:15:03 +0500 Subject: [Freeswitch-users] Call quality In-Reply-To: References: <-4241905887276648699@unknownmsgid> Message-ID: Please be sure that no-transcoding is done. Make sure the codecs of inbound and outbound legs are same on provider<->gateway and then gateway<->controller and way back. If transcoding isn't the case, in my experiences virtual machines show these little distortions in call. On Wed, Nov 2, 2011 at 5:56 PM, Hynek Cihlar wrote: > The motivation is to fix the audio distortions. > > The architecture is as follows. One freeswitch acting as a "gateway" with > configured sip gateway connected to a VOIP provider. Second freeswitch > acting as a "controller" conects to the "gateway". Incoming call passes the > "gateway" to the "controller" where it is handled with ESL. Routing is > decided in an ESL application and in its simplest case is forwarded back to > the "gateway" through the VOIP provider to another PSTN number. The > forwarding is implemented in the "controller" such that uuid_originate is > issued and then bridged with the incoming call. No transocding should take > place, neither between the"gateway" and VOIP provider nor between the > "controller" and the "gateway". > > Although no transcoding takes place and the system is idle, still there > are little distortions in the audio. Distortions that don't exist if I > connect to the VOIP directly with a SIP phone. > > I'm looking for a way to find out, where the distortion is introduced. > > Hynek > > > > > On Wed, Nov 2, 2011 at 1:40 PM, Sammy Govind wrote: > >> Please explain your query bit more, keeping in mind the ideal world >> scenario as your's, what exactly you require to know. >> >> >> On Wed, Nov 2, 2011 at 5:09 PM, Hynek Cihlar wrote: >> >>> In this case, it's the "ideal" world. Testing environment, all media >>> servers on one OS, one native server box, CPU idle, no load, no >>> transcoding. One media switch is connected to a VOIP provider though >>> through SIP. But the provider sits in the same telehouse and again no >>> transcoding takes place. >>> >>> Any idea where to direct the investigation? >>> >>> Hynek >>> >>> >>> >>> >>> On Wed, Nov 2, 2011 at 6:42 AM, Sammy Govind wrote: >>> >>>> Network latency, transcoding processing, jitter handling, call session >>>> establishment on each Media-Server/switches etc all add up in overall call >>>> quality factor. >>>> >>>> On Tue, Nov 1, 2011 at 11:39 PM, Hynek Cihlar wrote: >>>> >>>>> Would anybody know, if the call quality could be dependant on the >>>>> number of media switches the call is routed through? >>>>> >>>>> Currently I'm testing a system with several freeswitches in the call >>>>> way (up to 4) and I'm noticing very short gaps, not very disturbing >>>>> but noticable. >>>>> >>>>> What's your experience? >>>>> >>>>> Sent from my mobile device >>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/4d3b1c37/attachment.html From avi at avimarcus.net Wed Nov 2 16:25:54 2011 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 2 Nov 2011 15:25:54 +0200 Subject: [Freeswitch-users] Call quality In-Reply-To: References: <-4241905887276648699@unknownmsgid> Message-ID: In practice, internally there is little reason to have each machine actually handling the media. Set bypass_media=true before bridging the call. This doesn't address the issue, though, if there is one. Can you see which machine introduces the distortion? e.g. Is it ok if you only use 3? -Avi On Wed, Nov 2, 2011 at 3:15 PM, Sammy Govind wrote: > Please be sure that no-transcoding is done. Make sure the codecs of > inbound and outbound legs are same on provider<->gateway and then > gateway<->controller and way back. > If transcoding isn't the case, in my experiences virtual machines show > these little distortions in call. > > > On Wed, Nov 2, 2011 at 5:56 PM, Hynek Cihlar wrote: > >> The motivation is to fix the audio distortions. >> >> The architecture is as follows. One freeswitch acting as a "gateway" with >> configured sip gateway connected to a VOIP provider. Second freeswitch >> acting as a "controller" conects to the "gateway". Incoming call passes the >> "gateway" to the "controller" where it is handled with ESL. Routing is >> decided in an ESL application and in its simplest case is forwarded back to >> the "gateway" through the VOIP provider to another PSTN number. The >> forwarding is implemented in the "controller" such that uuid_originate is >> issued and then bridged with the incoming call. No transocding should take >> place, neither between the"gateway" and VOIP provider nor between the >> "controller" and the "gateway". >> >> Although no transcoding takes place and the system is idle, still there >> are little distortions in the audio. Distortions that don't exist if I >> connect to the VOIP directly with a SIP phone. >> >> I'm looking for a way to find out, where the distortion is introduced. >> >> Hynek >> >> >> >> >> On Wed, Nov 2, 2011 at 1:40 PM, Sammy Govind wrote: >> >>> Please explain your query bit more, keeping in mind the ideal world >>> scenario as your's, what exactly you require to know. >>> >>> >>> On Wed, Nov 2, 2011 at 5:09 PM, Hynek Cihlar wrote: >>> >>>> In this case, it's the "ideal" world. Testing environment, all media >>>> servers on one OS, one native server box, CPU idle, no load, no >>>> transcoding. One media switch is connected to a VOIP provider though >>>> through SIP. But the provider sits in the same telehouse and again no >>>> transcoding takes place. >>>> >>>> Any idea where to direct the investigation? >>>> >>>> Hynek >>>> >>>> >>>> >>>> >>>> On Wed, Nov 2, 2011 at 6:42 AM, Sammy Govind wrote: >>>> >>>>> Network latency, transcoding processing, jitter handling, call session >>>>> establishment on each Media-Server/switches etc all add up in overall call >>>>> quality factor. >>>>> >>>>> On Tue, Nov 1, 2011 at 11:39 PM, Hynek Cihlar wrote: >>>>> >>>>>> Would anybody know, if the call quality could be dependant on the >>>>>> number of media switches the call is routed through? >>>>>> >>>>>> Currently I'm testing a system with several freeswitches in the call >>>>>> way (up to 4) and I'm noticing very short gaps, not very disturbing >>>>>> but noticable. >>>>>> >>>>>> What's your experience? >>>>>> >>>>>> Sent from my mobile device >>>>>> >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/9a0c6734/attachment-0001.html From yehavi.bourvine at gmail.com Wed Nov 2 17:39:56 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 2 Nov 2011 16:39:56 +0200 Subject: [Freeswitch-users] Oddities with Polycom's vesion 4.0.0 In-Reply-To: References: Message-ID: If you have done the same mistake as I then downgrading must be done via special image supplied by Polycom. That's because the file system format has been changed. Regards, __Yehavi: 2011/11/1 Yehavi Bourvine > Hello, > > I've upgraded our Polycom phones from firmware 3.3.2 to 4.0.0 and have > the following oddities: > > - On shared lines the call lists are "mixed": > - No calls are in outgoing calls. > - answered incoming and outgoing calls are placed in incoming calls. > - unanswered incoming and outgoing calls are placed in the answered > calls. > - When the destination is busy there is no busy sign on the screen - the > call is disconnected immediately > and the phone returns to an idle state. > > I've opened today a support call with Polycom, and will update when I get > replies. > > Regards, __Yehavi: > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/0d48a2b2/attachment.html From brian at freeswitch.org Wed Nov 2 17:47:57 2011 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Nov 2011 09:47:57 -0500 Subject: [Freeswitch-users] Oddities with Polycom's vesion 4.0.0 In-Reply-To: References: Message-ID: <14F1A68B-B231-495E-AEE3-626937DD4D3D@freeswitch.org> Thanks for the heads up. This is usually why I try to avoid .0 releases if possible. /b On Nov 2, 2011, at 9:39 AM, Yehavi Bourvine wrote: > If you have done the same mistake as I then downgrading must be done via > special image supplied by Polycom. That's because the file system format > has been changed. > > Regards, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/5f521950/attachment.html From anthony.minessale at gmail.com Wed Nov 2 17:54:52 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Nov 2011 09:54:52 -0500 Subject: [Freeswitch-users] Call quality In-Reply-To: <-4241905887276648699@unknownmsgid> References: <-4241905887276648699@unknownmsgid> Message-ID: Always start by updating all of them to the very latest GIT. Then systematically eliminate things until you zero in on the problem. On Tue, Nov 1, 2011 at 1:39 PM, Hynek Cihlar wrote: > Would anybody know, if the call quality could be dependant on the > number of media switches the call is routed through? > > Currently I'm testing a system with several freeswitches in the call > way (up to 4) and I'm noticing very short gaps, not very disturbing > but noticable. > > What's your experience? > > Sent from my mobile device > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/d80e7dcf/attachment.html From kiruthika.bkite at gmail.com Wed Nov 2 07:35:11 2011 From: kiruthika.bkite at gmail.com (kiruthika sri) Date: Wed, 2 Nov 2011 10:05:11 +0530 Subject: [Freeswitch-users] FreeTDM PRI tapping issue Message-ID: Hi, I am trying to use the ftmod_pritap module for passive call recording. I did the followings. * Installed wanpipe 3.5.23 for sangoma card AFT A102. * Installed tap 1.4 (customized version of librpi for passive tapping) * Installed freeswitch and freetdm with --with-pritap option * Configured wanpipe, freetdm and freeswitch. Facing the following problem : * FreeSWITCH is getting killed * Getting SIGSEGV # wanrouter messages freeswitch[19720]: segfault at 8 ip 00ae00d1 sp b6ead208 error 4 in libpri.so.1.4[ac5000+4c000] # gdb freeswitch core Program terminated with signal 11, Segmentation fault. #0 0x00ae00d1 in q931_call_getcrv (ctrl=0x99fd9c0, call=0x0, callmode=0xb6ead24c) at q931.c:5335 5335 *callmode = call->cr & 0x7; Kindly do the help on this. Thanks in advance. Regards Kiruthika.U -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/6ea07474/attachment.html From msc at freeswitch.org Wed Nov 2 18:21:45 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Nov 2011 08:21:45 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today - New Dialplan Feature, SIP 101 Message-ID: Hello all! In our conference call today we are going to discuss a cool new dialplan feature (better logical OR processing) and then Ken Rice and I will be discussing some basic SIP knowledge to help get everyone up to speed when helping newbies. Talk to you all soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/3600de93/attachment.html From moises.silva at gmail.com Wed Nov 2 18:31:15 2011 From: moises.silva at gmail.com (Moises Silva) Date: Wed, 2 Nov 2011 11:31:15 -0400 Subject: [Freeswitch-users] FreeTDM PRI tapping issue In-Reply-To: References: Message-ID: getting segfault right away? after a few calls? days? On Wed, Nov 2, 2011 at 12:35 AM, kiruthika sri wrote: > > Hi, > > I am trying to use the ftmod_pritap module for passive call recording. > > I did the followings. > * Installed wanpipe 3.5.23 for sangoma card AFT A102. > * Installed tap 1.4 (customized version of librpi for passive tapping) > * Installed freeswitch and freetdm with --with-pritap option > * Configured wanpipe, freetdm and freeswitch. > > Facing the following problem : > * FreeSWITCH is getting killed > * Getting SIGSEGV > > # wanrouter messages > > freeswitch[19720]: segfault at 8 ip 00ae00d1 sp b6ead208 error 4 in > libpri.so.1.4[ac5000+4c000] > > # gdb freeswitch core > > Program terminated with signal 11, Segmentation fault. > #0 0x00ae00d1 in q931_call_getcrv (ctrl=0x99fd9c0, call=0x0, > callmode=0xb6ead24c) at q931.c:5335 > 5335 *callmode = call->cr & 0x7; > > Kindly do the help on this. > > Thanks in advance. > > Regards > Kiruthika.U > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/f0043c32/attachment.html From eagle.antonio at gmail.com Wed Nov 2 18:48:10 2011 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Wed, 2 Nov 2011 15:48:10 +0000 Subject: [Freeswitch-users] LCR and Continue On Fail In-Reply-To: References: Message-ID: So no one has this problem ?!?! Or continue_on_fail Is not compatible with ${lcr_auto_route} ?? Regards A/T 2011/10/26 Antonio Teixeira > Hello Guys. > > Logs & Dialplan > http://pastebin.freeswitch.org/17613 > > I'm currently trying to connect LCR with the Fall back provided with > Continue On Fail. > I could be misunderstading the use of continue on fail i read several > posts on the mailing and in theory this should work :) > > Anyway my objective is that > If failure _cause == UNALLOCATED_NUMBER > continue using the lcr reach > else: > hangup the call. > > So this is me or could be a bug ?? > > Regards & Thanks for the help :) > Antonio Teixeira > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/45d9182a/attachment-0001.html From helmut.kuper at ewetel.de Wed Nov 2 19:48:28 2011 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 02 Nov 2011 17:48:28 +0100 Subject: [Freeswitch-users] ekiga, celt codec, Freeswitch and audio problems Message-ID: <4EB1745C.8010004@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to setup a phone call using celt codec from ekiga to freeswitch to ekiga, where fs doesn't bypass media. ekiga: 3.2.7 (codec list is CELT at 48000, CELT at 32000, G722, PCMA) Freeswitch: git-a14b20a 2011-09-24 09-58-04 -0500 celt: 0.7.1 (Unfortunately this is what ekiga is using) When FS bypasses the media everything works fine. When ekiga is calling into FS and FS plays a mp3 file to it, it sounds fantastic on ekiga side. When FS doesn't bypasses media the spoken words are distorted. When I use G722 with ekiga and FS in between, there is clear audio. I played around with rtp timer and rewrite_timestamps, but no success. Is it because of celt 0.7.1? Any ideas where to look further? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk6xdFwACgkQ4tZeNddg3dwVfQCdGIX+BSksJp9gfW5C5QWqrS/i zBAAoLm5i8n7wGaEkIFkOKolUPgM3zYE =8WAc -----END PGP SIGNATURE----- From brad at tech21.com Wed Nov 2 19:54:51 2011 From: brad at tech21.com (Brad Mina) Date: Wed, 2 Nov 2011 09:54:51 -0700 Subject: [Freeswitch-users] Public IP in Contact header In-Reply-To: References: Message-ID: RTP-IP 178.238.X.Y Ext-RTP-IP 178.238.X.Y SIP-IP 178.238.X.Y Ext-SIP-IP 178.238.X.Y Let's take a look at what 'ext' might be! Keep in mind you'll need to change it on your external profile as well. On Wed, Nov 2, 2011 at 5:39 AM, Hynek Cihlar wrote: > Can anybody please point me to settings or a freeswitch state that makes > sofia put public IP into the Contact header? Currently I have SIP clients > behind NAT registering to freeswitch. Freeswitch registers these clients > with their respective private IPs. Bellow is the profile status. > > > > ================================================================================================= > Name internal > Domain Name N/A > Auto-NAT false > DBName sofia_reg_internal > Pres Hosts 178.238.X.Y,178.238.X.Y > Dialplan XML > Context default > Challenge Realm auto_from > RTP-IP 178.238.X.Y > Ext-RTP-IP 178.238.X.Y > SIP-IP 178.238.X.Y > Ext-SIP-IP 178.238.X.Y > URL sip:mod_sofia at 178.238.X.Y:5060 > BIND-URL sip:mod_sofia at 178.238.X.Y:5060;maddr=178.238.X.Y > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS IN PCMA,PCMU,GSM,G7221 at 32000h,G7221 at 16000h,G722 > CODECS OUT PCMA,PCMU,GSM,G7221 at 32000h,G7221 at 16000h,G722 > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 0 > FAILED-CALLS-IN 0 > CALLS-OUT 0 > FAILED-CALLS-OUT 0 > > Registrations: > > ================================================================================================= > Call-ID: 3454998XYZ at 192_168_59_102 > User: 1 at ribtjuhd > Contact: "383372XYZ" > Agent: A580 IP/022270000000 > Status: Registered(UDP)(unknown) EXP(2011-11-02 13:30:43) > EXPSECS(100) > Host: eva > IP: 81.90.X.Y > Port: 22055 > Auth-User: 1 > Auth-Realm: ribtjuhd > MWI-Account: 1 at ribtjuhd > > > > > Thanks! > > Hynek > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/940ddba6/attachment.html From brian at freeswitch.org Wed Nov 2 20:16:43 2011 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Nov 2011 12:16:43 -0500 Subject: [Freeswitch-users] Public IP in Contact header In-Reply-To: References: Message-ID: <9899D608-E5EC-4C6D-BC17-1E7C0CA9A527@freeswitch.org> You also have to put in local-network-acl so it knows when to lie and when not to lie in the contact. /b On Nov 2, 2011, at 11:54 AM, Brad Mina wrote: > RTP-IP 178.238.X.Y > Ext-RTP-IP 178.238.X.Y > SIP-IP 178.238.X.Y > Ext-SIP-IP 178.238.X.Y > > Let's take a look at what 'ext' might be! Keep in mind you'll need to > change it on your external profile as well. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/1f8c3678/attachment.html From jack at livecall.com Wed Nov 2 20:17:06 2011 From: jack at livecall.com (Jack) Date: Wed, 02 Nov 2011 10:17:06 -0700 Subject: [Freeswitch-users] mod_rtmp login In-Reply-To: <5e0d82ec10c5ba170c566d22b449a71a.squirrel@emailmg.globat.com> References: <0e6a9dcb6a5cb714cd6a130ec6497142.squirrel@emailmg.globat.com> <4EA77BD1.7080302@livecall.com> <1f1fc7140c13295f010812918da115bf.squirrel@emailmg.globat.com> <4EA86414.3040903@livecall.com> <5e0d82ec10c5ba170c566d22b449a71a.squirrel@emailmg.globat.com> Message-ID: <4EB17B12.1050801@livecall.com> Ricardo, It could be that your hard phone is coming in on port 5080 instead of port 5060 and ending up in your public dialplan instead of your default dialplan. Jack On 11/1/2011 12:43 PM, rsaavedra at ecogizmos.com wrote: > Thank you, > I'm using on the flex client the user 1002 at 1.2.3.4, the user 1000 at 1.2.3.4 > on a sip hard phone (aastra 480). > > My problem in this moment is: the user 1002 can hear but the user 1000 > can't hear. > > Best regards, > > Ricardo Saavedra > > > >> That would be correct if your FreeSwitch server IP address is 1.2.3.4 >> >> >> On 10/26/2011 10:04 AM, rsaavedra at ecogizmos.com wrote: >>> Thank you, >>> >>> so in the flex client I will write the user like 1000 at 1.2.3.4 >>> >>> Best regards, >>> >>> Ricardo >>> >>>> Ricardo, >>>> If your 1000 is default it is probably set up with >>>> >>>> >>>> >>>> if you set the param in rtmp.conf.xml to match the context and then do >>>> a reloadxml or restart freeswitch it may authorize for you. >>>> >>>> >>>> Hope that helps! >>>> Jack >>>> >>>> >>>> >>>> On 10/25/2011 7:16 PM, rsaavedra at ecogizmos.com wrote: >>>>> Hello, >>>>> >>>>> I'm trying to use mod_rtmp but everytime I tried on the console I see >>>>> User >>>>> not authorized. >>>>> I was trying with user:1000 and password:1234. >>>>> >>>>> In the file rtmp.conf.xml I have all by default: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Best regards, >>>>> >>>>> Ricardo Saavedra >>>>> >>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From hynek.cihlar at gmail.com Wed Nov 2 20:42:48 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Wed, 2 Nov 2011 18:42:48 +0100 Subject: [Freeswitch-users] Public IP in Contact header In-Reply-To: <9899D608-E5EC-4C6D-BC17-1E7C0CA9A527@freeswitch.org> References: <9899D608-E5EC-4C6D-BC17-1E7C0CA9A527@freeswitch.org> Message-ID: Brad, Brian, thanks for your replies Actually, the switch is not used in the standard setup, where the external profile is used as a gateway and the internal (default) profile for directly connected sip clients. In my case, the switch has two profiles. The so called external is connected to another switch, just passing calls. The connection is routed. The second profile (internal) which is giving me troubles, is used for registering SIP clients. The SIP clients are usually behind their own NAT routers. SIP-IP and EXT-SIP-IP are both the same public IP, since the router is publicly visible on the internal profile. On the external profile, they are assigned to a private internal IP. The switch is sitting on two subnets. Now, the local-network-acl is set to localnet.auto on both profiles. I was assuming ACLs are used solely for security! Hynek On Wed, Nov 2, 2011 at 6:16 PM, Brian West wrote: > You also have to put in local-network-acl so it knows when to lie and when > not to lie in the contact. > > /b > > On Nov 2, 2011, at 11:54 AM, Brad Mina wrote: > > RTP-IP 178.238.X.Y > Ext-RTP-IP 178.238.X.Y > SIP-IP 178.238.X.Y > Ext-SIP-IP 178.238.X.Y > > Let's take a look at what 'ext' might be! Keep in mind you'll need to > change it on your external profile as well. > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/48f076b7/attachment-0001.html From vipkilla at gmail.com Wed Nov 2 20:58:11 2011 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 2 Nov 2011 13:58:11 -0400 Subject: [Freeswitch-users] ekiga, celt codec, Freeswitch and audio problems In-Reply-To: <4EB1745C.8010004@ewetel.de> References: <4EB1745C.8010004@ewetel.de> Message-ID: It is most likely because of the version of CELT. The bit stream for CELT is not frozen, I'd ask the devs to update Ekiga to use latest CELT. On Wed, Nov 2, 2011 at 12:48 PM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > I tried to setup a phone call using celt codec from ekiga to freeswitch > to ekiga, where fs doesn't bypass media. > > ekiga: 3.2.7 (codec list is CELT at 48000, CELT at 32000, G722, PCMA) > Freeswitch: git-a14b20a 2011-09-24 09-58-04 -0500 > celt: 0.7.1 (Unfortunately this is what ekiga is using) > > When FS bypasses the media everything works fine. > > When ekiga is calling into FS and FS plays a mp3 file to it, it sounds > fantastic on ekiga side. > > When FS doesn't bypasses media the spoken words are distorted. > > When I use G722 with ekiga and FS in between, there is clear audio. > > > I played around with rtp timer and rewrite_timestamps, but no success. > > Is it because of celt 0.7.1? Any ideas where to look further? > > regards > helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.10 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ > > iEYEARECAAYFAk6xdFwACgkQ4tZeNddg3dwVfQCdGIX+BSksJp9gfW5C5QWqrS/i > zBAAoLm5i8n7wGaEkIFkOKolUPgM3zYE > =8WAc > -----END PGP SIGNATURE----- > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From avi at avimarcus.net Wed Nov 2 22:09:22 2011 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 2 Nov 2011 21:09:22 +0200 Subject: [Freeswitch-users] LCR and Continue On Fail In-Reply-To: References: Message-ID: I'd really love an answer to this, too. Seeing 3 user_busy in a row followed by normal_temporary_failure because it tried 4 gateways.. then not even returning a busy signal.. pretty much sucks. -Avi On Wed, Nov 2, 2011 at 5:48 PM, Antonio Teixeira wrote: > So no one has this problem ?!?! > > Or > > continue_on_fail > > Is not compatible with ${lcr_auto_route} ?? > > > Regards > A/T > > > > 2011/10/26 Antonio Teixeira > >> Hello Guys. >> >> Logs & Dialplan >> http://pastebin.freeswitch.org/17613 >> >> I'm currently trying to connect LCR with the Fall back provided with >> Continue On Fail. >> I could be misunderstading the use of continue on fail i read several >> posts on the mailing and in theory this should work :) >> >> Anyway my objective is that >> If failure _cause == UNALLOCATED_NUMBER >> continue using the lcr reach >> else: >> hangup the call. >> >> So this is me or could be a bug ?? >> >> Regards & Thanks for the help :) >> Antonio Teixeira >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/fb6bdb85/attachment.html From nasida at live.ru Wed Nov 2 22:18:53 2011 From: nasida at live.ru (Yuriy Nasida) Date: Wed, 2 Nov 2011 23:18:53 +0400 Subject: [Freeswitch-users] double dtmf In-Reply-To: References: , , , Message-ID: Hello again, So I have made and analyse the two *.cap files. the first file contains call from pstn with dtmf issue. the second file contain call from sip softphone without dtmf issue. As far as I see when I call from softphone and send dtmf, in this moment the media stream is empty (because softphone cut dtmf from media in accordance with rfc2833). But when I call from pstn (from cell phone for example) and send dtmf, in this moment the media stream is not empty. I see the hits which correspond the sending of dtmf. I think that therefore I receive double dtmf on endpoint. How I can to suppress dtmf in inband ? "Please let me know if you have addressed this issue."Sure. Thanks. Date: Wed, 2 Nov 2011 10:44:08 +0530 From: vellayappan.n at mobax.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] double dtmf Hi, I too have the same issue. When calling from PSTN number, the freeswitch says Incoming call from "0000000000" instead of the actual number. Moreover it detects only few digits (say 2 or 3) not all the dtmf numbers. Please let me know if you have addressed this issue. 2011/11/2 Yuriy Nasida Thank you for your answer. "are you doing something explicitly to send the DTMF" No. I just press buttons in my cell phone (I mean startpoint). In my opinion it is strange. I make answer() in lua script and use session:getDigits(). All works fine. In the end of script I use execute_extension which implements bridge.session:execute("execute_extension", "".. recieved_digit .." XML ".. user_context .."") Thus the call was answered in lua script. Next the call go to endpoint. How dtmf must be transmitted in this case ? Why I have not issues when I call from sip devices? Any help are welcomed. Thanks. From: avi at avimarcus.net Date: Tue, 1 Nov 2011 21:22:11 +0200 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] double dtmf "when I try to send dtmf second time for endpoint " - are you doing something explicitly to send the DTMF? I just ignore the dtmf, and when FS bridges it sends the call on properly. -Avi 2011/11/1 Yuriy Nasida Hello community, I have some issue with delivery of dtmf.The call flow is: one sip provider - FS - second sip provider - endpoint which expects correct dtmf On FS I have simple custom lua application which implement logic of calling cards system. I have not issues when I call to lua application. I can enter destination number (by using dtmf apparently) and all works fine. But when I try to send dtmf second time for endpoint which expects correct dtmf I have issue with double dtmf. It is noted that when I call from sip device I have not ANY issues. But I have issue when I call from any cell phone or landline phone. Any ideas are welcomed. 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 -- Regards Vellayappan FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/4cfb57ac/attachment.html From helmut.kuper at ewetel.de Wed Nov 2 22:24:06 2011 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 02 Nov 2011 20:24:06 +0100 Subject: [Freeswitch-users] ekiga, celt codec, Freeswitch and audio problems In-Reply-To: References: <4EB1745C.8010004@ewetel.de> Message-ID: <4EB198D6.4010601@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Vik, Am 02.11.2011 18:58, schrieb Vik Killa: > It is most likely because of the version of CELT. The bit stream > for CELT is not frozen, I'd ask the devs to update Ekiga to use > latest CELT. yes, that's true. But I made sure that both ekiga and FS are using the same CELT version (0.7.1), so I guess it should work. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk6xmNYACgkQ4tZeNddg3dyzhwCeKfqkS95urMFcOAzHhVvhoirM ObgAn3CQL864C9gU7xBkj2l19MORBZQ4 =2ZHt -----END PGP SIGNATURE----- From helmut.kuper at ewetel.de Wed Nov 2 23:12:58 2011 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 02 Nov 2011 21:12:58 +0100 Subject: [Freeswitch-users] no Ringing-Signal on incomming calls via mod_freetdm/sangoma_isdn Message-ID: <4EB1A44A.3000000@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I switched from old openzap PRI-Stack to freetdm with sangoma_isdn because of some odd problems with our EWSD switch. Setup was quite easy. Only one thing is strange. When I call into FS from pstn to a sip phone I get no ringing resp ALERTING signal to my pstn phone. Executing ring_ready in dialplan just befor bridging works. Unfortuantely it shows the same behaviour as using "instant_ringback=true", e.g. First ringing then busy ... I had that problem also using old openzap. So today I looked around in FS code and found that when mod_sofia sends a SWITCH_MESSAGE_INDICATE_RINGING on receiving a sip 180 reply back to the caller's channel it works. Is there any reason why sofia doesn't do this by default? Oh, I use FS git-1d3417a 2011-06-07 17-35-49 -0400. best regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk6xpEoACgkQ4tZeNddg3dxGjwCePxDkj4fVdprToKPov5+OeM/3 YakAnitvJZod29hSW4J9T/Sn2Vuob0rU =4KAd -----END PGP SIGNATURE----- From nasida at live.ru Thu Nov 3 00:02:28 2011 From: nasida at live.ru (Yuriy Nasida) Date: Thu, 3 Nov 2011 01:02:28 +0400 Subject: [Freeswitch-users] No sound when 2 client in the same NAT In-Reply-To: References: Message-ID: As far as I see it must work. Try to make "cap" file by means of tcpdump on FS and see the media stream by means wireshark. Date: Tue, 1 Nov 2011 23:17:46 +0800 From: xyangni at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] No sound when 2 client in the same NAT Hi, I run FS on a VPS with public IP, and 2 x-lite client on 2 of my home pc which are in the same NAT. They works fine if each of them calls FS sample ivr or bridged to other external phone. But when they call each other, the call is completely silent. The sip trace is listed below. Can anyone please help. Thanks. recv 767 bytes from udp/[90.192.85.6]:6216 at 15:08:19.500457: ------------------------------------------------------------------------ INVITE sip:1015 at 178.79.188.153 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport Max-Forwards: 70 Contact: To: "1015" From: "1017";tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 235 v=0 o=- 8 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 43926 RTP/AVP 0 8 3 101 a=alt:1 1 : tAU3AHbD K56reERa 90.192.85.6 43928 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------send 379 bytes to udp/[90.192.85.6]:6216 at 15:08:19.501004: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Content-Length: 0 ------------------------------------------------------------------------send 869 bytes to udp/[90.192.85.6]:6216 at 15:08:19.502160: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" ;tag=K2eK1a4UZ0vvp Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="178.79.188.153", nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------recv 326 bytes from udp/[90.192.85.6]:6216 at 15:08:19.541053: ------------------------------------------------------------------------ ACK sip:1015 at 178.79.188.153 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-e329ce130338d93d-1---d8754z-;rport To: "1015" ;tag=K2eK1a4UZ0vvp From: "1017";tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 1 ACK Content-Length: 0 ------------------------------------------------------------------------recv 1031 bytes from udp/[90.192.85.6]:6216 at 15:08:19.552533: ------------------------------------------------------------------------ INVITE sip:1015 at 178.79.188.153 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport Max-Forwards: 70 Contact: To: "1015" From: "1017";tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="1017",realm="178.79.188.153",nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0",uri="sip:1015 at 178.79.188.153",response="603f04a293a7535b109d9fae13085db8",cnonce="fc27cb770d3b4701e15f7aadc4690a64",nc=00000001,qop=auth,algorithm=MD5 User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 235 v=0 o=- 8 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 43926 RTP/AVP 0 8 3 101 a=alt:1 1 : tAU3AHbD K56reERa 90.192.85.6 43928 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------send 379 bytes to udp/[90.192.85.6]:6216 at 15:08:19.552996: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Content-Length: 0 ------------------------------------------------------------------------2011-11-01 15:08:19.545486 [NOTICE] switch_channel.c:915 New Channel sofia/internal/1017 at 178.79.188.153 [d46f4a05-6572-48e9-bea9-828e70addb74] 2011-11-01 15:08:19.545486 [INFO] mod_dialplan_xml.c:336 Processing 1017 <1017>->1015 in context default2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *1 execute_extension::dx XML features 2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1017.2011-11-01-15-08-19.wav2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *3 execute_extension::cf XML features 2011-11-01 15:08:19.545486 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *4 execute_extension::att_xfer XML features2011-11-01 15:08:19.545486 [NOTICE] switch_channel.c:915 New Channel sofia/internal/sip:1015 at 90.192.85.6:63216 [b6fc17cc-1585-4e43-803d-f431c90fc196] send 1309 bytes to udp/[90.192.85.6]:63216 at 15:08:19.565528: ------------------------------------------------------------------------ INVITE sip:1015 at 90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKFyaDU717BB1XS Max-Forwards: 69 From: "Extension 1017" ;tag=Nm144052Sj91D To: Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 317 X-FS-Support: update_display Remote-Party-ID: "Extension 1017" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1320132861 1320132862 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27238 RTP/AVP 0 98 99 9 8 3 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------recv 438 bytes from udp/[90.192.85.6]:63216 at 15:08:19.707081: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: To: ;tag=054e4538 From: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------2011-11-01 15:08:19.705449 [NOTICE] sofia.c:5375 Ring-Ready sofia/internal/sip:1015 at 90.192.85.6:63216! 2011-11-01 15:08:19.705449 [INFO] switch_ivr_originate.c:1115 Sending early media2011-11-01 15:08:19.705449 [NOTICE] mod_sofia.c:2475 Pre-Answer sofia/internal/1017 at 178.79.188.153! send 1188 bytes to udp/[90.192.85.6]:6216 at 15:08:19.713171: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" ;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 251 Remote-Party-ID: "1015" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1320132951 1320132952 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27148 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 ------------------------------------------------------------------------ recv 733 bytes from udp/[90.192.85.6]:63216 at 15:08:23.699243: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: To: ;tag=054e4538 From: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 1 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 44478 RTP/AVP 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 428 bytes to udp/[90.192.85.6]:63216 at 15:08:23.700969: ------------------------------------------------------------------------ ACK sip:1015 at 90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKg735v2jB9KQgN Max-Forwards: 70 From: "Extension 1017" ;tag=Nm144052Sj91D To: ;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------2011-11-01 15:08:23.685455 [NOTICE] sofia.c:5983 Channel [sofia/internal/sip:1015 at 90.192.85.6:63216] has been answered send 1158 bytes to udp/[90.192.85.6]:6216 at 15:08:23.706877: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" ;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 251 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1320132951 1320132952 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27148 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 ------------------------------------------------------------------------ 2011-11-01 15:08:23.705453 [NOTICE] switch_ivr_originate.c:3194 Channel [sofia/internal/1017 at 178.79.188.153] has been answeredsend 1158 bytes to udp/[90.192.85.6]:6216 at 15:08:24.207213: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-cf0ad11b8800c42c-1---d8754z-;rport=6216 From: "1017";tag=ab30a26d To: "1015" ;tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 251 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1320132951 1320132952 IN IP4 178.79.188.153 s=FreeSWITCH c=IN IP4 178.79.188.153 t=0 0 m=audio 27148 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 ------------------------------------------------------------------------ recv 733 bytes from udp/[90.192.85.6]:63216 at 15:08:24.380014: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: To: ;tag=054e4538 From: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 1 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 44478 RTP/AVP 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 428 bytes to udp/[90.192.85.6]:63216 at 15:08:24.380297: ------------------------------------------------------------------------ ACK sip:1015 at 90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKg735v2jB9KQgN Max-Forwards: 70 From: "Extension 1017" ;tag=Nm144052Sj91D To: ;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------recv 733 bytes from udp/[90.192.85.6]:63216 at 15:08:25.063077: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKFyaDU717BB1XS Contact: To: ;tag=054e4538 From: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 1 2 IN IP4 90.192.85.6 s=CounterPath eyeBeam 1.5 c=IN IP4 90.192.85.6 t=0 0 m=audio 44478 RTP/AVP 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------send 428 bytes to udp/[90.192.85.6]:63216 at 15:08:25.063384: ------------------------------------------------------------------------ ACK sip:1015 at 90.192.85.6:63216;rinstance=4c1fe319ab93f311 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKg735v2jB9KQgN Max-Forwards: 70 From: "Extension 1017" ;tag=Nm144052Sj91D To: ;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 19743985 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------recv 711 bytes from udp/[90.192.85.6]:6216 at 15:08:25.157427: ------------------------------------------------------------------------ ACK sip:1015 at 178.79.188.153:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-52077a3eda00e179-1---d8754z-;rport Max-Forwards: 70 Contact: To: "1015";tag=mB8B35mZv9jFj From: "1017";tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 ACK Proxy-Authorization: Digest username="1017",realm="178.79.188.153",nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0",uri="sip:1015 at 178.79.188.153",response="603f04a293a7535b109d9fae13085db8",cnonce="fc27cb770d3b4701e15f7aadc4690a64",nc=00000001,qop=auth,algorithm=MD5 User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------recv 711 bytes from udp/[90.192.85.6]:6216 at 15:08:25.585421: ------------------------------------------------------------------------ ACK sip:1015 at 178.79.188.153:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:6216;branch=z9hG4bK-d8754z-52077a3eda00e179-1---d8754z-;rport Max-Forwards: 70 Contact: To: "1015";tag=mB8B35mZv9jFj From: "1017";tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 2 ACK Proxy-Authorization: Digest username="1017",realm="178.79.188.153",nonce="67fa1677-e806-4c64-8337-5b1cd866e3c0",uri="sip:1015 at 178.79.188.153",response="603f04a293a7535b109d9fae13085db8",cnonce="fc27cb770d3b4701e15f7aadc4690a64",nc=00000001,qop=auth,algorithm=MD5 User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------recv 533 bytes from udp/[90.192.85.6]:63216 at 15:09:24.414921: ------------------------------------------------------------------------ BYE sip:mod_sofia at 178.79.188.153:5060 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:63216;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Extension 1017";tag=Nm144052Sj91D From: ;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: eyeBeam release 1102q stamp 51814 Reason: SIP;description="User Hung Up" Content-Length: 0 ------------------------------------------------------------------------2011-11-01 15:09:24.425460 [NOTICE] sofia.c:573 Hangup sofia/internal/sip:1015 at 90.192.85.6:63216 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] send 576 bytes to udp/[90.192.85.6]:63216 at 15:09:24.426280: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:63216;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport=63216 From: ;tag=054e4538 To: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------2011-11-01 15:09:24.445471 [NOTICE] switch_ivr_bridge.c:1355 Hangup sofia/internal/1017 at 178.79.188.153 [CS_EXECUTE] [NORMAL_CLEARING] 2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1395 Session 2 (sofia/internal/sip:1015 at 90.192.85.6:63216) Ended2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/sip:1015 at 90.192.85.6:63216 [CS_DESTROY] send 623 bytes to udp/[90.192.85.6]:6216 at 15:09:24.448813: ------------------------------------------------------------------------ BYE sip:1017 at 90.192.85.6:6216 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKHgXyyX3e6vD3g Max-Forwards: 70 From: "1015" ;tag=mB8B35mZv9jFj To: "1017" ;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1395 Session 1 (sofia/internal/1017 at 178.79.188.153) Ended 2011-11-01 15:09:24.445471 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/1017 at 178.79.188.153 [CS_DESTROY]recv 533 bytes from udp/[90.192.85.6]:63216 at 15:09:25.132183: ------------------------------------------------------------------------ BYE sip:mod_sofia at 178.79.188.153:5060 SIP/2.0 Via: SIP/2.0/UDP 90.192.85.6:63216;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Extension 1017";tag=Nm144052Sj91D From: ;tag=054e4538 Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: eyeBeam release 1102q stamp 51814 Reason: SIP;description="User Hung Up" Content-Length: 0 ------------------------------------------------------------------------send 576 bytes to udp/[90.192.85.6]:63216 at 15:09:25.132451: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 90.192.85.6:63216;branch=z9hG4bK-d8754z-0b44212243020a35-1---d8754z-;rport=63216 From: ;tag=054e4538 To: "Extension 1017";tag=Nm144052Sj91D Call-ID: 2caf7664-7f3e-122f-0a83-f23c91df8b99 CSeq: 2 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------send 623 bytes to udp/[90.192.85.6]:6216 at 15:09:25.449077: ------------------------------------------------------------------------ BYE sip:1017 at 90.192.85.6:6216 SIP/2.0 Via: SIP/2.0/UDP 178.79.188.153;rport;branch=z9hG4bKHgXyyX3e6vD3g Max-Forwards: 70 From: "1015" ;tag=mB8B35mZv9jFj To: "1017" ;tag=ab30a26d Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4349ec0 2011-10-31 14-38-41 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------recv 376 bytes from udp/[90.192.85.6]:6216 at 15:09:25.483104: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKHgXyyX3e6vD3g Contact: To: "1017";tag=ab30a26d From: "1015";tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------recv 376 bytes from udp/[90.192.85.6]:6216 at 15:09:26.386027: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 178.79.188.153;rport=5060;branch=z9hG4bKHgXyyX3e6vD3g Contact: To: "1017";tag=ab30a26d From: "1015";tag=mB8B35mZv9jFj Call-ID: ZGRhMmQ1NDBiNDhkMzhjMGFlOTA4ZDgwMTEyNTBlYzM. CSeq: 19744018 BYE User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/ec2fe2d1/attachment-0001.html From cjbujold at accra.ca Thu Nov 3 00:03:45 2011 From: cjbujold at accra.ca (Charles Bujold) Date: Wed, 2 Nov 2011 18:03:45 -0300 Subject: [Freeswitch-users] Problem with outbound call to second office Message-ID: <010701cc99a2$e91e98f0$bb5bcad0$@accra.ca> Strange situation. Have 2 offices each with phones connected to the Freeswitch server in Office 1.. Office 1 Freeswitch Server PFsense Internet Office 2 Telephone 2 Telephone 1 ? 192.168.0.15 ? 192.162.0.1 ? 156.233.20.37 ? Telco Router ? 192.168.1.123 192.168.0.10 152.43.37.129 If the telephone in Office 2, call the telephone in office 1 everything works perfectly. If Office 1 telephone, calls Office 2 Telephone, Telephone 2 rings they can pick up while Freeswitch still gives to Telephone 1 a ring tone even after they pick up and eventually send telephone 1 to voicemail. I placed wireshark on the Freeswitch Server (Ubuntu 10.4 ) and I see the following: Call Originates from 192.168.0.10 to 192.168.0.15 with a SIP packet to 152.43.37.129 I can see that the Call RTP/UDP is to be port 25542 received from Telephone 2 I see that Freeswitch sends a packet to 192.168.0.10 (Telephone 1) I see that Telephone 2 at 152.43.37.129 is asking to talk on port 25542 And then I see that the Freeswitch server send an ICMP packet to 152.43.37.129 stating that the port 25542 is unreachable. Obviously they are talking, just not communicating! The PFsense firewall has all ports open and UDP enable. I even placed a rule that all traffic from 152.43.37.129 be forwarded to the Freeswitch server, and still no voice can be heard. What I don?t get, is where does the problem exist, since Office 2 can call us and everything works. This suggest that there is no firewall issue SIP and RTP/UDP is working. Yet when we call, we see that the port has been negotiated which means the SIP port is working and then we get the error that the port is unreachable. My question: - Is the problem in the Ubuntu server ( no firewall) or is it elsewhere? They seem to get in without a problem, why can?t we get out? Just to clarify Telephone 1 can make any calls to any other Telephone number (local and long distance) and everything works perfectly. Is the problem in the Telco router? Unable to put a tracer on the Telco router but inside Office 2 wireshark states that it is not receiving anything from Office 1 which corroborates what we see in the Freeswitch server wireshark trace. When I place a wireshark trace at the PFsense server I see the exact same thing as what I see on the Freeswitch server wireshark trace. I can see the Port is unreachable error go to the Office 2 telephone and in Office 2 I see receiving the ICMP message that the port is unreachable. My second question: - Why would the port be unreachable and is it on the Ubuntu server? PFsense server? or Telco router? Third question: - Is there a place in Freeswitch I can check to verify that the RTP ports are in the appropriate range? Any suggestion on how to troubleshoot or resolve this issue would be greatly appreciated. CJB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/14ab1665/attachment.html From tyler at phone.com Thu Nov 3 01:03:41 2011 From: tyler at phone.com (Tyler Winter) Date: Wed, 2 Nov 2011 15:03:41 -0700 Subject: [Freeswitch-users] Problem 'adding' video to an audio initiated call, fmtp In-Reply-To: <539AC2B9-E137-4CAF-BFDC-DED583F66D7F@phone.com> References: <539AC2B9-E137-4CAF-BFDC-DED583F66D7F@phone.com> Message-ID: <5544517E-9301-404C-A55E-31B49719E14B@phone.com> I was able to add video to a non-video initiated call by using the sip_renegotiate_codec_on_reinvite setting which ended up being elusive. My understanding of this setting from reviewing the code, is: - the default behavior of FS is that once the codec is negotiated, re-INVITEs skip the codec negotiation logic and only process changes in media IP/port/etc. - by turning this setting on, every re-INVITE goes through the same codec negotiation logic as happens during the initial INVITE. Hope this helps someone else too. -Tyler On Sep 27, 2011, at 2:58 PM, Tyler Winter wrote: > Hi, > > I'm trying to add video to a non-video initiated call. I should mention that I have no problems initiated and answering a call as video (using Bria softphone and H.264 codec). I captured several test calls to compare the SDP. I found that adding video doesn't work only when initiating the call as audio. In other words, I can initiate the call as video on the A-leg, and answer the call as audio on the B-leg and then add video on the B-leg and video starts working. I think problem is due to there being no fmtp "media attribute" on the 200 OK w/SDP response from FS to the A-leg when I initialize the call as audio, then add video to it. > > I believe I narrowed down the problem to be the following: > - Caller (a-leg) initiated the call as audio. > - Callee (b-leg) answered the call as video > - Caller (a-leg) "adds" video which sends an INVITE to the switch to negotiate the video codecs > - FS responds to Caller (a-leg) with 200 OK (SDP) but the "fmtp" media attribute is missing. > > On a working video call the fmtp attribute is appended to the SDP along with the rest of the video codec details. > > Here's an example: > > this is the INVITEs SDP information to FS from the caller (a-leg) on a working video call (initiated as video): > Media Description, name and address (m): video 55752 RTP/AVP 123 124 > Bandwidth Information (b): TIAS:1574400 > Media Attribute (a): rtpmap:123 H264/90000 > Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 > Media Attribute (a): rtpmap:124 H264/90000 > Media Attribute (a): fmtp:124 profile-level-id=42801f;packetization-mode=1;level-asymmetry-allowed=1 > Media Attribute (a): sendrecv > > 200 OK from FS to the Caller (a-leg): > Media Description, name and address (m): video 16454 RTP/AVP 123 > Media Attribute (a): rtpmap:123 H264/90000 > Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 > > Now, when I try to initiate a call as audio, then add video: > > INVITE from the caller (a-leg) to FS ("adding" video during call): > Media Description, name and address (m): video 64698 RTP/AVP 123 124 > Bandwidth Information (b): TIAS:1574400 > Media Attribute (a): rtpmap:123 H264/90000 > Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 > Media Attribute (a): rtpmap:124 H264/90000 > Media Attribute (a): fmtp:124 profile-level-id=42801f;packetization-mode=1;level-asymmetry-allowed=1 > Media Attribute (a): sendrecv > > 200 OK from FS to the caller (a-leg): > Media Description, name and address (m): video 23678 RTP/AVP 123 > Media Attribute (a): rtpmap:123 H264/90000 > > So, it appears that the "fmtp" attribute is missing on the 200 OK response: > Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 > > I can upload the captures somewhere if the full details will be helpful to anyone. If anyone has had any luck with this feature, I'm interested in how you accomplished it. I also tried using the setting "sip_force_video_fmtp" to no avail. > > I appreciate any help. > > Regards, > > Tyler > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/b28e3ad6/attachment.html From rsaavedra at ecogizmos.com Thu Nov 3 01:49:40 2011 From: rsaavedra at ecogizmos.com (rsaavedra at ecogizmos.com) Date: Wed, 2 Nov 2011 17:49:40 -0500 Subject: [Freeswitch-users] mod_rtmp login In-Reply-To: <4EB17B12.1050801@livecall.com> References: <0e6a9dcb6a5cb714cd6a130ec6497142.squirrel@emailmg.globat.com> <4EA77BD1.7080302@livecall.com> <1f1fc7140c13295f010812918da115bf.squirrel@emailmg.globat.com> <4EA86414.3040903@livecall.com> <5e0d82ec10c5ba170c566d22b449a71a.squirrel@emailmg.globat.com> <4EB17B12.1050801@livecall.com> Message-ID: <07ff3a6f73d69de3d1cb558ac5acacd5.squirrel@emailmg.globat.com> Thank you, now everything is working. > Ricardo, > It could be that your hard phone is coming in on port 5080 instead of > port 5060 and ending up in your public dialplan instead of your default > dialplan. > Jack > > On 11/1/2011 12:43 PM, rsaavedra at ecogizmos.com wrote: >> Thank you, >> I'm using on the flex client the user 1002 at 1.2.3.4, the user >> 1000 at 1.2.3.4 >> on a sip hard phone (aastra 480). >> >> My problem in this moment is: the user 1002 can hear but the user 1000 >> can't hear. >> >> Best regards, >> >> Ricardo Saavedra >> >> >> >>> That would be correct if your FreeSwitch server IP address is 1.2.3.4 >>> >>> >>> On 10/26/2011 10:04 AM, rsaavedra at ecogizmos.com wrote: >>>> Thank you, >>>> >>>> so in the flex client I will write the user like 1000 at 1.2.3.4 >>>> >>>> Best regards, >>>> >>>> Ricardo >>>> >>>>> Ricardo, >>>>> If your 1000 is default it is probably set up with >>>>> >>>>> >>>>> >>>>> if you set the param in rtmp.conf.xml to match the context and then >>>>> do >>>>> a reloadxml or restart freeswitch it may authorize for you. >>>>> >>>>> >>>>> Hope that helps! >>>>> Jack >>>>> >>>>> >>>>> >>>>> On 10/25/2011 7:16 PM, rsaavedra at ecogizmos.com wrote: >>>>>> Hello, >>>>>> >>>>>> I'm trying to use mod_rtmp but everytime I tried on the console I >>>>>> see >>>>>> User >>>>>> not authorized. >>>>>> I was trying with user:1000 and password:1234. >>>>>> >>>>>> In the file rtmp.conf.xml I have all by default: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Ricardo Saavedra >>>>>> >>>>>> >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lloyd.aloysius at gmail.com Thu Nov 3 02:22:36 2011 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Wed, 2 Nov 2011 19:22:36 -0400 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: DynDns not supporting NAPTR. I received reply from them "We do not support NAPTR records."I am currently using DynDns for my DNS needs. Any other recommendation to make Polycom phone work with DNS SRV records without NAPTR records. Thanks and regards, Lloyd On Tue, Nov 1, 2011 at 4:57 PM, Lloyd Aloysius wrote: > Vik, > > I have the same setting in my configuration files. But the phone never > register. I will follow the Brian instructions [create NAPTR Records] > shortly. > > Thanks > Lloyd > > > On Tue, Nov 1, 2011 at 10:46 AM, Vik Killa wrote: > >> I have Polycoms working with DNS SRV without NAPTR records >> It was a matter of polycom provisioning files. >> I attached the 2 important files and that last important file named >> polycom_0004f21a0e87.cfg looks like this: >> >> >> > reg.1.displayName="Test UA" >> reg.1.address="1001" >> reg.1.label="Test UA" >> reg.1.type="private" >> reg.1.auth.userId="1001" >> reg.1.auth.password="PASSWORD" >> reg.1.server.1.address="mypbx.mydomain.com" >> /> >> >> >> >> On Tue, Nov 1, 2011 at 10:13 AM, Lloyd Aloysius >> wrote: >> > Thank you for the reply . >> > I miss the NAPTR records. Let me setup the NAPTR records and test again. >> > Thanks and regards, >> > >> > Lloyd >> > >> > >> > >> > On Tue, Nov 1, 2011 at 9:27 AM, Brian West >> wrote: >> >> >> >> Ok you missed the part about the NAPTR records didn't you? Make sure >> you >> >> setup your NAPTR records correctly. I really don't get how people >> think >> >> they are optional. >> >> I know this says >> >> optional >> http://wiki.freeswitch.org/wiki/SIP_TLS#Step_5_-_DNS_NAPTR_.26_SRV_Records_.28optional.29 >> >> but they aren't. >> >> /b >> >> On Nov 1, 2011, at 8:22 AM, Lloyd Aloysius wrote: >> >> >> >> But this fails to register. >> >> >> >> Thanks >> >> Lloyd >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111102/ac302f41/attachment.html From kiruthika.bkite at gmail.com Thu Nov 3 07:41:44 2011 From: kiruthika.bkite at gmail.com (kiruthika sri) Date: Thu, 3 Nov 2011 10:11:44 +0530 Subject: [Freeswitch-users] FreeTDM PRI tapping issue In-Reply-To: References: Message-ID: Getting segfault after getting RINGING event in the first call itself. On Wed, Nov 2, 2011 at 9:01 PM, Moises Silva wrote: > getting segfault right away? after a few calls? days? > > On Wed, Nov 2, 2011 at 12:35 AM, kiruthika sri wrote: > >> >> Hi, >> >> I am trying to use the ftmod_pritap module for passive call recording. >> >> I did the followings. >> * Installed wanpipe 3.5.23 for sangoma card AFT A102. >> * Installed tap 1.4 (customized version of librpi for passive tapping) >> * Installed freeswitch and freetdm with --with-pritap option >> * Configured wanpipe, freetdm and freeswitch. >> >> Facing the following problem : >> * FreeSWITCH is getting killed >> * Getting SIGSEGV >> >> # wanrouter messages >> >> freeswitch[19720]: segfault at 8 ip 00ae00d1 sp b6ead208 error 4 in >> libpri.so.1.4[ac5000+4c000] >> >> # gdb freeswitch core >> >> Program terminated with signal 11, Segmentation fault. >> #0 0x00ae00d1 in q931_call_getcrv (ctrl=0x99fd9c0, call=0x0, >> callmode=0xb6ead24c) at q931.c:5335 >> 5335 *callmode = call->cr & 0x7; >> >> Kindly do the help on this. >> >> Thanks in advance. >> >> Regards >> Kiruthika.U >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/9a3d8b52/attachment.html From hynek.cihlar at gmail.com Thu Nov 3 09:44:19 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Thu, 3 Nov 2011 07:44:19 +0100 Subject: [Freeswitch-users] SOLVED Public IP in Contact header Message-ID: I played with the ACLs but it didn't have any effect on the contact header. So I went into the code and after some investigation I stepped on the * sip-force-contact* variable. When this guy is set to * NDLB-connectile-dysfunction*, freeswitch overwrites the contact header with the IP/PORT it received the IP packet on, regardless of the actual nat state. Hynek On Wed, Nov 2, 2011 at 6:42 PM, Hynek Cihlar wrote: > Brad, Brian, thanks for your replies > > Actually, the switch is not used in the standard setup, where the external > profile is used as a gateway and the internal (default) profile for > directly connected sip clients. > > In my case, the switch has two profiles. The so called external is > connected to another switch, just passing calls. The connection is routed. > The second profile (internal) which is giving me troubles, is used for > registering SIP clients. The SIP clients are usually behind their own NAT > routers. > > SIP-IP and EXT-SIP-IP are both the same public IP, since the router is > publicly visible on the internal profile. On the external profile, they are > assigned to a private internal IP. The switch is sitting on two subnets. > > Now, the local-network-acl is set to localnet.auto on both profiles. I was > assuming ACLs are used solely for security! > > Hynek > > > > On Wed, Nov 2, 2011 at 6:16 PM, Brian West wrote: > >> You also have to put in local-network-acl so it knows when to lie and >> when not to lie in the contact. >> >> /b >> >> On Nov 2, 2011, at 11:54 AM, Brad Mina wrote: >> >> RTP-IP 178.238.X.Y >> Ext-RTP-IP 178.238.X.Y >> SIP-IP 178.238.X.Y >> Ext-SIP-IP 178.238.X.Y >> >> Let's take a look at what 'ext' might be! Keep in mind you'll need to >> change it on your external profile as well. >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/be20b45f/attachment.html From spencer at 5ninesolutions.com Thu Nov 3 12:11:40 2011 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 3 Nov 2011 02:11:40 -0700 Subject: [Freeswitch-users] Timezone with time conditions Message-ID: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com> Hello all, Has anyone found an elegant solution to using time conditions with multiple locations? For example suppose you have a multi tenant system with time conditions, is there any way to set the call time zone on a per call basis? Thanks, Spencer From roland at haenel.me Thu Nov 3 12:22:07 2011 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Thu, 3 Nov 2011 10:22:07 +0100 Subject: [Freeswitch-users] Audio clipping in mod_conference Message-ID: Hi, During thorough testing of our conference solution which is based on FreeSwitch / mod_conference, we noticed some 'audio clipping' during the conference. Typically, if a conference member didn't have audio for some time and then begins to speak, the first syllable will be clipped and cannot be heard by the other conference participants. We ruled out a number of possible causes, and finally discovered that the energy level detection algorith seems to be the source of the problem. Setting the energy level for all participants to "0" heals the problem (of course now the noise might be higher since every participant's noise contributes to the total noise), but this setting is not very useful since the "talking detection" seems to rely on the same level, i.e. with energy "0", all members are marked talking all the time (which effectively renders the "talking" display useless). Is there any chance to tune the energy level detection time (!), or to de-couple talking detection an energy level detection in FreeSwitch? Best regards, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/9342b123/attachment-0001.html From hynek.cihlar at gmail.com Thu Nov 3 13:22:50 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Thu, 3 Nov 2011 11:22:50 +0100 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: Is it only conference related or may it affect other apps? Hynek 2011/11/3 Roland H?nel > Hi, > > During thorough testing of our conference solution which is based on > FreeSwitch / mod_conference, we noticed some 'audio clipping' during the > conference. Typically, if a conference member didn't have audio for some > time and then begins to speak, the first syllable will be clipped and > cannot be heard by the other conference participants. > > We ruled out a number of possible causes, and finally discovered that the > energy level detection algorith seems to be the source of the problem. > Setting the energy level for all participants to "0" heals the problem (of > course now the noise might be higher since every participant's noise > contributes to the total noise), but this setting is not very useful since > the "talking detection" seems to rely on the same level, i.e. with energy > "0", all members are marked talking all the time (which effectively renders > the "talking" display useless). > > Is there any chance to tune the energy level detection time (!), or to > de-couple talking detection an energy level detection in FreeSwitch? > > Best regards, > Roland > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/ac785f4f/attachment.html From vipkilla at gmail.com Thu Nov 3 15:12:27 2011 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 3 Nov 2011 08:12:27 -0400 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: Lloyd, as I said, I have Polycom's working with DNS SRV without NAPTR records (our DNS server, Windows Server 2003, does not have NAPTR capabilities.) We are running Polycom firmware 3.2.2 I struggled for a while getting it to work, but with the correct provisioning files, I got it to work. Please follow my configs strictly as any difference can cause it to not work. On Wed, Nov 2, 2011 at 7:22 PM, Lloyd Aloysius wrote: > DynDns not supporting NAPTR. I?received? reply from them "We do not support > NAPTR records."I am currently using DynDns for my DNS needs. > Any other recommendation to make Polycom phone work with DNS SRV records > without NAPTR records. > > > Thanks and regards, > > Lloyd > > > On Tue, Nov 1, 2011 at 4:57 PM, Lloyd Aloysius > wrote: >> >> Vik, >> I have the same setting in my configuration files. But the phone never >> register. I will follow the Brian instructions [create NAPTR Records] >> shortly. >> >> Thanks >> Lloyd >> >> On Tue, Nov 1, 2011 at 10:46 AM, Vik Killa wrote: >>> >>> I have Polycoms working with DNS SRV without NAPTR records >>> It was a matter of polycom provisioning files. >>> I attached the 2 important files and that last important file named >>> polycom_0004f21a0e87.cfg looks like this: >>> >>> >>> ? ? ? ?>> ? ? ? ? ? ? ? ?reg.1.displayName="Test UA" >>> ? ? ? ? ? ? ? ?reg.1.address="1001" >>> ? ? ? ? ? ? ? ?reg.1.label="Test UA" >>> ? ? ? ? ? ? ? ?reg.1.type="private" >>> ? ? ? ? ? ? ? ?reg.1.auth.userId="1001" >>> ? ? ? ? ? ? ? ?reg.1.auth.password="PASSWORD" >>> ? ? ? ? ? ? ? ?reg.1.server.1.address="mypbx.mydomain.com" >>> ? ? ? ? /> >>> >>> >>> >>> On Tue, Nov 1, 2011 at 10:13 AM, Lloyd Aloysius >>> wrote: >>> > Thank you for the reply . >>> > I miss the NAPTR records. Let me setup the NAPTR records and test >>> > again. >>> > Thanks and regards, >>> > >>> > Lloyd >>> > >>> > >>> > >>> > On Tue, Nov 1, 2011 at 9:27 AM, Brian West >>> > wrote: >>> >> >>> >> Ok you missed the part about the NAPTR records didn't you? ?Make sure >>> >> you >>> >> setup your NAPTR records correctly. ?I really don't get how people >>> >> think >>> >> they are optional. >>> >> I know this says >>> >> >>> >> optional?http://wiki.freeswitch.org/wiki/SIP_TLS#Step_5_-_DNS_NAPTR_.26_SRV_Records_.28optional.29 >>> >> but they aren't. >>> >> /b >>> >> On Nov 1, 2011, at 8:22 AM, Lloyd Aloysius wrote: >>> >> >>> >> But this fails to register. >>> >> >>> >> Thanks >>> >> Lloyd >>> >> >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> > >>> > >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 at peely.com Thu Nov 3 15:45:55 2011 From: freeswitch at peely.com (peely) Date: Thu, 3 Nov 2011 05:45:55 -0700 (PDT) Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? Message-ID: <1320324355799-6958891.post@n2.nabble.com> Hi, I've been playing about with the HA setup, and have set something up similar to the HA page on the Wiki with two FS boxes using MySQL Master <> Master replication and FreeSWITCH on the secondary box listening on un-mounted IPs. This bit all works very nicely and when both boxes are up I can "fsctl crash" the current active box, then mount the IPs on the soon-to-be-live box then issue "sofia recover", it's all very impressive! What I notice though is that when I bring the "crashed" box back up, starting the freeswitch process causes all calls in the database to be removed which are then replicated across to the active box's database, if I issue a "show calls count" on both boxes the figure goes to 0 immediately after the freeswitch process is started on the secondary. >From what I can see this doesn't affect things on the newly active box, a "status" shows channels in use, much greater than the number of calls, which gradually picks up again. I'm guessing freeswitch issues something like 'delete from calls' when it starts? For HA setups is this an appropriate thing to do? For reference, I'm currently running FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500). Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6958891.html Sent from the freeswitch-users mailing list archive at Nabble.com. From nmenoni at glotweb.com Thu Nov 3 15:52:02 2011 From: nmenoni at glotweb.com (=?ISO-8859-1?Q?Nicol=E1s_Menoni?=) Date: Thu, 3 Nov 2011 10:52:02 -0200 Subject: [Freeswitch-users] IM isn't working Message-ID: Hello, I'm having a problem that maybe at first it sounds basic but I couldn't solve it yet. The IM (instant message) was working for a time but after a moment it didn't work more. I'm able to send IMs from FS_CLI using "chat" command, but I can't make that the sip users could chat between them. The FS_CLI doesn't give me any info about the problem, I can't see anything from the logs, it llooks like as if everything is ok. Any help? Thank you! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/55670819/attachment.html From yehavi.bourvine at gmail.com Thu Nov 3 16:10:46 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 3 Nov 2011 15:10:46 +0200 Subject: [Freeswitch-users] How to paste text from Linux to pastebin.freeswitch.org? Message-ID: Hello, I have some logfile on Linux and would like to copy it to pastebin. Is there some way to upload the file? Currently I mark each screen and copy it, but there must be an easier and more reliable way... Furthermore, this way all the escape sequences in the log are shown as "ESC" and not parsed. Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/57df3cbd/attachment.html From miha at softnet.si Thu Nov 3 16:23:34 2011 From: miha at softnet.si (Miha Zoubek) Date: Thu, 03 Nov 2011 14:23:34 +0100 Subject: [Freeswitch-users] Radius client problems Message-ID: <4EB295D6.8000605@softnet.si> Hi, As now my freeswitch is working properly I would like to implement radius client on it for AAA. I have done as is written on http://wiki.freeswitch.org/wiki/Mod_rad_auth. My dialplan look like this: After I try to make phone call I get his: *2011-11-03 13:33:54.582920 [ERR] mod_rad_auth.c:347 Error loading radius dictionary 2011-11-03 13:33:54.582920 [ERR] mod_rad_auth.c:546 An error occured during RADIUS Authentication(RC=-1) 2011-11-03 13:33:54.582920 [ERR] mod_rad_auth.c:702 An error occured during radius authorization.* ok, I thought that link location is wrong but it is not: References: <539AC2B9-E137-4CAF-BFDC-DED583F66D7F@phone.com> <5544517E-9301-404C-A55E-31B49719E14B@phone.com> Message-ID: <4EB2A33A.8080504@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi tyler, I tried this tpday, but no success. No video codec processing - only audio. Am 02.11.2011 23:03, schrieb Tyler Winter: > I was able to add video to a non-video initiated call by using the > *sip_renegotiate_codec_on_reinvite* setting which ended up being > elusive. > > My understanding of this setting from reviewing the code, is: - the > default behavior of FS is that once the codec is negotiated, > re-INVITEs skip the codec negotiation logic and only process > changes in media IP/port/etc. - by turning this setting on, every > re-INVITE goes through the same codec negotiation logic as happens > during the initial INVITE. > > Hope this helps someone else too. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk6yozoACgkQ4tZeNddg3dzDtQCbBV8ZoCrOf8wAT5hZg1I52IgW hFIAn2TYCfU4BjTZZ54AyTwKkujbmsa6 =JPSB -----END PGP SIGNATURE----- From roland at haenel.me Thu Nov 3 17:23:05 2011 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Thu, 3 Nov 2011 15:23:05 +0100 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: Without exact knowledge, I would guess this is conference-specific, since in normal one-to-one conversations, there is no such thing as energy-level detection and audio mixing. If you consider VAD (voice activity detection) performed by SIP endpoints (aka phones), this is a similar thing but in this particular case, I have verified that VAD is not an issue. Roland 2011/11/3 Hynek Cihlar > Is it only conference related or may it affect other apps? > > Hynek > > > > 2011/11/3 Roland H?nel > >> Hi, >> >> During thorough testing of our conference solution which is based on >> FreeSwitch / mod_conference, we noticed some 'audio clipping' during the >> conference. Typically, if a conference member didn't have audio for some >> time and then begins to speak, the first syllable will be clipped and >> cannot be heard by the other conference participants. >> >> We ruled out a number of possible causes, and finally discovered that the >> energy level detection algorith seems to be the source of the problem. >> Setting the energy level for all participants to "0" heals the problem (of >> course now the noise might be higher since every participant's noise >> contributes to the total noise), but this setting is not very useful since >> the "talking detection" seems to rely on the same level, i.e. with energy >> "0", all members are marked talking all the time (which effectively renders >> the "talking" display useless). >> >> Is there any chance to tune the energy level detection time (!), or to >> de-couple talking detection an energy level detection in FreeSwitch? >> >> Best regards, >> Roland >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Gru?, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/4967e3ac/attachment.html From yehavi.bourvine at gmail.com Thu Nov 3 17:30:28 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 3 Nov 2011 16:30:28 +0200 Subject: [Freeswitch-users] Automatic fax formating sending via Freeswitch Message-ID: Hello, This is not really a FreeSwitch question, but I guess people here might have answer to this... I have to generate fax from two files: template and list of fields to replace in the template. I would like the template to be a tectual thing so I can use sed to replace the fields in it, and then convert the result to PFD/TIFF and send it via FS. Is there some tectual format (like Tex, SVG) that can be eaily created by some WISIWIG software? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/b2d59c32/attachment.html From lloyd.aloysius at sunteltech.ca Thu Nov 3 17:53:50 2011 From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius) Date: Thu, 3 Nov 2011 10:53:50 -0400 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: Vik, The configuration files are correct. 1. Ploycom Phones never send a registration to *pbx.mydomain.com* 2. IF I create *A* record [*pbx.mydomain.com *] , then it send the Registration request and Success. I think polycom need DNS SRV and NATPR records. May be you have A record for *pbx.mydomain.com *that is the reason it is working. ===== === ======== sip.cfg ==== ======== Thanks and regards, Lloyd * * * * On Thu, Nov 3, 2011 at 8:12 AM, Vik Killa wrote: > Lloyd, as I said, I have Polycom's working with DNS SRV without NAPTR > records (our DNS server, Windows Server 2003, does not have NAPTR > capabilities.) > We are running Polycom firmware 3.2.2 > I struggled for a while getting it to work, but with the correct > provisioning files, I got it to work. Please follow my configs > strictly as any difference can cause it to not work. > > > On Wed, Nov 2, 2011 at 7:22 PM, Lloyd Aloysius > wrote: > > DynDns not supporting NAPTR. I received reply from them "We do not > support > > NAPTR records."I am currently using DynDns for my DNS needs. > > Any other recommendation to make Polycom phone work with DNS SRV records > > without NAPTR records. > > > > > > Thanks and regards, > > > > Lloyd > > > > > > On Tue, Nov 1, 2011 at 4:57 PM, Lloyd Aloysius > > > wrote: > >> > >> Vik, > >> I have the same setting in my configuration files. But the phone never > >> register. I will follow the Brian instructions [create NAPTR Records] > >> shortly. > >> > >> Thanks > >> Lloyd > >> > >> On Tue, Nov 1, 2011 at 10:46 AM, Vik Killa wrote: > >>> > >>> I have Polycoms working with DNS SRV without NAPTR records > >>> It was a matter of polycom provisioning files. > >>> I attached the 2 important files and that last important file named > >>> polycom_0004f21a0e87.cfg looks like this: > >>> > >>> > >>> >>> reg.1.displayName="Test UA" > >>> reg.1.address="1001" > >>> reg.1.label="Test UA" > >>> reg.1.type="private" > >>> reg.1.auth.userId="1001" > >>> reg.1.auth.password="PASSWORD" > >>> reg.1.server.1.address="mypbx.mydomain.com" > >>> /> > >>> > >>> > >>> > >>> On Tue, Nov 1, 2011 at 10:13 AM, Lloyd Aloysius > >>> wrote: > >>> > Thank you for the reply . > >>> > I miss the NAPTR records. Let me setup the NAPTR records and test > >>> > again. > >>> > Thanks and regards, > >>> > > >>> > Lloyd > >>> > > >>> > > >>> > > >>> > On Tue, Nov 1, 2011 at 9:27 AM, Brian West > >>> > wrote: > >>> >> > >>> >> Ok you missed the part about the NAPTR records didn't you? Make > sure > >>> >> you > >>> >> setup your NAPTR records correctly. I really don't get how people > >>> >> think > >>> >> they are optional. > >>> >> I know this says > >>> >> > >>> >> optional > http://wiki.freeswitch.org/wiki/SIP_TLS#Step_5_-_DNS_NAPTR_.26_SRV_Records_.28optional.29 > >>> >> but they aren't. > >>> >> /b > >>> >> On Nov 1, 2011, at 8:22 AM, Lloyd Aloysius wrote: > >>> >> > >>> >> But this fails to register. > >>> >> > >>> >> Thanks > >>> >> Lloyd > >>> >> > >>> >> > >>> >> FreeSWITCH-users mailing list > >>> >> FreeSWITCH-users at lists.freeswitch.org > >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >> > >>> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >> http://www.freeswitch.org > >>> >> > >>> > > >>> > > >>> > > >>> > FreeSWITCH-users mailing list > >>> > FreeSWITCH-users at lists.freeswitch.org > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > > >>> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> > http://www.freeswitch.org > >>> > > >>> > > >>> > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/df6d0961/attachment-0001.html From hynek.cihlar at gmail.com Thu Nov 3 18:13:26 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Thu, 3 Nov 2011 16:13:26 +0100 Subject: [Freeswitch-users] Call quality In-Reply-To: References: <-4241905887276648699@unknownmsgid> Message-ID: Thanks all for the suggestions. I will keep you posted. Hynek On Wed, Nov 2, 2011 at 3:54 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Always start by updating all of them to the very latest GIT. > Then systematically eliminate things until you zero in on the problem. > > > On Tue, Nov 1, 2011 at 1:39 PM, Hynek Cihlar wrote: > >> Would anybody know, if the call quality could be dependant on the >> number of media switches the call is routed through? >> >> Currently I'm testing a system with several freeswitches in the call >> way (up to 4) and I'm noticing very short gaps, not very disturbing >> but noticable. >> >> What's your experience? >> >> Sent from my mobile device >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/a9c84a52/attachment.html From infos at madovsky.org Thu Nov 3 18:54:30 2011 From: infos at madovsky.org (Madovsky) Date: Thu, 3 Nov 2011 11:54:30 -0400 Subject: [Freeswitch-users] Automatic fax formating sending via Freeswitch References: Message-ID: <40650EFDEA0042788AF84CBE8236745A@e1705> convert from Imagemagick ----- Original Message ----- From: Yehavi Bourvine To: FreeSWITCH Users Help Sent: Thursday, November 03, 2011 10:30 AM Subject: [Freeswitch-users] Automatic fax formating sending via Freeswitch Hello, This is not really a FreeSwitch question, but I guess people here might have answer to this... I have to generate fax from two files: template and list of fields to replace in the template. I would like the template to be a tectual thing so I can use sed to replace the fields in it, and then convert the result to PFD/TIFF and send it via FS. Is there some tectual format (like Tex, SVG) that can be eaily created by some WISIWIG software? 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/20111103/dafe7e25/attachment.html From dipkumar.mehta at gmail.com Thu Nov 3 19:24:15 2011 From: dipkumar.mehta at gmail.com (Dip Mehta) Date: Thu, 3 Nov 2011 21:54:15 +0530 Subject: [Freeswitch-users] Bad Contact Header Message-ID: Hi , I am trying to register x-lite 4.1 on freeswitch. And i am using the default extensions : 1003 and password 1234 However i am unable to get this registered and get the following error on the console. Output of nano /usr/local/freeswitch/conf/directory/default/1003.xml recv 572 bytes from udp/[122.176.XX.XX:]:13334 at 16:18:23.774556: ------------------------------------------------------------------------ REGISTER sip:199.167.XX.XX: SIP/2.0 Via: SIP/2.0/UDP 122.176.144.226:13334 ;branch=z9hG4bK-d8754z-9ba4ee0f5e6ae5a7-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Extension 1003" From: "Extension 1003";tag=77bad5c7 Call-ID: MTRmNzQ3MDUyMTM4MGNhOGUzZjdlNWYyNWRlZDNjNmQ. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0 ------------------------------------------------------------------------ send 715 bytes to udp/[122.176.XX.XX:]:13334 at 16:18:23.775265: ------------------------------------------------------------------------ * SIP/2.0 401 Unauthorized* Via: SIP/2.0/UDP 122.176.144.226:13334 ;branch=z9hG4bK-d8754z-9ba4ee0f5e6ae5a7-1---d8754z-;rport=13334 From: "Extension 1003";tag=77bad5c7 To: "Extension 1003" ;tag=gpjScQ0eXXj6K Call-ID: MTRmNzQ3MDUyMTM4MGNhOGUzZjdlNWYyNWRlZDNjNmQ. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7b10506 2011-10-26 08-45-04 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="199.167.145.102", nonce="740c2df0-0637-11e1-9e6e-6b39c14e0024", algorithm=MD5, qop="auth" Content-Length: 0 send 340 bytes to udp/[122.176.144.226]:13293 at 16:10:09.247665: ------------------------------------------------------------------------ * SIP/2.0 400 Bad Contact Header* Via: SIP/2.0/UDP 122.176.XX.XX:;branch=z9hG4bK-d8754z-7c90ddcd2100ff77-1---d8754z-;rport=13293 From: "1003" ;tag=d09ca84c To: "1003" ;tag=DU6e75D452FeH Call-ID: YjM3NWZiZjdkMzMwNTJhMjZkNmRhMWFkMGVkZGM1ZWI. CSeq: 4 REGISTER Content-Length: Errors suggest this to be authentication issue. However i am not able to resolve this. Please advice. Regards Dip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/568c8a9c/attachment.html From cliff at develix.com Thu Nov 3 19:59:18 2011 From: cliff at develix.com (Cliff Wells) Date: Thu, 03 Nov 2011 09:59:18 -0700 Subject: [Freeswitch-users] How to paste text from Linux to pastebin.freeswitch.org? In-Reply-To: References: Message-ID: <1320339558.2327.46.camel@portable-evil> https://launchpad.net/nautilus-pastebin On Thu, 2011-11-03 at 15:10 +0200, Yehavi Bourvine wrote: > Hello, > > I have some logfile on Linux and would like to copy it to pastebin. > Is there some way to upload the file? Currently I mark each screen and > copy it, but there must be an easier and more reliable way... > Furthermore, this way all the escape sequences in the log are shown as > "ESC" and not parsed. > > 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 From anthony.minessale at gmail.com Thu Nov 3 20:12:19 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Nov 2011 12:12:19 -0500 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: yes? set it to something between what it was and 0 ? try 200 100 50 ? 2011/11/3 Roland H?nel > Hi, > > During thorough testing of our conference solution which is based on > FreeSwitch / mod_conference, we noticed some 'audio clipping' during the > conference. Typically, if a conference member didn't have audio for some > time and then begins to speak, the first syllable will be clipped and > cannot be heard by the other conference participants. > > We ruled out a number of possible causes, and finally discovered that the > energy level detection algorith seems to be the source of the problem. > Setting the energy level for all participants to "0" heals the problem (of > course now the noise might be higher since every participant's noise > contributes to the total noise), but this setting is not very useful since > the "talking detection" seems to rely on the same level, i.e. with energy > "0", all members are marked talking all the time (which effectively renders > the "talking" display useless). > > Is there any chance to tune the energy level detection time (!), or to > de-couple talking detection an energy level detection in FreeSwitch? > > Best regards, > Roland > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/9a3844d2/attachment-0001.html From dave at clancysystems.com Thu Nov 3 20:39:54 2011 From: dave at clancysystems.com (Dave) Date: Thu, 3 Nov 2011 11:39:54 -0600 Subject: [Freeswitch-users] Help with Using Mod_AVMD Message-ID: <87A1ED3B86EA4F8AAB18439700A28B42@clancysystems.com> Hello, I am creating an external application that uses FreeSWITCH ESL to initiate an outbound call and then run a LUA script. Within the LUA script I am executing AVMD. I want to see if its a human or voicemail on the other end. The outbound call is made and the LUA script executes. Currently I would just like to see a consoleLog entry made by the onInput function in the LUA script, but that function does not seem to be called. AVMD starts the file is streamed then AVMD stops. As shown in the log. I think I'm missing something. Dave Goodwin Pertinent FreeSWITCH Log Info: ------------------------------------------------------------------------------------------------------- EXECUTE sofia/internal/7202126254 at xxx.xx.xxx.xx avmd(start) 2011-11-03 11:05:27.291678 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/7202126254 at xxx.xx.xxx.xx 2011-11-03 11:05:28.311249 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms 2011-11-03 11:05:28.350313 [DEBUG] switch_rtp.c:3082 Correct ip/port confirmed. 2011-11-03 11:05:40.235535 [DEBUG] switch_ivr_play_say.c:1649 done playing file EXECUTE sofia/internal/7202126254 at xxx.xx.xxx.xx avmd(stop) 2011-11-03 11:05:40.237488 [DEBUG] switch_core_media_bug.c:467 Removing BUG from sofia/internal/7202126254 at xxx.xx.xxx.xx Lua Script: ------------------------------------------------------------------------------------------------------- function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '1' and human_detected == false then human_detected = true freeswitch.consoleLog("INFO","Human Detected\n") return "break" end if type == "event" and voicemail_detected == false then voicemail_detected = true freeswitch.consoleLog("INFO","Voicemail Detected\n") return "break" else freeswitch.consoleLog("INFO","Nothing Happened\n") return "break" end end if session:ready() then --- the call has been answered. session:setInputCallback("onInput") session:sleep(1000) session:execute("avmd","start") session:sleep(1000) session:streamFile("ivr/" .. wav_file) session:execute("avmd","stop") session:sleep(1000) fileName = "Completed " .. tostring(call_from_num) .. os.time() .. tostring(call_to_num) freeswitch.consoleLog("INFO","FileName = " .. fileName .. "\n") local myfile, ErrStr = io.open("log/outcall/" .. fileName,"w") if myfile then myfile:write(call_from_num .. "|", call_to_num .. "|" , os.date()) myfile:flush() myfile:close() else freeswitch.consoleLog("INFO","Error writing File = '" .. ErrStr .. "'\n") end --if end --if -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/f63c8566/attachment.html From vipkilla at gmail.com Thu Nov 3 20:44:33 2011 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 3 Nov 2011 13:44:33 -0400 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: I definitely do NOT have NATPR records and the A record for pbx.mydomain.com does NOT point to my FS server. Trust me, I created the DNS records. I know I have DNS SRV running WITHOUT NATPR because I have 2 FS servers receiving calls/registrations from these Polycoms. Your polycom configs are NOT the same as mine. I'm TELLING you, you need the exact SAME polycom configs as me. I spent hours trying to get it to work and I succeeded, so either take my advice and use same configs or get NATPR records. On Thu, Nov 3, 2011 at 10:53 AM, Lloyd Aloysius wrote: > Vik, > ?The configuration files are correct. > 1. Ploycom Phones never send a registration to pbx.mydomain.com > 2. IF I create A record [pbx.mydomain.com?] , then it send the Registration > request and Success. > I think polycom need DNS SRV and NATPR records. > May be you have ?A record for?pbx.mydomain.com that is the reason it is > working. > ===== > > > > ? reg.1.type="private" reg.1.thirdPartyName="" reg.1.auth.userId="204" > reg.1.auth.password="************" reg.1.server.1.address="" > reg.1.server.1.port="" reg.1.server.1.transport="" reg.1.server.1.expires="" > reg.1.server.1.register="" reg.1.server.1.retryTimeout="" > reg.1.server.1.expires.lineSeize="" reg.1.acd-login-logout="0" > reg.1.acd-agent-available="0" reg.1.ringType="2" reg.1.lineKeys="2" > reg.1.callsPerLineKey="1" reg.2.displayName="" reg.2.address="" > reg.2.label="" reg.2.type="" reg.2.thirdPartyName="" reg.2.auth.userId="" > reg.2.auth.password="" reg.2.server.1.address="" reg.2.server.1.port="" > reg.2.server.1.transport="" reg.2.server.1.expires="" > reg.2.server.1.register="" reg.2.server.1.retryTimeout="" > reg.2.server.1.expires.lineSeize="" reg.2.acd-login-logout="" > reg.2.acd-agent-available="" reg.2.ringType="" reg.2.lineKeys="" > reg.2.callsPerLineKey="" /> > > === > ======== sip.cfg ==== > > > xsi:noNamespaceSchemaLocation="polycomConfig.xsd"> > ? > ? ? msg.mwi.2.callBackMode="registration"> > ? > ? > ? ? > ? ? ? voIpProt.SIP.outboundProxy.address=""> > ? ? > ? ? voIpProt.server.1.port="5060" voIpProt.server.1.transport="DNSnaptr" > voIpProt.server.1.expires="60" voIpProt.server.1.register="1" > voIpProt.server.1.retryTimeout="0" voIpProt.server.1.expires.lineSeize="30" > voIpProt.server.2.address="" voIpProt.server.2.port="0"> > ? > ? tcpIpApp.sntp.address="pool.ntp.org" tcpIpApp.sntp.gmtOffset="-18000" /> > > ======== > Thanks and regards, > > Lloyd > > > > On Thu, Nov 3, 2011 at 8:12 AM, Vik Killa wrote: >> >> Lloyd, as I said, I have Polycom's working with DNS SRV without NAPTR >> records (our DNS server, Windows Server 2003, does not have NAPTR >> capabilities.) >> We are running Polycom firmware 3.2.2 >> I struggled for a while getting it to work, but with the correct >> provisioning files, I got it to work. Please follow my configs >> strictly as any difference can cause it to not work. >> >> >> On Wed, Nov 2, 2011 at 7:22 PM, Lloyd Aloysius >> wrote: >> > DynDns not supporting NAPTR. I?received? reply from them "We do not >> > support >> > NAPTR records."I am currently using DynDns for my DNS needs. >> > Any other recommendation to make Polycom phone work with DNS SRV records >> > without NAPTR records. >> > >> > >> > Thanks and regards, >> > >> > Lloyd >> > >> > >> > On Tue, Nov 1, 2011 at 4:57 PM, Lloyd Aloysius >> > >> > wrote: >> >> >> >> Vik, >> >> I have the same setting in my configuration files. But the phone never >> >> register. I will follow the Brian instructions [create NAPTR Records] >> >> shortly. >> >> >> >> Thanks >> >> Lloyd >> >> >> >> On Tue, Nov 1, 2011 at 10:46 AM, Vik Killa wrote: >> >>> >> >>> I have Polycoms working with DNS SRV without NAPTR records >> >>> It was a matter of polycom provisioning files. >> >>> I attached the 2 important files and that last important file named >> >>> polycom_0004f21a0e87.cfg looks like this: >> >>> >> >>> >> >>> ? ? ? ?> >>> ? ? ? ? ? ? ? ?reg.1.displayName="Test UA" >> >>> ? ? ? ? ? ? ? ?reg.1.address="1001" >> >>> ? ? ? ? ? ? ? ?reg.1.label="Test UA" >> >>> ? ? ? ? ? ? ? ?reg.1.type="private" >> >>> ? ? ? ? ? ? ? ?reg.1.auth.userId="1001" >> >>> ? ? ? ? ? ? ? ?reg.1.auth.password="PASSWORD" >> >>> ? ? ? ? ? ? ? ?reg.1.server.1.address="mypbx.mydomain.com" >> >>> ? ? ? ? /> >> >>> >> >>> >> >>> >> >>> On Tue, Nov 1, 2011 at 10:13 AM, Lloyd Aloysius >> >>> wrote: >> >>> > Thank you for the reply . >> >>> > I miss the NAPTR records. Let me setup the NAPTR records and test >> >>> > again. >> >>> > Thanks and regards, >> >>> > >> >>> > Lloyd >> >>> > >> >>> > >> >>> > >> >>> > On Tue, Nov 1, 2011 at 9:27 AM, Brian West >> >>> > wrote: >> >>> >> >> >>> >> Ok you missed the part about the NAPTR records didn't you? ?Make >> >>> >> sure >> >>> >> you >> >>> >> setup your NAPTR records correctly. ?I really don't get how people >> >>> >> think >> >>> >> they are optional. >> >>> >> I know this says >> >>> >> >> >>> >> >> >>> >> optional?http://wiki.freeswitch.org/wiki/SIP_TLS#Step_5_-_DNS_NAPTR_.26_SRV_Records_.28optional.29 >> >>> >> but they aren't. >> >>> >> /b >> >>> >> On Nov 1, 2011, at 8:22 AM, Lloyd Aloysius wrote: >> >>> >> >> >>> >> But this fails to register. >> >>> >> >> >>> >> Thanks >> >>> >> Lloyd >> >>> >> >> >>> >> >> >>> >> FreeSWITCH-users mailing list >> >>> >> FreeSWITCH-users at lists.freeswitch.org >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >> >>> >> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> >> http://www.freeswitch.org >> >>> >> >> >>> > >> >>> > >> >>> > >> >>> > FreeSWITCH-users mailing list >> >>> > FreeSWITCH-users at lists.freeswitch.org >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> > >> >>> > >> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> > http://www.freeswitch.org >> >>> > >> >>> > >> >>> >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> > >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From roland at haenel.me Thu Nov 3 21:03:56 2011 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Thu, 3 Nov 2011 19:03:56 +0100 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: Setting it to 10 or so doesn't help, it seems that just the threshold of detection is changed (ok, that's what is expected), not the time until the decision is made. The audio clipping is a timing issue. Does there exist something like a "minimum time" that this algorithm needs to detect talking after silence? What happens to the audio that was sent by that member during this time, does.the algorithm discard that? Greetings, Roland Am 03.11.2011 18:20 schrieb "Anthony Minessale" : > yes? set it to something between what it was and 0 ? > > > > try 200 100 50 ? > > 2011/11/3 Roland H?nel > >> Hi, >> >> During thorough testing of our conference solution which is based on >> FreeSwitch / mod_conference, we noticed some 'audio clipping' during the >> conference. Typically, if a conference member didn't have audio for some >> time and then begins to speak, the first syllable will be clipped and >> cannot be heard by the other conference participants. >> >> We ruled out a number of possible causes, and finally discovered that the >> energy level detection algorith seems to be the source of the problem. >> Setting the energy level for all participants to "0" heals the problem (of >> course now the noise might be higher since every participant's noise >> contributes to the total noise), but this setting is not very useful since >> the "talking detection" seems to rely on the same level, i.e. with energy >> "0", all members are marked talking all the time (which effectively renders >> the "talking" display useless). >> >> Is there any chance to tune the energy level detection time (!), or to >> de-couple talking detection an energy level detection in FreeSwitch? >> >> Best regards, >> Roland >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/1545fff9/attachment-0001.html From covici at ccs.covici.com Thu Nov 3 21:36:19 2011 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 03 Nov 2011 14:36:19 -0400 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: <32764.1320345379@ccs.covici.com> I found that 0 was the only thing that prevented the clipping -- particularly on the recording. Anthony Minessale wrote: > yes? set it to something between what it was and 0 ? > > > > try 200 100 50 ? > > 2011/11/3 Roland H?nel > > > Hi, > > > > During thorough testing of our conference solution which is based on > > FreeSwitch / mod_conference, we noticed some 'audio clipping' during the > > conference. Typically, if a conference member didn't have audio for some > > time and then begins to speak, the first syllable will be clipped and > > cannot be heard by the other conference participants. > > > > We ruled out a number of possible causes, and finally discovered that the > > energy level detection algorith seems to be the source of the problem. > > Setting the energy level for all participants to "0" heals the problem (of > > course now the noise might be higher since every participant's noise > > contributes to the total noise), but this setting is not very useful since > > the "talking detection" seems to rely on the same level, i.e. with energy > > "0", all members are marked talking all the time (which effectively renders > > the "talking" display useless). > > > > Is there any chance to tune the energy level detection time (!), or to > > de-couple talking detection an energy level detection in FreeSwitch? > > > > Best regards, > > Roland > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From anthony.minessale at gmail.com Thu Nov 3 21:47:54 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Nov 2011 13:47:54 -0500 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: The setting is an arbitrary number of the energy score required to count as talking. check your environments, its very specific to the surroundings. We run it at 200 in the main FS conf and its rarely an issue. 2011/11/3 Roland H?nel > Setting it to 10 or so doesn't help, it seems that just the threshold of > detection is changed (ok, that's what is expected), not the time until the > decision is made. > > The audio clipping is a timing issue. > Does there exist something like a "minimum time" that this algorithm needs > to detect talking after silence? What happens to the audio that was sent by > that member during this time, does.the algorithm discard that? > > Greetings, > Roland > Am 03.11.2011 18:20 schrieb "Anthony Minessale" < > anthony.minessale at gmail.com>: > > yes? set it to something between what it was and 0 ? >> >> >> >> try 200 100 50 ? >> >> 2011/11/3 Roland H?nel >> >>> Hi, >>> >>> During thorough testing of our conference solution which is based on >>> FreeSwitch / mod_conference, we noticed some 'audio clipping' during the >>> conference. Typically, if a conference member didn't have audio for some >>> time and then begins to speak, the first syllable will be clipped and >>> cannot be heard by the other conference participants. >>> >>> We ruled out a number of possible causes, and finally discovered that >>> the energy level detection algorith seems to be the source of the problem. >>> Setting the energy level for all participants to "0" heals the problem (of >>> course now the noise might be higher since every participant's noise >>> contributes to the total noise), but this setting is not very useful since >>> the "talking detection" seems to rely on the same level, i.e. with energy >>> "0", all members are marked talking all the time (which effectively renders >>> the "talking" display useless). >>> >>> Is there any chance to tune the energy level detection time (!), or to >>> de-couple talking detection an energy level detection in FreeSwitch? >>> >>> Best regards, >>> Roland >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/6dfa0f21/attachment.html From anthony.minessale at gmail.com Thu Nov 3 21:57:35 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Nov 2011 13:57:35 -0500 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: <1320324355799-6958891.post@n2.nabble.com> References: <1320324355799-6958891.post@n2.nabble.com> Message-ID: it deletes based on hostname= so set the hostname different on the 2 boxes or use the switchname param in switch.conf.xml to give difft virtual hostnames On Thu, Nov 3, 2011 at 7:45 AM, peely wrote: > Hi, > > I've been playing about with the HA setup, and have set something up > similar > to the HA page on the Wiki with two FS boxes using MySQL Master <> Master > replication and FreeSWITCH on the secondary box listening on un-mounted > IPs. > > This bit all works very nicely and when both boxes are up I can "fsctl > crash" the current active box, then mount the IPs on the soon-to-be-live > box > then issue "sofia recover", it's all very impressive! > > What I notice though is that when I bring the "crashed" box back up, > starting the freeswitch process causes all calls in the database to be > removed which are then replicated across to the active box's database, if I > issue a "show calls count" on both boxes the figure goes to 0 immediately > after the freeswitch process is started on the secondary. > > >From what I can see this doesn't affect things on the newly active box, a > "status" shows channels in use, much greater than the number of calls, > which > gradually picks up again. > > I'm guessing freeswitch issues something like 'delete from calls' when it > starts? For HA setups is this an appropriate thing to do? > > For reference, I'm currently running FreeSWITCH Version 1.0.head > (git-fa3fa30 2011-10-07 11-20-36 -0500). > > > Cheers, > > > > Neil. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6958891.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/72a62495/attachment.html From evan at corpwest.com Thu Nov 3 03:11:28 2011 From: evan at corpwest.com (Evan P. Hall) Date: Thu, 3 Nov 2011 00:11:28 +0000 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: It will not look up the NAPTR or SRV records at all since you have specified port 5060. Don't specify the port number and it should work. We run a large number of polycom phones without NAPTR records here. However we don't use the outboundProxy attributes at all. We use voIpProt.server.x.address or the phone's reg.x.server.x.address to tell the phones where to register. You shouldn't need to specify outboundProxy unless you need your signaling packets to go to a different server than the registration. -Evan From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lloyd Aloysius Sent: Tuesday, November 01, 2011 6:23 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV Brian, Thank you for the reply. I set the voIpProt.SIP.transport="DNSnaptr" http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/7f316d41/attachment-0001.html From rml at tollfreeforwarding.com Thu Nov 3 04:50:20 2011 From: rml at tollfreeforwarding.com (RaviRaj Mulasa) Date: Thu, 3 Nov 2011 01:50:20 +0000 Subject: [Freeswitch-users] Send Event from FreeSWITCH to RTMP client Message-ID: <8B94625BC339264DBA61E314BE9EC2CF230E9493@EXCH125.IFN.com> Hi Please let me know how to send an Event from FreeSWITCH to a RTMP client. I saw a Java script function in index.html in Flex Client provided by FreeSWITCH : function onEvent(data) { onDebug("Got event: " + data); } originate rtmp/rtmp-session-id &event(TEST) , no luck! Thanks RaviRaj. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/884864e7/attachment.html From roman at dissauer.net Thu Nov 3 06:20:38 2011 From: roman at dissauer.net (Roman Dissauer) Date: Thu, 3 Nov 2011 11:20:38 +0800 Subject: [Freeswitch-users] one gateway two registrations Message-ID: <2F46CC4C-70D7-4B94-A229-9E22834D5B64@dissauer.net> I have a gateway defined in directory, this works fine but it ties to register over both internal and external sip_profile. How can I restrict it to just the external profile? Thanks! Roman From th982a at googlemail.com Thu Nov 3 22:05:27 2011 From: th982a at googlemail.com (Tamer Higazi) Date: Thu, 03 Nov 2011 20:05:27 +0100 Subject: [Freeswitch-users] securing sip channel Message-ID: <4EB2E5F7.3090001@googlemail.com> Hi! I am interested to know if there is a way to secure the registration against account hacking. for example, I would create an SIP Account server side, there would be the possibility that only users with the static IP-Address could login to the assigned account. Is there one other way like kerberos ticket authentication or public secret key with ipsec?! For any advise I would thank you Tamer From Jacob.E.Miles at L-3Com.com Thu Nov 3 23:15:30 2011 From: Jacob.E.Miles at L-3Com.com (Jacob.E.Miles at L-3Com.com) Date: Thu, 3 Nov 2011 15:15:30 -0500 Subject: [Freeswitch-users] Newbie Message-ID: I am new to FreeSWITCH, testing to make sure it has all the functionality we need, so I have a few questions I would like to ask. I am coming from using Asterisk PBX and some Cisco Call Manager. As you see my question below you will realize we are currently using SCCP (skinny) based phones, I am trying to talk out customer into moving to SIP phones but no luck yet. 1) Is there good SCCP (skinny) support? 2) If so, which phones (79XX)? 3) Is there Asterisk AMI type messaging to control SCCP phones? Jacob Miles Software Engineer III L-3 Communications - Integrated Systems Greenville Jacob.E.Miles at L-3Com.com 903.457.4422 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/bf01460e/attachment.html From rml at tollfreeforwarding.com Thu Nov 3 23:39:21 2011 From: rml at tollfreeforwarding.com (RaviRaj Mulasa) Date: Thu, 3 Nov 2011 20:39:21 +0000 Subject: [Freeswitch-users] FW: Send Event/User Channel Variable from FreeSWITCH to RTMP client Message-ID: <8B94625BC339264DBA61E314BE9EC2CF230E95D1@EXCH125.IFN.com> Hi Please let us know how to send an Event/ User Channel Variable from FreeSWITCH to a RTMP client. Thanks RaviRaj. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/f3ee1db4/attachment.html From roland at haenel.me Fri Nov 4 00:12:12 2011 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Thu, 3 Nov 2011 22:12:12 +0100 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: Anthony, The problem is not that audio is clipped because its "too silent", so adjusting the value seems to be the wrong handle for this problem. It seems that the time needed to make the decision (regardless of the energy threshold value) is the core of the issue, i.e. the level may bei 100, 200 or 300, it takes some XX ms to detect talking and all audio that happens before the detection gets lost. Right? Greetings, Roland 2011/11/3 Anthony Minessale > The setting is an arbitrary number of the energy score required to count > as talking. > > > check your environments, its very specific to the surroundings. > We run it at 200 in the main FS conf and its rarely an issue. > > > > > 2011/11/3 Roland H?nel > >> Setting it to 10 or so doesn't help, it seems that just the threshold of >> detection is changed (ok, that's what is expected), not the time until the >> decision is made. >> >> The audio clipping is a timing issue. >> Does there exist something like a "minimum time" that this algorithm >> needs to detect talking after silence? What happens to the audio that was >> sent by that member during this time, does.the algorithm discard that? >> >> Greetings, >> Roland >> Am 03.11.2011 18:20 schrieb "Anthony Minessale" < >> anthony.minessale at gmail.com>: >> >> yes? set it to something between what it was and 0 ? >>> >>> >>> >>> try 200 100 50 ? >>> >>> 2011/11/3 Roland H?nel >>> >>>> Hi, >>>> >>>> During thorough testing of our conference solution which is based on >>>> FreeSwitch / mod_conference, we noticed some 'audio clipping' during the >>>> conference. Typically, if a conference member didn't have audio for some >>>> time and then begins to speak, the first syllable will be clipped and >>>> cannot be heard by the other conference participants. >>>> >>>> We ruled out a number of possible causes, and finally discovered that >>>> the energy level detection algorith seems to be the source of the problem. >>>> Setting the energy level for all participants to "0" heals the problem (of >>>> course now the noise might be higher since every participant's noise >>>> contributes to the total noise), but this setting is not very useful since >>>> the "talking detection" seems to rely on the same level, i.e. with energy >>>> "0", all members are marked talking all the time (which effectively renders >>>> the "talking" display useless). >>>> >>>> Is there any chance to tune the energy level detection time (!), or to >>>> de-couple talking detection an energy level detection in FreeSwitch? >>>> >>>> Best regards, >>>> Roland >>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Gru?, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/adb8d962/attachment-0001.html From anthony.minessale at gmail.com Fri Nov 4 00:22:27 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Nov 2011 16:22:27 -0500 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: it's not, most likely you have a jitterbuffer on your phone that is recharging when you start talking. are you on the latest GIT where we have the waste bandwidth flag on permanently for all callers? To avoid problems like this conferences now always send rtp to people even when they are not getting any useful input. 2011/11/3 Roland H?nel > Anthony, > > The problem is not that audio is clipped because its "too silent", so > adjusting the value seems to be the wrong handle for this problem. > > It seems that the time needed to make the decision (regardless of the > energy threshold value) is the core of the issue, i.e. the level may bei > 100, 200 or 300, it takes some XX ms to detect talking and all audio that > happens before the detection gets lost. Right? > > Greetings, > Roland > > > > 2011/11/3 Anthony Minessale > >> The setting is an arbitrary number of the energy score required to count >> as talking. >> >> >> check your environments, its very specific to the surroundings. >> We run it at 200 in the main FS conf and its rarely an issue. >> >> >> >> >> 2011/11/3 Roland H?nel >> >>> Setting it to 10 or so doesn't help, it seems that just the threshold of >>> detection is changed (ok, that's what is expected), not the time until the >>> decision is made. >>> >>> The audio clipping is a timing issue. >>> Does there exist something like a "minimum time" that this algorithm >>> needs to detect talking after silence? What happens to the audio that was >>> sent by that member during this time, does.the algorithm discard that? >>> >>> Greetings, >>> Roland >>> Am 03.11.2011 18:20 schrieb "Anthony Minessale" < >>> anthony.minessale at gmail.com>: >>> >>> yes? set it to something between what it was and 0 ? >>>> >>>> >>>> >>>> try 200 100 50 ? >>>> >>>> 2011/11/3 Roland H?nel >>>> >>>>> Hi, >>>>> >>>>> During thorough testing of our conference solution which is based on >>>>> FreeSwitch / mod_conference, we noticed some 'audio clipping' during the >>>>> conference. Typically, if a conference member didn't have audio for some >>>>> time and then begins to speak, the first syllable will be clipped and >>>>> cannot be heard by the other conference participants. >>>>> >>>>> We ruled out a number of possible causes, and finally discovered that >>>>> the energy level detection algorith seems to be the source of the problem. >>>>> Setting the energy level for all participants to "0" heals the problem (of >>>>> course now the noise might be higher since every participant's noise >>>>> contributes to the total noise), but this setting is not very useful since >>>>> the "talking detection" seems to rely on the same level, i.e. with energy >>>>> "0", all members are marked talking all the time (which effectively renders >>>>> the "talking" display useless). >>>>> >>>>> Is there any chance to tune the energy level detection time (!), or to >>>>> de-couple talking detection an energy level detection in FreeSwitch? >>>>> >>>>> Best regards, >>>>> Roland >>>>> >>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Gru?, > Roland > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/f76b8cee/attachment.html From kris at kriskinc.com Fri Nov 4 00:29:32 2011 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 3 Nov 2011 17:29:32 -0400 Subject: [Freeswitch-users] Newbie In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/SCCP On Thu, Nov 3, 2011 at 4:15 PM, wrote: > I am new to FreeSWITCH, testing to make sure it has all the functionality we > need, so I have a few questions I would like to ask. I am coming from using > Asterisk PBX and some Cisco Call Manager.? As you see my question below you > will realize we are currently using SCCP (skinny) based phones, I am trying > to talk out customer into moving to SIP phones but no luck yet. > > > > 1)????? Is there good SCCP (skinny) support? > > 2)????? If so, which phones (79XX)? > > 3)????? Is there Asterisk AMI type messaging to control SCCP phones? > > > > Jacob Miles > > Software Engineer III > > L-3 Communications - Integrated Systems Greenville > > Jacob.E.Miles at L-3Com.com > > 903.457.4422 > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From avi at avimarcus.net Fri Nov 4 00:51:33 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 3 Nov 2011 23:51:33 +0200 Subject: [Freeswitch-users] securing sip channel In-Reply-To: <4EB2E5F7.3090001@googlemail.com> References: <4EB2E5F7.3090001@googlemail.com> Message-ID: You can use static IP instead of the password, or hard code it in addition to the password. But what I think you really want for the registration is "SIPS" via TLS/SSL: http://wiki.freeswitch.org/wiki/Tls Actually, this is only to secure it against a registration being sniffed... which already uses a hash to not send the password in plain text (but can still be stolen via MITM, e.g. on public wifi). Do you just want fail2ban to prevent brute forcing the passwords? http://wiki.freeswitch.org/wiki/Fail2ban -Avi On Thu, Nov 3, 2011 at 9:05 PM, Tamer Higazi wrote: > Hi! > I am interested to know if there is a way to secure the registration > against account hacking. > > for example, I would create an SIP Account server side, there would be > the possibility that only users with the static IP-Address could login > to the assigned account. Is there one other way like kerberos ticket > authentication or public secret key with ipsec?! > > > For any advise I would thank you > > > > Tamer > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/cfd74b8b/attachment.html From roland at haenel.me Fri Nov 4 00:56:30 2011 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Thu, 3 Nov 2011 22:56:30 +0100 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: Yes, I am on a current GIT (about 2 weeks old), and I have waste-bandwidth for every subscriber, and a packet capture confirms that FS sends RTP permanently. So jitter buffers shouldn't be an issue here. Even *if* jitter buffers would be recharging, that would only lead to a small delay, but not to audio being clipped (i.e., lost), right? My packet traces show that there are different ptime's involved (FreeSwitch is on 20ms by default, some phones have 30ms). Could that contribute to the issue? Roland 2011/11/3 Anthony Minessale > it's not, > > most likely you have a jitterbuffer on your phone that is recharging when > you start talking. > are you on the latest GIT where we have the waste bandwidth flag > on permanently for all callers? > To avoid problems like this conferences now always send rtp to people even > when they are not getting any useful input. > > > 2011/11/3 Roland H?nel > >> Anthony, >> >> The problem is not that audio is clipped because its "too silent", so >> adjusting the value seems to be the wrong handle for this problem. >> >> It seems that the time needed to make the decision (regardless of the >> energy threshold value) is the core of the issue, i.e. the level may bei >> 100, 200 or 300, it takes some XX ms to detect talking and all audio that >> happens before the detection gets lost. Right? >> >> Greetings, >> Roland >> >> >> >> 2011/11/3 Anthony Minessale >> >>> The setting is an arbitrary number of the energy score required to count >>> as talking. >>> >>> >>> check your environments, its very specific to the surroundings. >>> We run it at 200 in the main FS conf and its rarely an issue. >>> >>> >>> >>> >>> 2011/11/3 Roland H?nel >>> >>>> Setting it to 10 or so doesn't help, it seems that just the threshold >>>> of detection is changed (ok, that's what is expected), not the time until >>>> the decision is made. >>>> >>>> The audio clipping is a timing issue. >>>> Does there exist something like a "minimum time" that this algorithm >>>> needs to detect talking after silence? What happens to the audio that was >>>> sent by that member during this time, does.the algorithm discard that? >>>> >>>> Greetings, >>>> Roland >>>> Am 03.11.2011 18:20 schrieb "Anthony Minessale" < >>>> anthony.minessale at gmail.com>: >>>> >>>> yes? set it to something between what it was and 0 ? >>>>> >>>>> >>>>> >>>>> try 200 100 50 ? >>>>> >>>>> 2011/11/3 Roland H?nel >>>>> >>>>>> Hi, >>>>>> >>>>>> During thorough testing of our conference solution which is based on >>>>>> FreeSwitch / mod_conference, we noticed some 'audio clipping' during the >>>>>> conference. Typically, if a conference member didn't have audio for some >>>>>> time and then begins to speak, the first syllable will be clipped and >>>>>> cannot be heard by the other conference participants. >>>>>> >>>>>> We ruled out a number of possible causes, and finally discovered that >>>>>> the energy level detection algorith seems to be the source of the problem. >>>>>> Setting the energy level for all participants to "0" heals the problem (of >>>>>> course now the noise might be higher since every participant's noise >>>>>> contributes to the total noise), but this setting is not very useful since >>>>>> the "talking detection" seems to rely on the same level, i.e. with energy >>>>>> "0", all members are marked talking all the time (which effectively renders >>>>>> the "talking" display useless). >>>>>> >>>>>> Is there any chance to tune the energy level detection time (!), or >>>>>> to de-couple talking detection an energy level detection in FreeSwitch? >>>>>> >>>>>> Best regards, >>>>>> Roland >>>>>> >>>>>> >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:+19193869900 >>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Gru?, >> Roland >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Gru?, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/f6a69b34/attachment-0001.html From anthony.minessale at gmail.com Fri Nov 4 01:05:30 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Nov 2011 17:05:30 -0500 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: well waste bandwidth became obsolete on oct 4th You could test your theory by using a 30ms/20ms phone instead. You should really remove variables until you come up with something. Basically the default FS conf on sip:888 at conference.freeswitch.org is running the default params with 200 as default energy. So call that first and see if you get the same issue because we have 30+ people call there every wednesday without incident and we literally spend like 12 hours a day on the same server for private development. Then setup a vanilla out of the box FS build and try exten 3000 The conference. and see if you get your problem, try other phones etc. look at all the variables to see what you may be doing differently than most people who do not experience your issue. 2011/11/3 Roland H?nel > Yes, I am on a current GIT (about 2 weeks old), and I have waste-bandwidth > for every subscriber, and a packet capture confirms that FS sends RTP > permanently. So jitter buffers shouldn't be an issue here. > > Even *if* jitter buffers would be recharging, that would only lead to a > small delay, but not to audio being clipped (i.e., lost), right? > > My packet traces show that there are different ptime's involved > (FreeSwitch is on 20ms by default, some phones have 30ms). Could that > contribute to the issue? > > Roland > > > 2011/11/3 Anthony Minessale > >> it's not, >> >> most likely you have a jitterbuffer on your phone that is recharging when >> you start talking. >> are you on the latest GIT where we have the waste bandwidth flag >> on permanently for all callers? >> To avoid problems like this conferences now always send rtp to people >> even when they are not getting any useful input. >> >> >> 2011/11/3 Roland H?nel >> >>> Anthony, >>> >>> The problem is not that audio is clipped because its "too silent", so >>> adjusting the value seems to be the wrong handle for this problem. >>> >>> It seems that the time needed to make the decision (regardless of the >>> energy threshold value) is the core of the issue, i.e. the level may bei >>> 100, 200 or 300, it takes some XX ms to detect talking and all audio that >>> happens before the detection gets lost. Right? >>> >>> Greetings, >>> Roland >>> >>> >>> >>> 2011/11/3 Anthony Minessale >>> >>>> The setting is an arbitrary number of the energy score required to >>>> count as talking. >>>> >>>> >>>> check your environments, its very specific to the surroundings. >>>> We run it at 200 in the main FS conf and its rarely an issue. >>>> >>>> >>>> >>>> >>>> 2011/11/3 Roland H?nel >>>> >>>>> Setting it to 10 or so doesn't help, it seems that just the threshold >>>>> of detection is changed (ok, that's what is expected), not the time until >>>>> the decision is made. >>>>> >>>>> The audio clipping is a timing issue. >>>>> Does there exist something like a "minimum time" that this algorithm >>>>> needs to detect talking after silence? What happens to the audio that was >>>>> sent by that member during this time, does.the algorithm discard that? >>>>> >>>>> Greetings, >>>>> Roland >>>>> Am 03.11.2011 18:20 schrieb "Anthony Minessale" < >>>>> anthony.minessale at gmail.com>: >>>>> >>>>> yes? set it to something between what it was and 0 ? >>>>>> >>>>>> >>>>>> >>>>>> try 200 100 50 ? >>>>>> >>>>>> 2011/11/3 Roland H?nel >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> During thorough testing of our conference solution which is based on >>>>>>> FreeSwitch / mod_conference, we noticed some 'audio clipping' during the >>>>>>> conference. Typically, if a conference member didn't have audio for some >>>>>>> time and then begins to speak, the first syllable will be clipped and >>>>>>> cannot be heard by the other conference participants. >>>>>>> >>>>>>> We ruled out a number of possible causes, and finally discovered >>>>>>> that the energy level detection algorith seems to be the source of the >>>>>>> problem. Setting the energy level for all participants to "0" heals the >>>>>>> problem (of course now the noise might be higher since every participant's >>>>>>> noise contributes to the total noise), but this setting is not very useful >>>>>>> since the "talking detection" seems to rely on the same level, i.e. with >>>>>>> energy "0", all members are marked talking all the time (which effectively >>>>>>> renders the "talking" display useless). >>>>>>> >>>>>>> Is there any chance to tune the energy level detection time (!), or >>>>>>> to de-couple talking detection an energy level detection in FreeSwitch? >>>>>>> >>>>>>> Best regards, >>>>>>> Roland >>>>>>> >>>>>>> >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> IRC: irc.freenode.net #freeswitch >>>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> pstn:+19193869900 >>>>>> >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Gru?, >>> Roland >>> >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Gru?, > Roland > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/4668ac1b/attachment.html From covici at ccs.covici.com Fri Nov 4 01:08:45 2011 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 03 Nov 2011 18:08:45 -0400 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: Message-ID: <27795.1320358125@ccs.covici.com> I will have to check that out, although I am pretty sure I did test, but that was when I first started. Anthony Minessale wrote: > it's not, > > most likely you have a jitterbuffer on your phone that is recharging when > you start talking. > are you on the latest GIT where we have the waste bandwidth flag > on permanently for all callers? > To avoid problems like this conferences now always send rtp to people even > when they are not getting any useful input. > > > 2011/11/3 Roland H?nel > > > Anthony, > > > > The problem is not that audio is clipped because its "too silent", so > > adjusting the value seems to be the wrong handle for this problem. > > > > It seems that the time needed to make the decision (regardless of the > > energy threshold value) is the core of the issue, i.e. the level may bei > > 100, 200 or 300, it takes some XX ms to detect talking and all audio that > > happens before the detection gets lost. Right? > > > > Greetings, > > Roland > > > > > > > > 2011/11/3 Anthony Minessale > > > >> The setting is an arbitrary number of the energy score required to count > >> as talking. > >> > >> > >> check your environments, its very specific to the surroundings. > >> We run it at 200 in the main FS conf and its rarely an issue. > >> > >> > >> > >> > >> 2011/11/3 Roland H?nel > >> > >>> Setting it to 10 or so doesn't help, it seems that just the threshold of > >>> detection is changed (ok, that's what is expected), not the time until the > >>> decision is made. > >>> > >>> The audio clipping is a timing issue. > >>> Does there exist something like a "minimum time" that this algorithm > >>> needs to detect talking after silence? What happens to the audio that was > >>> sent by that member during this time, does.the algorithm discard that? > >>> > >>> Greetings, > >>> Roland > >>> Am 03.11.2011 18:20 schrieb "Anthony Minessale" < > >>> anthony.minessale at gmail.com>: > >>> > >>> yes? set it to something between what it was and 0 ? > >>>> > >>>> > >>>> > >>>> try 200 100 50 ? > >>>> > >>>> 2011/11/3 Roland H?nel > >>>> > >>>>> Hi, > >>>>> > >>>>> During thorough testing of our conference solution which is based on > >>>>> FreeSwitch / mod_conference, we noticed some 'audio clipping' during the > >>>>> conference. Typically, if a conference member didn't have audio for some > >>>>> time and then begins to speak, the first syllable will be clipped and > >>>>> cannot be heard by the other conference participants. > >>>>> > >>>>> We ruled out a number of possible causes, and finally discovered that > >>>>> the energy level detection algorith seems to be the source of the problem. > >>>>> Setting the energy level for all participants to "0" heals the problem (of > >>>>> course now the noise might be higher since every participant's noise > >>>>> contributes to the total noise), but this setting is not very useful since > >>>>> the "talking detection" seems to rely on the same level, i.e. with energy > >>>>> "0", all members are marked talking all the time (which effectively renders > >>>>> the "talking" display useless). > >>>>> > >>>>> Is there any chance to tune the energy level detection time (!), or to > >>>>> de-couple talking detection an energy level detection in FreeSwitch? > >>>>> > >>>>> Best regards, > >>>>> Roland > >>>>> > >>>>> > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE: > >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> > >>>> -- > >>>> Anthony Minessale II > >>>> > >>>> FreeSWITCH http://www.freeswitch.org/ > >>>> ClueCon http://www.cluecon.com/ > >>>> Twitter: http://twitter.com/FreeSWITCH_wire > >>>> > >>>> AIM: anthm > >>>> MSN:anthony_minessale at hotmail.com > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >>>> IRC: irc.freenode.net #freeswitch > >>>> > >>>> FreeSWITCH Developer Conference > >>>> sip:888 at conference.freeswitch.org > >>>> googletalk:conf+888 at conference.freeswitch.org > >>>> pstn:+19193869900 > >>>> > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > >> googletalk:conf+888 at conference.freeswitch.org > >> pstn:+19193869900 > >> > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > -- > > Gru?, > > Roland > > > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From lloyd.aloysius at gmail.com Fri Nov 4 01:09:57 2011 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Thu, 3 Nov 2011 18:09:57 -0400 Subject: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV In-Reply-To: References: Message-ID: Evan ... Thank you very much. Yes remove the port number solve my registration. Even polycom tech support could not help to solve the problem last two days. -- Vik , I trust your statement. Thank you for the help. I did not think port number is making this much of problem. Yes .. your configuration do not have port number and it is correct. Thanks again. Thanks and regards, Lloyd On Wed, Nov 2, 2011 at 8:11 PM, Evan P. Hall wrote: > It will not look up the NAPTR or SRV records at all since you have > specified port 5060. Don?t specify the port number and it should work. * > *** > > ** ** > > We run a large number of polycom phones without NAPTR records here. > However we don?t use the outboundProxy attributes at all. We use > voIpProt.server.x.address or the phone?s reg.x.server.x.address to tell the > phones where to register. You shouldn?t need to specify outboundProxy > unless you need your signaling packets to go to a different server than the > registration.**** > > ** ** > > -Evan**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Lloyd > Aloysius > *Sent:* Tuesday, November 01, 2011 6:23 AM > *To:* FreeSWITCH Users Help > > *Subject:* Re: [Freeswitch-users] FreeSWITCH - Polycom and DNS SRV**** > > ** ** > > Brian,**** > > ** ** > > Thank you for the reply. I set the voIpProt.SIP.transport="DNSnaptr"**** > > ** ** > > ** ** > > **** > > > voIpProt.SIP.outboundProxy.address="sip.mydomain.com"**** > > voIpProt.SIP.transport="DNSnaptr"**** > > voIpProt.SIP.outboundProxy.port="5060 />**** > > **** > > ** ** > > But this fails to register.**** > > > Thanks**** > > Lloyd**** > > ** ** > > On Tue, Nov 1, 2011 at 9:11 AM, Brian West wrote:** > ** > > I'm going to guess you have not set voIpProt.server.x.transport="DNSnaptr" > and you're missing NAPTR records too.**** > > ** ** > > /b**** > > ** ** > > On Nov 1, 2011, at 7:48 AM, Lloyd Aloysius wrote:**** > > ** ** > > Hi , > > I am testing the Polycom Phones [IP335,IP450 and IP6000] and DNS SRV > records. All my DNS SRV records and other phone models [Cisico , Asstra, > Yealink] working without any issue. > > But for Polycom Phones .. I need to specify the "Outbound Proxy" as IP > address otherwise phone never register.**** > > * voIpProt.SIP.outboundProxy.address="IP ADDRESS"***** > > > > I could not find why we need to specify "Outbound Proxy? How to avoid > using IP address? > > Does Polycom phones fully support DNS SRV records? > > Any help is appreciated. > > > > Thanks and regards, > > Lloyd**** > > ** ** > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/2de71362/attachment.html From anthony.minessale at gmail.com Fri Nov 4 01:10:33 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Nov 2011 17:10:33 -0500 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: <27795.1320358125@ccs.covici.com> References: <27795.1320358125@ccs.covici.com> Message-ID: well you call 888 every week and its the same software =p On Thu, Nov 3, 2011 at 5:08 PM, wrote: > I will have to check that out, although I am pretty sure I did test, but > that was when I first started. > > Anthony Minessale wrote: > > > it's not, > > > > most likely you have a jitterbuffer on your phone that is recharging when > > you start talking. > > are you on the latest GIT where we have the waste bandwidth flag > > on permanently for all callers? > > To avoid problems like this conferences now always send rtp to people > even > > when they are not getting any useful input. > > > > > > 2011/11/3 Roland H?nel > > > > > Anthony, > > > > > > The problem is not that audio is clipped because its "too silent", so > > > adjusting the value seems to be the wrong handle for this problem. > > > > > > It seems that the time needed to make the decision (regardless of the > > > energy threshold value) is the core of the issue, i.e. the level may > bei > > > 100, 200 or 300, it takes some XX ms to detect talking and all audio > that > > > happens before the detection gets lost. Right? > > > > > > Greetings, > > > Roland > > > > > > > > > > > > 2011/11/3 Anthony Minessale > > > > > >> The setting is an arbitrary number of the energy score required to > count > > >> as talking. > > >> > > >> > > >> check your environments, its very specific to the surroundings. > > >> We run it at 200 in the main FS conf and its rarely an issue. > > >> > > >> > > >> > > >> > > >> 2011/11/3 Roland H?nel > > >> > > >>> Setting it to 10 or so doesn't help, it seems that just the > threshold of > > >>> detection is changed (ok, that's what is expected), not the time > until the > > >>> decision is made. > > >>> > > >>> The audio clipping is a timing issue. > > >>> Does there exist something like a "minimum time" that this algorithm > > >>> needs to detect talking after silence? What happens to the audio > that was > > >>> sent by that member during this time, does.the algorithm discard > that? > > >>> > > >>> Greetings, > > >>> Roland > > >>> Am 03.11.2011 18:20 schrieb "Anthony Minessale" < > > >>> anthony.minessale at gmail.com>: > > >>> > > >>> yes? set it to something between what it was and 0 ? > > >>>> > > >>>> > > >>>> > > >>>> try 200 100 50 ? > > >>>> > > >>>> 2011/11/3 Roland H?nel > > >>>> > > >>>>> Hi, > > >>>>> > > >>>>> During thorough testing of our conference solution which is based > on > > >>>>> FreeSwitch / mod_conference, we noticed some 'audio clipping' > during the > > >>>>> conference. Typically, if a conference member didn't have audio > for some > > >>>>> time and then begins to speak, the first syllable will be clipped > and > > >>>>> cannot be heard by the other conference participants. > > >>>>> > > >>>>> We ruled out a number of possible causes, and finally discovered > that > > >>>>> the energy level detection algorith seems to be the source of the > problem. > > >>>>> Setting the energy level for all participants to "0" heals the > problem (of > > >>>>> course now the noise might be higher since every participant's > noise > > >>>>> contributes to the total noise), but this setting is not very > useful since > > >>>>> the "talking detection" seems to rely on the same level, i.e. with > energy > > >>>>> "0", all members are marked talking all the time (which > effectively renders > > >>>>> the "talking" display useless). > > >>>>> > > >>>>> Is there any chance to tune the energy level detection time (!), > or to > > >>>>> de-couple talking detection an energy level detection in > FreeSwitch? > > >>>>> > > >>>>> Best regards, > > >>>>> Roland > > >>>>> > > >>>>> > > >>>>> > > >>>>> FreeSWITCH-users mailing list > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>>>> UNSUBSCRIBE: > > >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>>>> http://www.freeswitch.org > > >>>>> > > >>>>> > > >>>> > > >>>> > > >>>> -- > > >>>> Anthony Minessale II > > >>>> > > >>>> FreeSWITCH http://www.freeswitch.org/ > > >>>> ClueCon http://www.cluecon.com/ > > >>>> Twitter: http://twitter.com/FreeSWITCH_wire > > >>>> > > >>>> AIM: anthm > > >>>> MSN:anthony_minessale at hotmail.com > > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >>>> IRC: irc.freenode.net #freeswitch > > >>>> > > >>>> FreeSWITCH Developer Conference > > >>>> sip:888 at conference.freeswitch.org > > >>>> googletalk:conf+888 at conference.freeswitch.org > > >>>> pstn:+19193869900 > > >>>> > > >>>> > > >>>> FreeSWITCH-users mailing list > > >>>> FreeSWITCH-users at lists.freeswitch.org > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>>> UNSUBSCRIBE: > > >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>>> http://www.freeswitch.org > > >>>> > > >>>> > > >>> > > >>> FreeSWITCH-users mailing list > > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> http://www.freeswitch.org > > >>> > > >>> > > >> > > >> > > >> -- > > >> Anthony Minessale II > > >> > > >> FreeSWITCH http://www.freeswitch.org/ > > >> ClueCon http://www.cluecon.com/ > > >> Twitter: http://twitter.com/FreeSWITCH_wire > > >> > > >> AIM: anthm > > >> MSN:anthony_minessale at hotmail.com > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >> IRC: irc.freenode.net #freeswitch > > >> > > >> FreeSWITCH Developer Conference > > >> sip:888 at conference.freeswitch.org > > >> googletalk:conf+888 at conference.freeswitch.org > > >> pstn:+19193869900 > > >> > > >> > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > >> > > > > > > > > > -- > > > Gru?, > > > Roland > > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > ---------------------------------------------------- > > Alternatives: > > > > ---------------------------------------------------- > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/0f30987f/attachment-0001.html From covici at ccs.covici.com Fri Nov 4 02:06:10 2011 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 03 Nov 2011 19:06:10 -0400 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: References: <27795.1320358125@ccs.covici.com> Message-ID: <3018.1320361570@ccs.covici.com> Does auto-record make a difference -- I know you don't use that feature? Anthony Minessale wrote: > well you call 888 every week and its the same software =p > > > On Thu, Nov 3, 2011 at 5:08 PM, wrote: > > > I will have to check that out, although I am pretty sure I did test, but > > that was when I first started. > > > > Anthony Minessale wrote: > > > > > it's not, > > > > > > most likely you have a jitterbuffer on your phone that is recharging when > > > you start talking. > > > are you on the latest GIT where we have the waste bandwidth flag > > > on permanently for all callers? > > > To avoid problems like this conferences now always send rtp to people > > even > > > when they are not getting any useful input. > > > > > > > > > 2011/11/3 Roland H?nel > > > > > > > Anthony, > > > > > > > > The problem is not that audio is clipped because its "too silent", so > > > > adjusting the value seems to be the wrong handle for this problem. > > > > > > > > It seems that the time needed to make the decision (regardless of the > > > > energy threshold value) is the core of the issue, i.e. the level may > > bei > > > > 100, 200 or 300, it takes some XX ms to detect talking and all audio > > that > > > > happens before the detection gets lost. Right? > > > > > > > > Greetings, > > > > Roland > > > > > > > > > > > > > > > > 2011/11/3 Anthony Minessale > > > > > > > >> The setting is an arbitrary number of the energy score required to > > count > > > >> as talking. > > > >> > > > >> > > > >> check your environments, its very specific to the surroundings. > > > >> We run it at 200 in the main FS conf and its rarely an issue. > > > >> > > > >> > > > >> > > > >> > > > >> 2011/11/3 Roland H?nel > > > >> > > > >>> Setting it to 10 or so doesn't help, it seems that just the > > threshold of > > > >>> detection is changed (ok, that's what is expected), not the time > > until the > > > >>> decision is made. > > > >>> > > > >>> The audio clipping is a timing issue. > > > >>> Does there exist something like a "minimum time" that this algorithm > > > >>> needs to detect talking after silence? What happens to the audio > > that was > > > >>> sent by that member during this time, does.the algorithm discard > > that? > > > >>> > > > >>> Greetings, > > > >>> Roland > > > >>> Am 03.11.2011 18:20 schrieb "Anthony Minessale" < > > > >>> anthony.minessale at gmail.com>: > > > >>> > > > >>> yes? set it to something between what it was and 0 ? > > > >>>> > > > >>>> > > > >>>> > > > >>>> try 200 100 50 ? > > > >>>> > > > >>>> 2011/11/3 Roland H?nel > > > >>>> > > > >>>>> Hi, > > > >>>>> > > > >>>>> During thorough testing of our conference solution which is based > > on > > > >>>>> FreeSwitch / mod_conference, we noticed some 'audio clipping' > > during the > > > >>>>> conference. Typically, if a conference member didn't have audio > > for some > > > >>>>> time and then begins to speak, the first syllable will be clipped > > and > > > >>>>> cannot be heard by the other conference participants. > > > >>>>> > > > >>>>> We ruled out a number of possible causes, and finally discovered > > that > > > >>>>> the energy level detection algorith seems to be the source of the > > problem. > > > >>>>> Setting the energy level for all participants to "0" heals the > > problem (of > > > >>>>> course now the noise might be higher since every participant's > > noise > > > >>>>> contributes to the total noise), but this setting is not very > > useful since > > > >>>>> the "talking detection" seems to rely on the same level, i.e. with > > energy > > > >>>>> "0", all members are marked talking all the time (which > > effectively renders > > > >>>>> the "talking" display useless). > > > >>>>> > > > >>>>> Is there any chance to tune the energy level detection time (!), > > or to > > > >>>>> de-couple talking detection an energy level detection in > > FreeSwitch? > > > >>>>> > > > >>>>> Best regards, > > > >>>>> Roland > > > >>>>> > > > >>>>> > > > >>>>> > > > >>>>> FreeSWITCH-users mailing list > > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >>>>> UNSUBSCRIBE: > > > >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>>> http://www.freeswitch.org > > > >>>>> > > > >>>>> > > > >>>> > > > >>>> > > > >>>> -- > > > >>>> Anthony Minessale II > > > >>>> > > > >>>> FreeSWITCH http://www.freeswitch.org/ > > > >>>> ClueCon http://www.cluecon.com/ > > > >>>> Twitter: http://twitter.com/FreeSWITCH_wire > > > >>>> > > > >>>> AIM: anthm > > > >>>> MSN:anthony_minessale at hotmail.com > > > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > >>>> IRC: irc.freenode.net #freeswitch > > > >>>> > > > >>>> FreeSWITCH Developer Conference > > > >>>> sip:888 at conference.freeswitch.org > > > >>>> googletalk:conf+888 at conference.freeswitch.org > > > >>>> pstn:+19193869900 > > > >>>> > > > >>>> > > > >>>> FreeSWITCH-users mailing list > > > >>>> FreeSWITCH-users at lists.freeswitch.org > > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >>>> UNSUBSCRIBE: > > > >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> http://www.freeswitch.org > > > >>>> > > > >>>> > > > >>> > > > >>> FreeSWITCH-users mailing list > > > >>> FreeSWITCH-users at lists.freeswitch.org > > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >>> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>> http://www.freeswitch.org > > > >>> > > > >>> > > > >> > > > >> > > > >> -- > > > >> Anthony Minessale II > > > >> > > > >> FreeSWITCH http://www.freeswitch.org/ > > > >> ClueCon http://www.cluecon.com/ > > > >> Twitter: http://twitter.com/FreeSWITCH_wire > > > >> > > > >> AIM: anthm > > > >> MSN:anthony_minessale at hotmail.com > > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > >> IRC: irc.freenode.net #freeswitch > > > >> > > > >> FreeSWITCH Developer Conference > > > >> sip:888 at conference.freeswitch.org > > > >> googletalk:conf+888 at conference.freeswitch.org > > > >> pstn:+19193869900 > > > >> > > > >> > > > >> FreeSWITCH-users mailing list > > > >> FreeSWITCH-users at lists.freeswitch.org > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >> http://www.freeswitch.org > > > >> > > > >> > > > > > > > > > > > > -- > > > > Gru?, > > > > Roland > > > > > > > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:+19193869900 > > > > > > ---------------------------------------------------- > > > Alternatives: > > > > > > ---------------------------------------------------- > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Your life is like a penny. You're going to lose it. The question is: > > How do > > you spend it? > > > > John Covici > > covici at ccs.covici.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From covici at ccs.covici.com Fri Nov 4 02:10:54 2011 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 03 Nov 2011 19:10:54 -0400 Subject: [Freeswitch-users] Audio clipping in mod_conference In-Reply-To: <3018.1320361570@ccs.covici.com> References: <27795.1320358125@ccs.covici.com> <3018.1320361570@ccs.covici.com> Message-ID: <3712.1320361854@ccs.covici.com> In my case if I set the energy level to > 0, people can hear each other fine, but there is clipping on the recording. covici at ccs.covici.com wrote: > Does auto-record make a difference -- I know you don't use that feature? > > Anthony Minessale wrote: > > > well you call 888 every week and its the same software =p > > > > > > On Thu, Nov 3, 2011 at 5:08 PM, wrote: > > > > > I will have to check that out, although I am pretty sure I did test, but > > > that was when I first started. > > > > > > Anthony Minessale wrote: > > > > > > > it's not, > > > > > > > > most likely you have a jitterbuffer on your phone that is recharging when > > > > you start talking. > > > > are you on the latest GIT where we have the waste bandwidth flag > > > > on permanently for all callers? > > > > To avoid problems like this conferences now always send rtp to people > > > even > > > > when they are not getting any useful input. > > > > > > > > > > > > 2011/11/3 Roland H?nel > > > > > > > > > Anthony, > > > > > > > > > > The problem is not that audio is clipped because its "too silent", so > > > > > adjusting the value seems to be the wrong handle for this problem. > > > > > > > > > > It seems that the time needed to make the decision (regardless of the > > > > > energy threshold value) is the core of the issue, i.e. the level may > > > bei > > > > > 100, 200 or 300, it takes some XX ms to detect talking and all audio > > > that > > > > > happens before the detection gets lost. Right? > > > > > > > > > > Greetings, > > > > > Roland > > > > > > > > > > > > > > > > > > > > 2011/11/3 Anthony Minessale > > > > > > > > > >> The setting is an arbitrary number of the energy score required to > > > count > > > > >> as talking. > > > > >> > > > > >> > > > > >> check your environments, its very specific to the surroundings. > > > > >> We run it at 200 in the main FS conf and its rarely an issue. > > > > >> > > > > >> > > > > >> > > > > >> > > > > >> 2011/11/3 Roland H?nel > > > > >> > > > > >>> Setting it to 10 or so doesn't help, it seems that just the > > > threshold of > > > > >>> detection is changed (ok, that's what is expected), not the time > > > until the > > > > >>> decision is made. > > > > >>> > > > > >>> The audio clipping is a timing issue. > > > > >>> Does there exist something like a "minimum time" that this algorithm > > > > >>> needs to detect talking after silence? What happens to the audio > > > that was > > > > >>> sent by that member during this time, does.the algorithm discard > > > that? > > > > >>> > > > > >>> Greetings, > > > > >>> Roland > > > > >>> Am 03.11.2011 18:20 schrieb "Anthony Minessale" < > > > > >>> anthony.minessale at gmail.com>: > > > > >>> > > > > >>> yes? set it to something between what it was and 0 ? > > > > >>>> > > > > >>>> > > > > >>>> > > > > >>>> try 200 100 50 ? > > > > >>>> > > > > >>>> 2011/11/3 Roland H?nel > > > > >>>> > > > > >>>>> Hi, > > > > >>>>> > > > > >>>>> During thorough testing of our conference solution which is based > > > on > > > > >>>>> FreeSwitch / mod_conference, we noticed some 'audio clipping' > > > during the > > > > >>>>> conference. Typically, if a conference member didn't have audio > > > for some > > > > >>>>> time and then begins to speak, the first syllable will be clipped > > > and > > > > >>>>> cannot be heard by the other conference participants. > > > > >>>>> > > > > >>>>> We ruled out a number of possible causes, and finally discovered > > > that > > > > >>>>> the energy level detection algorith seems to be the source of the > > > problem. > > > > >>>>> Setting the energy level for all participants to "0" heals the > > > problem (of > > > > >>>>> course now the noise might be higher since every participant's > > > noise > > > > >>>>> contributes to the total noise), but this setting is not very > > > useful since > > > > >>>>> the "talking detection" seems to rely on the same level, i.e. with > > > energy > > > > >>>>> "0", all members are marked talking all the time (which > > > effectively renders > > > > >>>>> the "talking" display useless). > > > > >>>>> > > > > >>>>> Is there any chance to tune the energy level detection time (!), > > > or to > > > > >>>>> de-couple talking detection an energy level detection in > > > FreeSwitch? > > > > >>>>> > > > > >>>>> Best regards, > > > > >>>>> Roland > > > > >>>>> > > > > >>>>> > > > > >>>>> > > > > >>>>> FreeSWITCH-users mailing list > > > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > >>>>> UNSUBSCRIBE: > > > > >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > >>>>> http://www.freeswitch.org > > > > >>>>> > > > > >>>>> > > > > >>>> > > > > >>>> > > > > >>>> -- > > > > >>>> Anthony Minessale II > > > > >>>> > > > > >>>> FreeSWITCH http://www.freeswitch.org/ > > > > >>>> ClueCon http://www.cluecon.com/ > > > > >>>> Twitter: http://twitter.com/FreeSWITCH_wire > > > > >>>> > > > > >>>> AIM: anthm > > > > >>>> MSN:anthony_minessale at hotmail.com > > > > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >>>> IRC: irc.freenode.net #freeswitch > > > > >>>> > > > > >>>> FreeSWITCH Developer Conference > > > > >>>> sip:888 at conference.freeswitch.org > > > > >>>> googletalk:conf+888 at conference.freeswitch.org > > > > >>>> pstn:+19193869900 > > > > >>>> > > > > >>>> > > > > >>>> FreeSWITCH-users mailing list > > > > >>>> FreeSWITCH-users at lists.freeswitch.org > > > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > >>>> UNSUBSCRIBE: > > > > >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > >>>> http://www.freeswitch.org > > > > >>>> > > > > >>>> > > > > >>> > > > > >>> FreeSWITCH-users mailing list > > > > >>> FreeSWITCH-users at lists.freeswitch.org > > > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > >>> UNSUBSCRIBE: > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > >>> http://www.freeswitch.org > > > > >>> > > > > >>> > > > > >> > > > > >> > > > > >> -- > > > > >> Anthony Minessale II > > > > >> > > > > >> FreeSWITCH http://www.freeswitch.org/ > > > > >> ClueCon http://www.cluecon.com/ > > > > >> Twitter: http://twitter.com/FreeSWITCH_wire > > > > >> > > > > >> AIM: anthm > > > > >> MSN:anthony_minessale at hotmail.com > > > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >> IRC: irc.freenode.net #freeswitch > > > > >> > > > > >> FreeSWITCH Developer Conference > > > > >> sip:888 at conference.freeswitch.org > > > > >> googletalk:conf+888 at conference.freeswitch.org > > > > >> pstn:+19193869900 > > > > >> > > > > >> > > > > >> FreeSWITCH-users mailing list > > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > >> UNSUBSCRIBE: > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > >> http://www.freeswitch.org > > > > >> > > > > >> > > > > > > > > > > > > > > > -- > > > > > Gru?, > > > > > Roland > > > > > > > > > > > > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > -- > > > > Anthony Minessale II > > > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > > ClueCon http://www.cluecon.com/ > > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > > > AIM: anthm > > > > MSN:anthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > IRC: irc.freenode.net #freeswitch > > > > > > > > FreeSWITCH Developer Conference > > > > sip:888 at conference.freeswitch.org > > > > googletalk:conf+888 at conference.freeswitch.org > > > > pstn:+19193869900 > > > > > > > > ---------------------------------------------------- > > > > Alternatives: > > > > > > > > ---------------------------------------------------- > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > -- > > > Your life is like a penny. You're going to lose it. The question is: > > > How do > > > you spend it? > > > > > > John Covici > > > covici at ccs.covici.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 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > ---------------------------------------------------- > > Alternatives: > > > > ---------------------------------------------------- > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From Jacob.E.Miles at L-3Com.com Fri Nov 4 00:48:25 2011 From: Jacob.E.Miles at L-3Com.com (Jacob.E.Miles at L-3Com.com) Date: Thu, 3 Nov 2011 16:48:25 -0500 Subject: [Freeswitch-users] Newbie In-Reply-To: References: Message-ID: Thanks for the link. My Setup: CentOS 6.0 FreeSWITCH snapshot 1.0.7 2 Cisco 7971 phones 1 Cisco CM Communicator Softphone I have things setup I think. My phones register and show a line number, but I am unable to make any calls. Executing "skinny status profile internal device SEP001122334455" crashes FreeSWITCH. Any help? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: Thursday, November 03, 2011 4:30 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Newbie http://wiki.freeswitch.org/wiki/SCCP On Thu, Nov 3, 2011 at 4:15 PM, wrote: > I am new to FreeSWITCH, testing to make sure it has all the functionality we > need, so I have a few questions I would like to ask. I am coming from using > Asterisk PBX and some Cisco Call Manager.? As you see my question below you > will realize we are currently using SCCP (skinny) based phones, I am trying > to talk out customer into moving to SIP phones but no luck yet. > > > > 1)????? Is there good SCCP (skinny) support? > > 2)????? If so, which phones (79XX)? > > 3)????? Is there Asterisk AMI type messaging to control SCCP phones? > > > > Jacob Miles > > Software Engineer III > > L-3 Communications - Integrated Systems Greenville > > Jacob.E.Miles at L-3Com.com > > 903.457.4422 > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.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 4 03:21:55 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Nov 2011 17:21:55 -0700 Subject: [Freeswitch-users] Help with Using Mod_AVMD In-Reply-To: <87A1ED3B86EA4F8AAB18439700A28B42@clancysystems.com> References: <87A1ED3B86EA4F8AAB18439700A28B42@clancysystems.com> Message-ID: Please note: mod_avmd is *not* answering machine detection. There is a private mod_amd specifically for this purpose. Contact consulting at freeswitch.org if you are interested in that. mod_avmd is designed to detect the "beep" played by voicemail systems and answering machines. If a beep is detected then mod_avmd emits an event to the FreeSWITCH event sub-system. I would attach with fs_cli and listen for the event. Try this at fs_cli: /log 0 /event plain CUSTOM Then make some test calls and see if there is actually a detection. Once you confirm positive detection then you can see what's up with the Lua input callback. -MC On Thu, Nov 3, 2011 at 10:39 AM, Dave wrote: > ** > Hello, > > I am creating an external application that uses FreeSWITCH ESL to initiate > an outbound call and then run a LUA script. Within the LUA script I am > executing AVMD. I want to see if its a human or voicemail on the other > end. The outbound call is made and the LUA script executes. > > Currently I would just like to see a consoleLog entry made by the onInput > function in the LUA script, but that function does not seem to be called. > AVMD starts the file is streamed then AVMD stops. As shown in the log. > > I think I'm missing something. > Dave Goodwin > > > Pertinent FreeSWITCH Log Info: > > ------------------------------------------------------------------------------------------------------- > EXECUTE sofia/internal/7202126254 at xxx.xx.xxx.xx avmd(start) > 2011-11-03 11:05:27.291678 [DEBUG] switch_core_media_bug.c:360 Attaching > BUG to sofia/internal/7202126254 at xxx.xx.xxx.xx > 2011-11-03 11:05:28.311249 [DEBUG] switch_ivr_play_say.c:1279 Codec > Activated L16 at 8000hz 1 channels 20ms > 2011-11-03 11:05:28.350313 [DEBUG] switch_rtp.c:3082 Correct ip/port > confirmed. > 2011-11-03 11:05:40.235535 [DEBUG] switch_ivr_play_say.c:1649 done playing > file > EXECUTE sofia/internal/7202126254 at xxx.xx.xxx.xxavmd(stop) > 2011-11-03 11:05:40.237488 [DEBUG] switch_core_media_bug.c:467 Removing > BUG from sofia/internal/7202126254 at xxx.xx.xxx.xx > > Lua Script: > > ------------------------------------------------------------------------------------------------------- > function onInput(session, type, obj) > if type == "dtmf" and obj['digit'] == '1' and human_detected == false > then > human_detected = true > freeswitch.consoleLog("INFO","Human Detected\n") > return "break" > end > if type == "event" and voicemail_detected == false then > voicemail_detected = true > freeswitch.consoleLog("INFO","Voicemail Detected\n") > return "break" > else > freeswitch.consoleLog("INFO","Nothing Happened\n") > return "break" > end > end > > if session:ready() then > --- the call has been answered. > session:setInputCallback("onInput") > session:sleep(1000) > session:execute("avmd","start") > session:sleep(1000) > session:streamFile("ivr/" .. wav_file) > session:execute("avmd","stop") > session:sleep(1000) > fileName = "Completed " .. tostring(call_from_num) .. os.time() .. > tostring(call_to_num) > freeswitch.consoleLog("INFO","FileName = " .. fileName .. "\n") > local myfile, ErrStr = io.open("log/outcall/" .. fileName,"w") > if myfile then > myfile:write(call_from_num .. "|", call_to_num .. "|" , os.date()) > myfile:flush() > myfile:close() > else > freeswitch.consoleLog("INFO","Error writing File = '" .. ErrStr .. "'\n") > end --if > end --if > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111103/2420349e/attachment-0001.html From hynek.cihlar at gmail.com Fri Nov 4 09:50:28 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Fri, 4 Nov 2011 07:50:28 +0100 Subject: [Freeswitch-users] QoS Message-ID: Hi all, here's an excellent article on Linux QoS http://wiki.linuxwall.info/doku.php/en:ressources:dossiers:networking:traffic_control . Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/60e164b6/attachment.html From miha at softnet.si Fri Nov 4 10:12:33 2011 From: miha at softnet.si (Miha Zoubek) Date: Fri, 04 Nov 2011 08:12:33 +0100 Subject: [Freeswitch-users] h323-conf-id, not found in dictionary Message-ID: <4EB39061.7070909@softnet.si> Hi, please help me with this issue as I realy do not know what should I do next or what I am doing wrong. I have put all configuration files on pastbin. http://pastebin.freeswitch.org/17673 Error that I am getting is: 2011-11-04 07:25:46.967596 [INFO] mod_dptools.c:1316 Before Auth 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not found in dictionary 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:546 An error occured during RADIUS Authentication(RC=-1) 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:702 An error occured during radius authorization. I do not why I get this as I have this attribute defined in my dictionary.cisco :( Please help! Thank you! BR, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/5b361f08/attachment.html From netcentrica at gmail.com Fri Nov 4 11:33:56 2011 From: netcentrica at gmail.com (Mateusz Bartczak) Date: Fri, 4 Nov 2011 09:33:56 +0100 Subject: [Freeswitch-users] How to check if user is behind NAT in a programmatic way? Message-ID: Hello I would like to allow direct p2p media (bypassing FS) between registered SIP users, but only if one of them is not behind NAT. I can't force bypass_media=true for all calls, because many of users are behind NAT and this will break communication. I only want to bypass_media for users where it's possible. Is there any way of checking this in a programmatic way? I've checked call variables, SIP registration info from database but I can't find any related variable. Any suggestions? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/6bcbf025/attachment.html From freeswitch at peely.com Fri Nov 4 12:45:57 2011 From: freeswitch at peely.com (peely) Date: Fri, 4 Nov 2011 02:45:57 -0700 (PDT) Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: References: <1320324355799-6958891.post@n2.nabble.com> Message-ID: <1320399957355-6962201.post@n2.nabble.com> Thanks Anthony, Ah, I mis-understood the requirement for the setting. The Wiki suggests both server MUST have the same name in order for sessions to be resumed on the standby, my boxes have different names to tell them apart but I set switchname to the same value on both boxes because I though I had to. I'll try setting the switchnames to different values and failover to see what happens then. Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6962201.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at peely.com Fri Nov 4 14:06:12 2011 From: freeswitch at peely.com (peely) Date: Fri, 4 Nov 2011 04:06:12 -0700 (PDT) Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: <1320399957355-6962201.post@n2.nabble.com> References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> Message-ID: <1320404772101-6962394.post@n2.nabble.com> OK, that works, keeping the switchnames different doesn't dump the calls table when the other switch starts and issuing a "sofia recover" recovers the calls still on the other box. Can I ask when exactly the "switchname" should be used then? FS 3277: http://jira.freeswitch.org/browse/FS-3277 Suggests it was implemented for the exact reason of keeping the names of two switches in an HA configuration identical, but it seems they are best left separate. I don't mind cleaning up some of the detail in the HA Wiki but would just like to know when the names should be the same and when they should be different i.e. shoudl the names be the same for kind kind of active / active setup? I don't know if Avi Marcus is reading this, maybe you can comment as I know you have a good deal of experience in getting HA setups up and running. Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6962394.html Sent from the freeswitch-users mailing list archive at Nabble.com. From b_ball_henry at hotmail.com Fri Nov 4 14:22:48 2011 From: b_ball_henry at hotmail.com (Henry Huang) Date: Fri, 4 Nov 2011 19:22:48 +0800 Subject: [Freeswitch-users] QoS In-Reply-To: References: Message-ID: Great article, thanks. Henry On Fri, Nov 4, 2011 at 2:50 PM, Hynek Cihlar wrote: > Hi all, here's an excellent article on Linux QoS > http://wiki.linuxwall.info/doku.php/en:ressources:dossiers:networking:traffic_control > . > > Hynek > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/ac58de49/attachment.html From chrisbware at interfree.it Fri Nov 4 16:13:34 2011 From: chrisbware at interfree.it (chrisbware at interfree.it) Date: 4 Nov 2011 13:13:34 -0000 Subject: [Freeswitch-users] Lua time condition Message-ID: <20111104131334.28165.qmail@community1.interfree.it> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/fcfe8b69/attachment.html From garmt.noname at gmail.com Fri Nov 4 16:27:34 2011 From: garmt.noname at gmail.com (grmt) Date: Fri, 4 Nov 2011 14:27:34 +0100 Subject: [Freeswitch-users] Lua time condition In-Reply-To: <20111104131334.28165.qmail@community1.interfree.it> References: <20111104131334.28165.qmail@community1.interfree.it> Message-ID: <009601cc9af5$84061d40$8c1257c0$@gmail.com> The default dialplan shows you a lot of options to handle time dependent call processing, however if you do insist on using lua: before = 900 after = 1800 now = os.date("%H%M") if (after > before) and (now > before) and (now > after) then . End Grmt From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of chrisbware at interfree.it Sent: Friday, November 04, 2011 14:14 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Lua time condition Hi guys, I'm a Lua newbie. I'd like to route an incoming call using a Lua script, according to time. Basically I need to specify a Lua condition like the following: If DATE_TIME1 < current_time < DATE_TIME2 then bridge(A) else bridge(B) I hope some Lua guru can help me! ---------------------------------------------------------------------------- --- Valore legale alle tue mail InterfreePEC - la tua Posta Elettronica Certificata http://pec.interfree.it ---------------------------------------------------------------------------- --- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/45398110/attachment.html From manieq at wp.eu Fri Nov 4 16:39:13 2011 From: manieq at wp.eu (Mariusz Czulada) Date: Fri, 04 Nov 2011 14:39:13 +0100 Subject: [Freeswitch-users] Refer-To does not allow to add 'to-tag' and 'from-tag' Message-ID: <4eb3eb01a11587.03702142@wp.pl> Hi All, I have a problem with my FSW instance (version 1.0.7 on RHER 5.6 64bit). Freeswitch does not allow to pass 'to-tag' or 'from-tag' in addition to "Replaces=" while deflecting a call. This action works: The session I try to deflect is set from Broadworks 16 via ALU IMS core (ISC 5450). Our supplier claims that to successfully handle REFER from FSW, Refer-To header should include also 'to-tag' and 'from-tag'. Example I got looks like this: Refer-To: sip:conf_t0_2 at 10.79.21.108?Replaces=1387BD211B831526 at 172.16.10.114;to-tag=1728507150-1263982355499;from-tag=xUDOxkDMyYD But when I set action to: in cosole log I can find: EXECUTE sofia/internal/+48399500015 at neofon.tp.pl deflect(sip:conf_t0_2 at 10.79.21.108?Replaces=BW131151222041111-105561828 at 10.8.80.100;from-tag=139928099-1320408711222-) and FSW freezes until other side issues BYE. Is this a problem on my side (like different action is needed or some vars should be set)? Or is it related to FSW/mod_sofia (source)? TIA for your help or suggesions, Mariusz From manieq at wp.eu Fri Nov 4 16:53:47 2011 From: manieq at wp.eu (Mariusz Czulada) Date: Fri, 04 Nov 2011 14:53:47 +0100 Subject: [Freeswitch-users] Odp: Refer-To does not allow to add 'to-tag' and 'from-tag' In-Reply-To: <4eb3eb01a11587.03702142@wp.pl> References: <4eb3eb01a11587.03702142@wp.pl> Message-ID: <4eb3ee6bb1fe10.03581368@wp.pl> I found two mistakes in my post: 1. first action: "$" missing 2. Log entry - this one is correct: EXECUTE sofia/internal/+48399500015 at neofon.tp.pl deflect(sip:conf_t0_2 at 10.79.21.108?Replaces=BW130358011041111-381482516 at 10.8.80.100;to-tag=NUm5tvU3X15aF;from-tag=1253648243-1320408238012-) The rest is still valid. Mariusz Dnia 4-11-2011 o godz. 14:39 Mariusz Czulada napisa?(a): > Hi All, > > I have a problem with my FSW instance (version 1.0.7 on RHER 5.6 64bit). > Freeswitch does not allow to pass 'to-tag' or 'from-tag' in addition to > "Replaces=" while deflecting a call. > > This action works: > > data="sip:conf_t0_2 at 10.79.21.108?Replaces={sip_call_id}" /> > > The session I try to deflect is set from Broadworks 16 via ALU IMS core > (ISC 5450). Our supplier claims that to successfully handle REFER from > FSW, Refer-To header should include also 'to-tag' and 'from-tag'. > Example I got looks like this: > > Refer-To: > sip:conf_t0_2 at 10.79.21.108?Replaces=1387BD211B831526 at 172.16.10.114;to-tag=1728507150-1263982355499;from-tag=xUDOxkDMyYD > > But when I set action to: > > data="sip:conf_t0_2 at 10.79.21.108?Replaces=${sip_call_id};to-tag=${sip_to_tag};from-tag=${sip_from_tag}" > /> > > in cosole log I can find: > > EXECUTE sofia/internal/+48399500015 at neofon.tp.pl > deflect(sip:conf_t0_2 at 10.79.21.108?Replaces=BW131151222041111-105561828 at 10.8.80.100;from-tag=139928099-1320408711222-) > > and FSW freezes until other side issues BYE. > > Is this a problem on my side (like different action is needed or some > vars should be set)? Or is it related to FSW/mod_sofia (source)? > > TIA for your help or suggesions, > > Mariusz > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kris at kriskinc.com Fri Nov 4 16:57:11 2011 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 4 Nov 2011 09:57:11 -0400 Subject: [Freeswitch-users] Bad Contact Header In-Reply-To: References: Message-ID: Could you please post a full siptrace? You haven't included the REGISTER packet with the auth response. On Thu, Nov 3, 2011 at 12:24 PM, Dip Mehta wrote: > Hi , > I am trying to register x-lite 4.1 on freeswitch. > And i am using the default extensions : 1003 and password 1234 > However i am unable to get this registered and get the following error on > the console. > Output of > nano /usr/local/freeswitch/conf/directory/default/1003.xml > > ? > ? ? > ? ? ? > ? ? ? > ? ? > ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? ? value="$${outbound_caller_name}"/> > ? ? ? value="$${outbound_caller_id}"/> > ? ? ? > ? ? > ? > > > > recv 572 bytes from udp/[122.176.XX.XX:]:13334 at 16:18:23.774556: > ? ?------------------------------------------------------------------------ > ? ?REGISTER sip:199.167.XX.XX:?SIP/2.0 > ? ?Via: SIP/2.0/UDP > 122.176.144.226:13334;branch=z9hG4bK-d8754z-9ba4ee0f5e6ae5a7-1---d8754z-;rport > ? ?Max-Forwards: 70 > ? ?Contact: > ? ?To: "Extension 1003" > ? ?From: "Extension 1003";tag=77bad5c7 > ? ?Call-ID: MTRmNzQ3MDUyMTM4MGNhOGUzZjdlNWYyNWRlZDNjNmQ. > ? ?CSeq: 1 REGISTER > ? ?Expires: 3600 > ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > ? ?User-Agent: X-Lite 4 release 4.1 stamp 63214 > ? ?Content-Length: 0 > ? ?------------------------------------------------------------------------ > send 715 bytes to udp/[122.176.XX.XX:]:13334 at 16:18:23.775265: > ? ?------------------------------------------------------------------------ > ? ?SIP/2.0 401 Unauthorized > ? ?Via: SIP/2.0/UDP > 122.176.144.226:13334;branch=z9hG4bK-d8754z-9ba4ee0f5e6ae5a7-1---d8754z-;rport=13334 > ? ?From: "Extension 1003";tag=77bad5c7 > ? ?To: "Extension 1003" ;tag=gpjScQ0eXXj6K > ? ?Call-ID: MTRmNzQ3MDUyMTM4MGNhOGUzZjdlNWYyNWRlZDNjNmQ. > ? ?CSeq: 1 REGISTER > ? ?User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7b10506 2011-10-26 08-45-04 > -0500 > ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > ? ?Supported: timer, precondition, path, replaces > ? ?WWW-Authenticate: Digest realm="199.167.145.102", > nonce="740c2df0-0637-11e1-9e6e-6b39c14e0024", algorithm=MD5, qop="auth" > ? ?Content-Length: 0 > > > send 340 bytes to udp/[122.176.144.226]:13293 at 16:10:09.247665: > ? ?------------------------------------------------------------------------ > ? ?SIP/2.0 400 Bad Contact Header > ? ?Via: SIP/2.0/UDP > 122.176.XX.XX:;branch=z9hG4bK-d8754z-7c90ddcd2100ff77-1---d8754z-;rport=13293 > ? ?From: "1003" ;tag=d09ca84c > ? ?To: "1003" ;tag=DU6e75D452FeH > ? ?Call-ID: YjM3NWZiZjdkMzMwNTJhMjZkNmRhMWFkMGVkZGM1ZWI. > ? ?CSeq: 4 REGISTER > ? ?Content-Length: > Errors suggest this to be authentication issue. > However i am not able to resolve this. > Please advice. > Regards > Dip > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From pkelly at gmail.com Fri Nov 4 17:47:42 2011 From: pkelly at gmail.com (Pete Kelly) Date: Fri, 4 Nov 2011 14:47:42 +0000 Subject: [Freeswitch-users] Transcoding iLBC/GSM Message-ID: Hi I am trying to get freeswitch to bridge 2 calls while maintaining iLBC or GSM for legA, and allowing legB to choose its own codec. This is because legA is on a low bandwidth connection so allowing PCMA/PCMU is unwise. This works fine on a freeswitch install where there is a Sangoma transcoding card installed, however when I try it on Freeswitch with no Transcode card, it just responds with a 488. I read in the docs that iLBC is only transcodable at ptime=30, do you think this could be the reason the transcode is not happening. The SDP looks like this from legA: v=0. o=- 3529404938 3529404938 IN IP4 10.15.20.104. s=pjmedia. c=IN IP4 188.39.51.2. t=0 0. m=audio 4213 RTP/AVP 104 3 98 97 99 8 0 96. a=rtcp:4001. a=rtpmap:104 iLBC/8000. a=rtpmap:3 GSM/8000. a=rtpmap:98 speex/16000. a=rtpmap:97 speex/8000. a=rtpmap:99 speex/32000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=sendrecv. a=rtpmap:96 telephone-event/8000. a=fmtp:96 0-15. Anyone got any ideas/pointers I can look at? Thanks Pete From anthony.minessale at gmail.com Fri Nov 4 19:15:40 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Nov 2011 11:15:40 -0500 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: <1320404772101-6962394.post@n2.nabble.com> References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> Message-ID: The main use for it was a way to supersede pulling the hostname off the machine so you could set any value you wish that would be treated as if it were the hostname for the purposes of the many things that are intentionally kept separate based on hostname. The best practice is to have different hostnames because any data that should be shared over the database intentionally does not consider the hostname. On Fri, Nov 4, 2011 at 6:06 AM, peely wrote: > OK, that works, keeping the switchnames different doesn't dump the calls > table when the other switch starts and issuing a "sofia recover" recovers > the calls still on the other box. > > Can I ask when exactly the "switchname" should be used then? > > FS 3277: http://jira.freeswitch.org/browse/FS-3277 > > Suggests it was implemented for the exact reason of keeping the names of > two > switches in an HA configuration identical, but it seems they are best left > separate. > > I don't mind cleaning up some of the detail in the HA Wiki but would just > like to know when the names should be the same and when they should be > different i.e. shoudl the names be the same for kind kind of active / > active > setup? > > I don't know if Avi Marcus is reading this, maybe you can comment as I know > you have a good deal of experience in getting HA setups up and running. > > > Thanks, > > > > Neil. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6962394.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/a389f0cc/attachment.html From anthony.minessale at gmail.com Fri Nov 4 19:17:46 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Nov 2011 11:17:46 -0500 Subject: [Freeswitch-users] Transcoding iLBC/GSM In-Reply-To: References: Message-ID: not enough data, do you have iLBC in your configs and mod_ilbc loaded? you need more data like a trace "console loglevel debug" "sofia global siptrace on" On Fri, Nov 4, 2011 at 9:47 AM, Pete Kelly wrote: > Hi > > I am trying to get freeswitch to bridge 2 calls while maintaining iLBC > or GSM for legA, and allowing legB to choose its own codec. > > This is because legA is on a low bandwidth connection so allowing > PCMA/PCMU is unwise. > > This works fine on a freeswitch install where there is a Sangoma > transcoding card installed, however when I try it on Freeswitch with > no Transcode card, it just responds with a 488. > > I read in the docs that iLBC is only transcodable at ptime=30, do you > think this could be the reason the transcode is not happening. The SDP > looks like this from legA: > > v=0. > o=- 3529404938 3529404938 IN IP4 10.15.20.104. > s=pjmedia. > c=IN IP4 188.39.51.2. > t=0 0. > m=audio 4213 RTP/AVP 104 3 98 97 99 8 0 96. > a=rtcp:4001. > a=rtpmap:104 iLBC/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:98 speex/16000. > a=rtpmap:97 speex/8000. > a=rtpmap:99 speex/32000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=sendrecv. > a=rtpmap:96 telephone-event/8000. > a=fmtp:96 0-15. > > Anyone got any ideas/pointers I can look at? > > Thanks > > Pete > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/43f05b43/attachment.html From anthony.minessale at gmail.com Fri Nov 4 19:35:22 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Nov 2011 11:35:22 -0500 Subject: [Freeswitch-users] one gateway two registrations In-Reply-To: <2F46CC4C-70D7-4B94-A229-9E22834D5B64@dissauer.net> References: <2F46CC4C-70D7-4B94-A229-9E22834D5B64@dissauer.net> Message-ID: you must have the param to parse the domain set in both gateways Comment out the section in the one where you do not want to register. On Wed, Nov 2, 2011 at 10:20 PM, Roman Dissauer wrote: > I have a gateway defined in directory, this works fine but it ties to > register over both internal and external sip_profile. How can I restrict it > to just the external profile? > > Thanks! > Roman > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/e6735e4c/attachment-0001.html From erik.dekkers at certhon.com Fri Nov 4 10:56:12 2011 From: erik.dekkers at certhon.com (Erik Dekkers) Date: Fri, 4 Nov 2011 08:56:12 +0100 Subject: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on internal calls In-Reply-To: References: Message-ID: Hi Anthony, Would it be possible to update the display with information from the user directory. Say I call 200 (Alice), then freeswitch needs to do a lookup in the user directory. Let freeswitch take the ?effective_caller_id_name? variable and updates the display with that value after the bridge. Thnx, Erik Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Anthony Minessale Verzonden: vrijdag 21 oktober 2011 18:59 Aan: FreeSWITCH Users Help Onderwerp: Re: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on internal calls if you don't know who you are calling that is the display update you will get. if you do, set it in the bridge with origination_callee_id_name {origination_callee_id_name='Fred Smith'}user/1234 On Fri, Oct 21, 2011 at 8:17 AM, Erik Dekkers > wrote: Hey list, Recently (at least since two days) freeswitch is sending UPDATE messages with P-Asserted-Identity: "Outbound Call" after a call is established on the callers phone. It looks like it?s not updating the callee?s name with this UPDATE message since the callers name is displayed correctly. This is the dialplan I?m using for internal call routing: Full debug + siptrace: http://pastebin.freeswitch.org/17568 If there?s anything I can test please let me know Regards, Erik Dekkers (wvds-nl) [cid:image001.jpg at 01CC9ACF.9A75F030] ABC Westland 555 Tel: +31 174 245 727 P.O. Box 90 Fax: +31 174 248 739 2685 ZH Poeldijk Mob: +31 624 423 009 www.wvds.nl erik.dekkers at wvds.nl ________________________________ Bezoek ons op de Horti Fair / Visit us at the Horti Fair 1-4 november 2011 in Amsterdam, stand 11.0302. [cid:image002.jpg at 01CC9ACF.9A75F030] ________________________________ Wilk van der Sande is gecertificeerd Philips Led Horti Partner. DISCLAIMER Op alle offertes van, opdrachten aan en overeenkomsten met Bosch Inveka B.V. en Wilk van der Sande B.V. zijn de AVAG-verkoopvoorwaarden van toepassing All our quotations, all orders placed and all contracts concluded with Bosch Inveka B.V. and Wilk van der Sande B.V. are subject to the AVAG-Conditions. FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 [cid:imaged26589.PNG at 6bbfb770.4c93a67a] ABC Westland 555 Tel: +31 174 22 50 80 P.O. Box 90 Fax: +31 174 22 50 81 2685 ZH Poeldijk erik.dekkers at certhon.com The Netherlands www.certhon.com DISCLAIMER All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/8a661851/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 11848 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/8a661851/attachment-0002.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 13200 bytes Desc: image002.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/8a661851/attachment-0003.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: imaged26589.PNG Type: image/png Size: 6607 bytes Desc: imaged26589.PNG Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/8a661851/attachment-0001.png From B.Tietz at pinguin.ag Fri Nov 4 14:24:09 2011 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 4 Nov 2011 12:24:09 +0100 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: <1320404772101-6962394.post@n2.nabble.com> References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> Message-ID: <07BF4904977CC645B485E970424193AD0E689458B8@localhost> Hi, That is the reason. I think FS is able to act as a cluster. So many machines act as one and should have the same name... Just an idea. greets. >Can I ask when exactly the "switchname" should be used then? > >FS 3277: http://jira.freeswitch.org/browse/FS-3277 > >Suggests it was implemented for the exact reason of keeping the names of two switches in an HA configuration identical, but it seems they are best >left separate. > >I don't mind cleaning up some of the detail in the HA Wiki but would just like to know when the names should be the same and when they should be >different i.e. shoudl the names be the same for kind kind of active / active setup? From msc at freeswitch.org Fri Nov 4 20:20:35 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Nov 2011 10:20:35 -0700 Subject: [Freeswitch-users] Timezone with time conditions In-Reply-To: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com> References: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com> Message-ID: Can you expand upon this question a bit? My guess is that "anything's possible" but I'd like to know more about what you're doing before making any suggestions. -MC On Thu, Nov 3, 2011 at 2:11 AM, Spencer Thomason wrote: > Hello all, > Has anyone found an elegant solution to using time conditions with > multiple locations? For example suppose you have a multi tenant system > with time conditions, is there any way to set the call time zone on a per > call basis? > > Thanks, > Spencer > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/72085001/attachment.html From msc at freeswitch.org Fri Nov 4 20:29:27 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Nov 2011 10:29:27 -0700 Subject: [Freeswitch-users] Newbie In-Reply-To: References: Message-ID: Jacob, First off, welcome to FreeSWITCH! We hope you enjoy your stay. FreeSWITCH is one of those rapidly evolving projects, so it's best to update to latest git when you are experiencing issues. If you can reproduce this issue on the latest version of FS then it's time to collect debug and open a Jira ticket on jira.freeswitch.org. IIRC, our community member Mathieu Parent is the author of mod_skinny and he is always interested in hearing about people's experiences with it. -MC On Thu, Nov 3, 2011 at 2:48 PM, wrote: > Thanks for the link. > > My Setup: > CentOS 6.0 > FreeSWITCH snapshot 1.0.7 > 2 Cisco 7971 phones > 1 Cisco CM Communicator Softphone > > I have things setup I think. My phones register and show a line number, > but I am unable to make any calls. > > Executing "skinny status profile internal device SEP001122334455" crashes > FreeSWITCH. > > Any help? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian > Kielhofner > Sent: Thursday, November 03, 2011 4:30 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Newbie > > http://wiki.freeswitch.org/wiki/SCCP > > On Thu, Nov 3, 2011 at 4:15 PM, wrote: > > I am new to FreeSWITCH, testing to make sure it has all the > functionality we > > need, so I have a few questions I would like to ask. I am coming from > using > > Asterisk PBX and some Cisco Call Manager. As you see my question below > you > > will realize we are currently using SCCP (skinny) based phones, I am > trying > > to talk out customer into moving to SIP phones but no luck yet. > > > > > > > > 1) Is there good SCCP (skinny) support? > > > > 2) If so, which phones (79XX)? > > > > 3) Is there Asterisk AMI type messaging to control SCCP phones? > > > > > > > > Jacob Miles > > > > Software Engineer III > > > > L-3 Communications - Integrated Systems Greenville > > > > Jacob.E.Miles at L-3Com.com > > > > 903.457.4422 > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/1099fbf2/attachment.html From stkn at freeswitch.org Fri Nov 4 23:33:50 2011 From: stkn at freeswitch.org (Stefan Knoblich) Date: Fri, 04 Nov 2011 21:33:50 +0100 Subject: [Freeswitch-users] Heads up: autogenerated file "build/config/config.sub" removed Message-ID: <4EB44C2E.70100@freeswitch.org> Hi list, i just committed a patch that removes one of the autogenerated files from the git tree, so if you get a message like: "build/config/config.sub: No such file or directory" after your next update, try 1) first and if that doesn't fix it, try 2): 1) automake --add-missing 2) The good old, slow and safe: ./bootstrap.sh Unfortunately, this is one of the few files that autotools can't re-create on its own on the next "make" run. Sorry for the inconvenience, stkn -- ------------------------------------------------------------------------------- Stefan Knoblich | Web: http://www.axsentis.de/ axsentis GmbH | http://oss.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ------------------------------------------------------------------------------- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From anthony.minessale at gmail.com Fri Nov 4 23:37:00 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Nov 2011 15:37:00 -0500 Subject: [Freeswitch-users] How to check if user is behind NAT in a programmatic way? In-Reply-To: References: Message-ID: sip_nat_detected will be set to true or fs_nat=true param will be present in contact in the db if you see neither of these FS is not detecting its nat, probably you have to disable some other way that the phone or ALG is hiding the nat. On Fri, Nov 4, 2011 at 3:33 AM, Mateusz Bartczak wrote: > Hello > > I would like to allow direct p2p media (bypassing FS) between registered > SIP users, but only if one of them is not behind NAT. > > I can't force bypass_media=true for all calls, because many of users are > behind NAT and this will break communication. I only want to bypass_media > for users where it's possible. > > Is there any way of checking this in a programmatic way? I've checked > call variables, SIP registration info from database but I can't find any > related variable. > > Any suggestions? > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/53809037/attachment.html From stkn at freeswitch.org Fri Nov 4 23:40:14 2011 From: stkn at freeswitch.org (Stefan Knoblich) Date: Fri, 04 Nov 2011 21:40:14 +0100 Subject: [Freeswitch-users] Heads up: autogenerated file "build/config/config.sub" removed In-Reply-To: <4EB44C2E.70100@freeswitch.org> References: <4EB44C2E.70100@freeswitch.org> Message-ID: <4EB44DAE.5070005@freeswitch.org> The exact error message is: configure.in:27: required file `build/config/config.sub' not found On 04.11.2011 21:33, Stefan Knoblich wrote: > Hi list, > > i just committed a patch that removes one of the autogenerated files > from the git tree, so if you get a message like: > > "build/config/config.sub: No such file or directory" > > > after your next update, try 1) first and if that doesn't fix it, try 2): > > 1) automake --add-missing > > 2) The good old, slow and safe: ./bootstrap.sh > > > Unfortunately, this is one of the few files that autotools can't > re-create on its own on the next "make" run. > > > Sorry for the inconvenience, > stkn > -- ------------------------------------------------------------------------------- Stefan Knoblich | Web: http://www.axsentis.de/ axsentis GmbH | http://oss.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ------------------------------------------------------------------------------- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From djbinter at gmail.com Fri Nov 4 23:58:25 2011 From: djbinter at gmail.com (DJB International) Date: Fri, 4 Nov 2011 13:58:25 -0700 Subject: [Freeswitch-users] Heads up: autogenerated file "build/config/config.sub" removed In-Reply-To: <4EB44DAE.5070005@freeswitch.org> References: <4EB44C2E.70100@freeswitch.org> <4EB44DAE.5070005@freeswitch.org> Message-ID: I did the bootstrap.sh, but the built is still broken somewhere: http://pastebin.freeswitch.org/17688 Thank you. On Fri, Nov 4, 2011 at 1:40 PM, Stefan Knoblich wrote: > The exact error message is: > > configure.in:27: required file `build/config/config.sub' not found > > > On 04.11.2011 21:33, Stefan Knoblich wrote: > > Hi list, > > > > i just committed a patch that removes one of the autogenerated files > > from the git tree, so if you get a message like: > > > > "build/config/config.sub: No such file or directory" > > > > > > after your next update, try 1) first and if that doesn't fix it, try 2): > > > > 1) automake --add-missing > > > > 2) The good old, slow and safe: ./bootstrap.sh > > > > > > Unfortunately, this is one of the few files that autotools can't > > re-create on its own on the next "make" run. > > > > > > Sorry for the inconvenience, > > stkn > > > > > -- > > ------------------------------------------------------------------------------- > Stefan Knoblich | Web: http://www.axsentis.de/ > axsentis GmbH | http://oss.axsentis.de/ > Eupener Str. 74, 50933 Koeln, Germany | > Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de > UST-ID: DE244977565 | JID: > s.knoblich at jabber.axsentis.de > > ------------------------------------------------------------------------------- > Web: http://stkn.techmage.de/ > Email: stkn at freeswitch.org > IRC: #freeswitch-de @ irc.freenode.net > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/83af17a5/attachment-0001.html From anthony.minessale at gmail.com Sat Nov 5 00:43:56 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Nov 2011 16:43:56 -0500 Subject: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on internal calls In-Reply-To: References: Message-ID: yes you can do it with dialplan logic probably you can control what it says with the origination_callee_id_name and origination_callee_id_number variables On Fri, Nov 4, 2011 at 2:56 AM, Erik Dekkers wrote: > Hi Anthony,**** > > ** ** > > Would it be possible to update the display with information from the user > directory. **** > > Say I call 200 (Alice), then freeswitch needs to do a lookup in the user > directory. Let freeswitch take the ?effective_caller_id_name? variable and > updates the display with that value after the bridge.**** > > ** ** > > Thnx,**** > > ** ** > > Erik**** > > ** ** > > *Van:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Namens *Anthony Minessale > *Verzonden:* vrijdag 21 oktober 2011 18:59 > *Aan:* FreeSWITCH Users Help > *Onderwerp:* Re: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" > on internal calls**** > > ** ** > > if you don't know who you are calling that is the display update you will > get.**** > > if you do, set it in the bridge with origination_callee_id_name**** > > ** ** > > {origination_callee_id_name='Fred Smith'}user/1234**** > > ** ** > > ** ** > > On Fri, Oct 21, 2011 at 8:17 AM, Erik Dekkers > wrote:**** > > Hey list,**** > > **** > > Recently (at least since two days) freeswitch is sending UPDATE messages > with P-Asserted-Identity: "Outbound Call" after a > call is established on the callers phone.**** > > It looks like it?s not updating the callee?s name with this UPDATE message > since the callers name is displayed correctly.**** > > **** > > This is the dialplan I?m using for internal call routing:**** > > **** > > **** > > ** > ** > > **** > > **** > > **** > > **** > > Full debug + siptrace: http://pastebin.freeswitch.org/17568**** > > **** > > If there?s anything I can test please let me know**** > > **** > > Regards,**** > > **** > > Erik Dekkers (wvds-nl)**** > > **** > > ABC Westland 555**** > > Tel:**** > > +31 174 245 727**** > > P.O. Box 90**** > > Fax:**** > > +31 174 248 739**** > > 2685 ZH Poeldijk**** > > Mob:**** > > +31 624 423 009**** > > www.wvds.nl**** > > erik.dekkers at wvds.nl**** > ------------------------------ > > Bezoek ons op de Horti Fair / Visit us at the Horti Fair > 1-4 november 2011 in Amsterdam, stand 11.0302.**** > > * > *** > ------------------------------ > > Wilk van der Sande is gecertificeerd Philips Led Horti Partner.**** > > ** ** > > DISCLAIMER > Op alle offertes van, opdrachten aan en overeenkomsten met Bosch Inveka > B.V. en Wilk van der Sande B.V. zijn de AVAG-verkoopvoorwaarden van > toepassing > All our quotations, all orders placed and all contracts concluded with > Bosch Inveka B.V. and Wilk van der Sande B.V. are subject to the > AVAG-Conditions. **** > > **** > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900**** > > > ABC Westland 555 Tel: +31 174 22 50 80 P.O. Box 90 Fax: +31 174 22 50 81 2685 > ZH Poeldijk erik.dekkers at certhon.com The Netherlands www.certhon.com > DISCLAIMER > All our quotations, all orders and all contracts are subject to the > AVAG-CONDITIONS. > > Op alle offertes, opdrachten en overeenkomsten zijn de > AVAG-verkoopvoorwaarden van toepassing. > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 6607 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/4c486cc5/attachment-0001.png From stkn at freeswitch.org Sat Nov 5 00:55:22 2011 From: stkn at freeswitch.org (Stefan Knoblich) Date: Fri, 04 Nov 2011 22:55:22 +0100 Subject: [Freeswitch-users] Heads up: autogenerated file "build/config/config.sub" removed In-Reply-To: References: <4EB44C2E.70100@freeswitch.org> <4EB44DAE.5070005@freeswitch.org> Message-ID: <4EB45F4A.20708@freeswitch.org> Fixed in git (209ef00) On 04.11.2011 21:58, DJB International wrote: > I did the bootstrap.sh, but the built is still broken somewhere: > > http://pastebin.freeswitch.org/17688 > > Thank you. > > On Fri, Nov 4, 2011 at 1:40 PM, Stefan Knoblich wrote: > >> The exact error message is: >> >> configure.in:27: required file `build/config/config.sub' not found >> >> >> On 04.11.2011 21:33, Stefan Knoblich wrote: >>> Hi list, >>> >>> i just committed a patch that removes one of the autogenerated files >>> from the git tree, so if you get a message like: >>> >>> "build/config/config.sub: No such file or directory" >>> >>> >>> after your next update, try 1) first and if that doesn't fix it, try 2): >>> >>> 1) automake --add-missing >>> >>> 2) The good old, slow and safe: ./bootstrap.sh >>> >>> >>> Unfortunately, this is one of the few files that autotools can't >>> re-create on its own on the next "make" run. >>> >>> >>> Sorry for the inconvenience, >>> stkn >>> >> >> >> -- >> >> ------------------------------------------------------------------------------- >> Stefan Knoblich | Web: http://www.axsentis.de/ >> axsentis GmbH | http://oss.axsentis.de/ >> Eupener Str. 74, 50933 Koeln, Germany | >> Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de >> UST-ID: DE244977565 | JID: >> s.knoblich at jabber.axsentis.de >> >> ------------------------------------------------------------------------------- >> Web: http://stkn.techmage.de/ >> Email: stkn at freeswitch.org >> IRC: #freeswitch-de @ irc.freenode.net >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -- ------------------------------------------------------------------------------- Stefan Knoblich | Web: http://www.axsentis.de/ axsentis GmbH | http://oss.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ------------------------------------------------------------------------------- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From buscom123+fs at gmail.com Sat Nov 5 01:41:45 2011 From: buscom123+fs at gmail.com (R H) Date: Fri, 4 Nov 2011 16:41:45 -0600 Subject: [Freeswitch-users] Mod_callcenter Double-Dialing Intermittently Message-ID: Hey Everyone, I may have a bug for Mod_Callcenter but I wanted to see if anyone else might notice a setup issue before I send this on to Jira. I recently launched a production callcenter system for our company. We are currently beta testing the system with a small subset of our callers and we have noticed an issue where every so often (IE.* Intermittent issue*) when a caller arrives in queue and the queue offers the call to the agent,* the offer happens twice. * What gets worse is that once the agent answers one of the calls the other call continues to attempt to reach the agent even while they are talking to the person they are receiving the second call for. When the call times out we get a "NO ANSWER" and then the callcenter system attempts the call again, and again, and again until the agent is done with the caller. Even more interesting is that the agent state and member state keep changing to reflect the state of the second calling thread that keeps trying even though the caller is actually speaking to the agent. Here is a log from the freeswitch events. This was taken using log level 5 in the cli. Lines in BLUE are cli statements I am recording when events are fired that I care about. I am using Mod_managed to listen to events and update information in a company database. Lines in RED are notes about the logs. 2011-11-04 15:44:06.340969 [ALERT] switch_cpp.cpp:1223 PsiQueueEventConsumer - CallLogId Not Detected, Attempting to Create New :: Unavailable(2406740599) calling cc+ --- A CALLER JUST JOINED THE QUEUE --- 2011-11-04 15:44:45.160968 [NOTICE] switch_channel.c:915 New Channel sofia/external/sip:4503 at 97.75.182.83:59868[35e23ce8-072e-11e1-a636-2d222f47338b] 2011-11-04 15:44:45.160968 [NOTICE] switch_channel.c:915 New Channel sofia/external/sip:4503 at 97.75.182.83:59868[35e24e68-072e-11e1-a63a-2d222f47338b] --- NOTICE HOW 2 CHANNELS WERE INITIATED TO THE SAME EXTENSION/AGENT --- --- AGENT STATE: 'Receiving' --- 2011-11-04 15:44:45.200941 [NOTICE] sofia.c:5375 Ring-Ready sofia/external/ sip:4503 at 97.75.182.83:59868! 2011-11-04 15:44:45.481059 [NOTICE] sofia.c:5375 Ring-Ready sofia/external/ sip:4503 at 97.75.182.83:59868! 2011-11-04 15:44:48.740993 [NOTICE] sofia.c:5983 Channel [sofia/external/ sip:4503 at 97.75.182.83:59868] has been answered --- THE AGENT ANSWERED ONLY ONE OF THE CALLS --- --- AGENT STATE: 'In A Queue Call' --- 2011-11-04 15:44:55.000947 [NOTICE] switch_ivr_originate.c:3167 Hangup sofia/external/sip:4503 at 97.75.182.83:59868 [CS_CONSUME_MEDIA] [NO_ANSWER] 2011-11-04 15:44:55.000947 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [NO_ANSWER] 2011-11-04 15:44:55.000947 [NOTICE] switch_core_session.c:1395 Session 605 (sofia/external/sip:4503 at 97.75.182.83:59868) Ended 2011-11-04 15:44:55.000947 [NOTICE] switch_core_session.c:1397 Close Channel sofia/external/sip:4503 at 97.75.182.83:59868 [CS_DESTROY] --- THE SECOND CHANNEL JUST TIMED OUT AND FLAGGED THE CALL AS A "NO ANSWER" --- 2011-11-04 15:44:55.000947 [NOTICE] switch_channel.c:915 New Channel sofia/external/sip:4503 at 97.75.182.83:59868[3bc0c008-072e-11e1-a642-2d222f47338b] --- AGENT STATE: 'Receiving' --- 2011-11-04 15:44:55.040931 [NOTICE] sofia.c:5375 Ring-Ready sofia/external/ sip:4503 at 97.75.182.83:59868! --- SECOND CHANNEL ATTEMPTS AGAIN TO REACH THE AGENT, THE AGENT IS ALREADY TALKING TO THE CALLER --- 2011-11-04 15:45:05.000941 [NOTICE] switch_ivr_originate.c:3167 Hangup sofia/external/sip:4503 at 97.75.182.83:59868 [CS_CONSUME_MEDIA] [NO_ANSWER] 2011-11-04 15:45:05.000941 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [NO_ANSWER] 2011-11-04 15:45:05.000941 [NOTICE] switch_core_session.c:1395 Session 606 (sofia/external/sip:4503 at 97.75.182.83:59868) Ended 2011-11-04 15:45:05.000941 [NOTICE] switch_core_session.c:1397 Close Channel sofia/external/sip:4503 at 97.75.182.83:59868 [CS_DESTROY] --- NO ANSWER TIMEOUT REPEATS --- 2011-11-04 15:45:05.060918 [NOTICE] switch_channel.c:915 New Channel sofia/external/sip:4503 at 97.75.182.83:59868[41be0a9c-072e-11e1-a647-2d222f47338b] --- AGENT STATE: 'Receiving' --- --- TRIES AGAIN --- 2011-11-04 15:45:05.081035 [NOTICE] sofia.c:5375 Ring-Ready sofia/external/ sip:4503 at 97.75.182.83:59868! 2011-11-04 15:45:15.001026 [NOTICE] switch_ivr_originate.c:3167 Hangup sofia/external/sip:4503 at 97.75.182.83:59868 [CS_CONSUME_MEDIA] [NO_ANSWER] 2011-11-04 15:45:15.001026 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [NO_ANSWER] 2011-11-04 15:45:15.001026 [NOTICE] switch_core_session.c:1395 Session 607 (sofia/external/sip:4503 at 97.75.182.83:59868) Ended 2011-11-04 15:45:15.001026 [NOTICE] switch_core_session.c:1397 Close Channel sofia/external/sip:4503 at 97.75.182.83:59868 [CS_DESTROY] --- LOOPS CONTINUALLY FOR 7 MINUTES --- .... --- TRIES AGAIN --- 2011-11-04 15:52:15.040968 [NOTICE] switch_channel.c:915 New Channel sofia/external/sip:4503 at 97.75.182.83:59868[4208943a-072f-11e1-a72a-2d222f47338b] --- AGENT STATE: 'Receiving' --- 2011-11-04 15:52:15.080896 [NOTICE] sofia.c:5375 Ring-Ready sofia/external/ sip:4503 at 97.75.182.83:59868! 2011-11-04 15:52:25.000919 [NOTICE] switch_ivr_originate.c:3167 Hangup sofia/external/sip:4503 at 97.75.182.83:59868 [CS_CONSUME_MEDIA] [NO_ANSWER] 2011-11-04 15:52:25.000919 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [NO_ANSWER] 2011-11-04 15:52:25.000919 [NOTICE] switch_core_session.c:1395 Session 653 (sofia/external/sip:4503 at 97.75.182.83:59868) Ended 2011-11-04 15:52:25.000919 [NOTICE] switch_core_session.c:1397 Close Channel sofia/external/sip:4503 at 97.75.182.83:59868 [CS_DESTROY] 2011-11-04 15:52:28.140959 [NOTICE] switch_ivr_bridge.c:658 Hangup sofia/external/sip:4503 at 97.75.182.83:59868 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] --- AGENT ACTUALLY HANGS UP ON THE REAL CALL --- --- AGENT STATE: 'Waiting' --- 2011-11-04 15:52:28.180984 [NOTICE] switch_core_session.c:1395 Session 604 (sofia/external/sip:4503 at 97.75.182.83:59868) Ended 2011-11-04 15:52:28.180984 [NOTICE] switch_core_session.c:1397 Close Channel sofia/external/sip:4503 at 97.75.182.83:59868 [CS_DESTROY] --- THE LOOP IS ENDED BECAUSE THE MEMBER IS NO LONGER IN THE SYSTEM --- While the agent state keeps changing the agent is actually in a call with the caller this entire time. Events are firing every 10 seconds informing me this agent had a "No Answer" on an inbound call when they are actually talking to the caller. Here are some settings in case you want to know: *freeswitch at internal> callcenter_config queue list* name|strategy|moh_sound|time_base_score|tier_rules_apply|tier_rule_wait_second|tier_rule_wait_multiply_level|tier_rule_no_agent_no_wait|discard_abandoned_after|abandoned_resume_allowed|max_wait_time|max_wait_time_with_no_agent|max_wait_time_with_no_agent_time_reached|record_template *callcenter at sip.**** .com|longest-idle-agent|local_stream://moh|system|false|300|true|false|60|false|0|0|5|/usr/local/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav * +OK *freeswitch at internal> callcenter_config agent list* name|system|uuid|type|contact|status|state|max_no_answer|wrap_up_time|reject_delay_time|busy_delay_time|no_answer_delay_time|last_bridge_start|last_bridge_end|last_offered_call|last_status_change|no_answer_count|calls_answered|talk_time|ready_time 7003 at sip. .com|single_box|6d3dabc4-0732-11e1-a7c6-2d222f47338b|callback|[call_timeout=10]user/4513 at sip..com|Available|In a queue call|0|30|0|0|0|1320444899|1320444795|1320444896|1320443529|0|4|1129|0 *7004 at sip..com|single_box||callback|[call_timeout=10]user/4503 at sip. .com|Available|Waiting|0|30|0|0|0|1320445006|1320445099|1320445000|1320434600|0|24|4254|0 * +OK And here are the queue settings in XML. Same as above but easier to read. So to summarize, most calls work exactly as expected, but a percentage of our total call volume is currently having this issue. We are taking around 200 calls per day at the moment but we plan to take many more in the coming month. We have another system with an IVR in front of this and it only hands callers of a particular ivr option to this server. I should also note that we are running Dual Quad Core Intel Xeon's, 8GB DDR2, Suse 11.4 - x86_64, Mono 1.8.2. We are also testing 3 Polycom 450's and 5 X-Lite SoftPhones with our agents. Any suggestions or help would be appreciated. Thanks! Ryan H -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111104/87786e11/attachment-0001.html From kiruthika.bkite at gmail.com Sat Nov 5 07:24:48 2011 From: kiruthika.bkite at gmail.com (kiruthika sri) Date: Sat, 5 Nov 2011 09:54:48 +0530 Subject: [Freeswitch-users] FreeTDM PRI tapping issue In-Reply-To: References: Message-ID: Kindly help to fix this issue. On Thu, Nov 3, 2011 at 10:11 AM, kiruthika sri wrote: > Getting segfault after getting RINGING event in the first call itself. > > > On Wed, Nov 2, 2011 at 9:01 PM, Moises Silva wrote: > >> getting segfault right away? after a few calls? days? >> >> On Wed, Nov 2, 2011 at 12:35 AM, kiruthika sri > > wrote: >> >>> >>> Hi, >>> >>> I am trying to use the ftmod_pritap module for passive call recording. >>> >>> I did the followings. >>> * Installed wanpipe 3.5.23 for sangoma card AFT A102. >>> * Installed tap 1.4 (customized version of librpi for passive tapping) >>> * Installed freeswitch and freetdm with --with-pritap option >>> * Configured wanpipe, freetdm and freeswitch. >>> >>> Facing the following problem : >>> * FreeSWITCH is getting killed >>> * Getting SIGSEGV >>> >>> # wanrouter messages >>> >>> freeswitch[19720]: segfault at 8 ip 00ae00d1 sp b6ead208 error 4 in >>> libpri.so.1.4[ac5000+4c000] >>> >>> # gdb freeswitch core >>> >>> Program terminated with signal 11, Segmentation fault. >>> #0 0x00ae00d1 in q931_call_getcrv (ctrl=0x99fd9c0, call=0x0, >>> callmode=0xb6ead24c) at q931.c:5335 >>> 5335 *callmode = call->cr & 0x7; >>> >>> Kindly do the help on this. >>> >>> Thanks in advance. >>> >>> Regards >>> Kiruthika.U >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111105/485eb3d2/attachment.html From fieldpeak at gmail.com Sat Nov 5 12:58:48 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Sat, 5 Nov 2011 17:58:48 +0800 Subject: [Freeswitch-users] Help!!! -OPTIONS sent to the private IP for Natted registeration users Message-ID: Hi friends, my FS implemnt has only a public IP not behind NAT, and there are some registed users behind NAT, below is configure for internal profile, to keep the NAT mapping in remote router device, i open the keep live from FS to remote users (), howerver, i found the OPTIONS (from FS) sent to the private IP address of the remote users, it should send to the public IP of users external IP (router public IP), can we modify the configure to fix it? additionaly, i make a test, change configuration to " ", that is enable OPTIONS only sent to the NATted device, it did send the OPTIONS to the Natted device's public ip correctly that FS dectected, however, some device was not dected as a Natted device while it is behind NAT like below status, both of them are behind NAT, below is two registeration messages, the first one was detect as NAtted device, but the second was not, what is the mechanism for FS detect if a remote user behind NAT or not? Could anybody help to address this problem, thanks a lot! 9065 Registered(UDP-NAT)(unknown) exp(2011-11-05 18:30:30) expsecs(3611) 1026 Registered(UDP)(unknown) exp(2011-11-05 17:33:33) expsecs(194) ------------------------------------------------------------------------ recv 823 bytes from udp/[183.37.75.168]:9066 at 09:09:32.335911: ------------------------------------------------------------------------ REGISTER sip:124.193.106.104 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.86:9066 ;branch=z9hG4bK-d87543-ac6cfe2f736efb21-1--d87543-;rport Max-Forwards: 70 Contact: ;expires=0 To: "13580358068" From: "13580358068";tag=636eb146 Call-ID: 3c29a86eff650823 at bXlwYw.. CSeq: 4 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeam release 3015c stamp 27107 Authorization: Digest username="13580358068",realm="124.193.106.104",nonce="a6eb2963-21fa-4875-aa62-11e67d956f64",uri="sip:124.193.106.104",response="5083e7fe5eb078ca091279d7b1b9389f",cnonce="8b9c85686cc356e7",nc=00000001,qop=auth,algorithm=MD5 Content-Length: 0 ************** recv 638 bytes from udp/[124.193.106.98]:1026 at 09:12:43.891159: ------------------------------------------------------------------------ REGISTER sip:124.193.106.104 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.4:8060 ;rport;branch=z9hG4bK3089092136;xxx-nat-type=prcone Route: From: ;tag=181867095 To: Call-ID: 1197657332 at 192.168.2.4 CSeq: 308 REGISTER Contact: Authorization: Digest username="15130351737", realm="124.193.106.104", nonce="03d8e8b2-19b5-4aa5-910f-196951870bc3", uri="sip:124.193.106.104", response="c30374736da747eb33dc719def41ed08", algorithm=MD5 Max-Forwards: 70 User-Agent: YT-2.11.926.8 Expires: 200 Content-Length: 0 ******************* Profile internal content: -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111105/ce712b4f/attachment-0001.html From tculjaga at gmail.com Sat Nov 5 13:00:56 2011 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 5 Nov 2011 11:00:56 +0100 Subject: [Freeswitch-users] h323-conf-id, not found in dictionary In-Reply-To: <4EB39061.7070909@softnet.si> References: <4EB39061.7070909@softnet.si> Message-ID: On Fri, Nov 4, 2011 at 8:12 AM, Miha Zoubek wrote: > Hi, > > please help me with this issue as I realy do not know what should I do > next or what I am doing wrong. > > I have put all configuration files on pastbin. > http://pastebin.freeswitch.org/17673 > > Error that I am getting is: > > 2011-11-04 07:25:46.967596 [INFO] mod_dptools.c:1316 Before Auth > 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:428 Unknown attribute: > key:h323-conf-id, not found in dictionary > 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:546 An error occured > during RADIUS Authentication(RC=-1) > 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:702 An error occured > during radius authorization. > > I do not why I get this as I have this attribute defined in my dictionary.cisco > :( > point to only one dictionary file and at the end of it add an include to reference your cisco dictionary. or use dictionary.all file i attached here. > > Please help! > > Thank you! > > BR, > Miha > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111105/74639ca8/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: dictionary.all Type: application/octet-stream Size: 21151 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111105/74639ca8/attachment-0001.obj From avi at avimarcus.net Sat Nov 5 19:12:34 2011 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 5 Nov 2011 18:12:34 +0200 Subject: [Freeswitch-users] Timezone with time conditions In-Reply-To: References: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com> Message-ID: The general thing is a GUI that allows users to create a time condition. If you're running an ITSP, rather than a PBX for one company, then your customers will likely be spanning across time zones. It's one thing to normalize the times to your local time zones.. But that doesn't take DST into account. So basically, the best way would seem to be to pass the time zone to the condition that evaluates TOD. When I looked at this a while ago, I couldn't figure out how to do it.. -Avi On Fri, Nov 4, 2011 at 7:20 PM, Michael Collins wrote: > Can you expand upon this question a bit? My guess is that "anything's > possible" but I'd like to know more about what you're doing before making > any suggestions. > > -MC > > > On Thu, Nov 3, 2011 at 2:11 AM, Spencer Thomason < > spencer at 5ninesolutions.com> wrote: > >> Hello all, >> Has anyone found an elegant solution to using time conditions with >> multiple locations? For example suppose you have a multi tenant system >> with time conditions, is there any way to set the call time zone on a per >> call basis? >> >> Thanks, >> Spencer >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111105/5752f2f0/attachment.html From gilles.gerlinger at free.fr Sat Nov 5 22:04:10 2011 From: gilles.gerlinger at free.fr (gigerlin) Date: Sat, 5 Nov 2011 12:04:10 -0700 (PDT) Subject: [Freeswitch-users] FS & gtalk Message-ID: <1320519850646-6966503.post@n2.nabble.com> Hi all, I have trouble getting FS and gtalk correctly. The only thing I can do is calling a gmail user connected on gtalk (on Windows XP) from the SIP phone I registered in the jingle profile (extension 1000). Gtalk displays the message "Incoming Call From Extension 1000 1000" but when the call is accepted by clicking the button on the gtalk UI, I am not sure the call is established (there is no audio and the communication is cleared after 51 seconds). I am running an up-to-date release of FS (from GIT) on Ubuntu server 10.10 behind NAT. I have installed the gnutls library and recompiled FS. *Sofia status gives: * Name Type Data State ================================================================================================= 192.168.0.3 alias internal ALIASED internal profile sip:mod_sofia at 192.168.0.3:5060 RUNNING (0) internal profile sip:mod_sofia at 192.168.0.3:5061 RUNNING (0) (TLS) external profile sip:mod_sofia at 192.168.0.3:5080 RUNNING (0) external profile sip:mod_sofia at 192.168.0.3:5081 RUNNING (0) (TLS) external::example.com gateway sip:joeuser at example.com NOREG internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5061 RUNNING (0) (TLS) ================================================================================================= 3 profiles 1 alias *Below is the debug trace before answering the call: * Dialplan: sofia/internal/1000 at 192.168.0.3 Action answer() Dialplan: sofia/internal/1000 at 192.168.0.3 Action bridge(dingaling/gmail.com/gilles.gerlinger at gmail.com) 2011-11-05 19:55:10.214583 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/1000 at 192.168.0.3) State Change CS_ROUTING -> CS_EXECUTE 2011-11-05 19:55:10.214583 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1000 at 192.168.0.3 [BREAK] 2011-11-05 19:55:10.214583 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1000 at 192.168.0.3) State ROUTING going to sleep 2011-11-05 19:55:10.214583 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.3) Running State Change CS_EXECUTE 2011-11-05 19:55:10.214583 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1000 at 192.168.0.3) State EXECUTE 2011-11-05 19:55:10.214583 [DEBUG] mod_sofia.c:241 sofia/internal/1000 at 192.168.0.3 SOFIA EXECUTE 2011-11-05 19:55:10.214583 [DEBUG] switch_core_state_machine.c:192 sofia/internal/1000 at 192.168.0.3 Standard EXECUTE EXECUTE sofia/internal/1000 at 192.168.0.3 hash(insert/192.168.0.3-spymap/1000/af83062a-07df-11e1-8d0e-cdd18f8a8f7f) EXECUTE sofia/internal/1000 at 192.168.0.3 hash(insert/192.168.0.3-last_dial/1000/4545) EXECUTE sofia/internal/1000 at 192.168.0.3 hash(insert/192.168.0.3-last_dial/global/af83062a-07df-11e1-8d0e-cdd18f8a8f7f) EXECUTE sofia/internal/1000 at 192.168.0.3 set(RFC2822_DATE=Sat, 05 Nov 2011 19:55:10 +0100) 2011-11-05 19:55:10.307908 [DEBUG] mod_dptools.c:1177 sofia/internal/1000 at 192.168.0.3 SET [RFC2822_DATE]=[Sat, 05 Nov 2011 19:55:10 +0100] EXECUTE sofia/internal/1000 at 192.168.0.3 answer() 2011-11-05 19:55:10.307908 [DEBUG] sofia_glue.c:3100 AUDIO RTP [sofia/internal/1000 at 192.168.0.3] 192.168.0.3 port 20082 -> 192.168.0.7 port 7078 codec: 111 ms: 20 2011-11-05 19:55:10.307908 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 320 bytes per 20ms 2011-11-05 19:55:10.307908 [DEBUG] sofia_glue.c:3363 Set 2833 dtmf send payload to 101 2011-11-05 19:55:10.307908 [DEBUG] sofia_glue.c:3369 Set 2833 dtmf receive payload to 101 2011-11-05 19:55:10.307908 [DEBUG] mod_sofia.c:746 Local SDP sofia/internal/1000 at 192.168.0.3: v=0 o=FreeSWITCH 1320499228 1320499229 IN IP4 192.168.0.3 s=FreeSWITCH c=IN IP4 192.168.0.3 t=0 0 m=audio 20082 RTP/AVP 111 101 a=rtpmap:111 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2011-11-05 19:55:10.307908 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1000 at 192.168.0.3 [BREAK] 2011-11-05 19:55:10.307908 [DEBUG] switch_channel.c:3161 (sofia/internal/1000 at 192.168.0.3) Callstate Change RINGING -> ACTIVE *And here is what is displayed after I accept the call in the gtalk UI: * 2011-11-05 19:55:10.307908 [NOTICE] mod_dptools.c:1044 Channel [sofia/internal/1000 at 192.168.0.3] has been answered EXECUTE sofia/internal/1000 at 192.168.0.3 bridge(dingaling/gmail.com/gilles.gerlinger at gmail.com) 2011-11-05 19:55:10.307908 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-11-05 19:55:10.307908 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at 192.168.0.3 [BREAK] 2011-11-05 19:55:10.307908 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at 192.168.0.3 [BREAK] 2011-11-05 19:55:10.307908 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at 192.168.0.3 [BREAK] 2011-11-05 19:55:10.307908 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at 192.168.0.3 [BREAK] 2011-11-05 19:55:11.053751 [NOTICE] switch_channel.c:915 New Channel dingaling/gmail.com/gilles.gerlinger at gmail.com [af9bd074-07df-11e1-8d13-cdd18f8a8f7f] 2011-11-05 19:55:11.053751 [DEBUG] mod_dingaling.c:1861 (dingaling/gmail.com/gilles.gerlinger at gmail.com) State Change CS_NEW -> CS_INIT 2011-11-05 19:55:11.053751 [DEBUG] switch_core_session.c:1177 Send signal dingaling/gmail.com/gilles.gerlinger at gmail.com [BREAK] 2011-11-05 19:55:11.053751 [DEBUG] mod_dingaling.c:1396 dingaling/gmail.com/gilles.gerlinger at gmail.com CHANNEL KILL 2011-11-05 19:55:11.053751 [DEBUG] sofia.c:5283 Channel sofia/internal/1000 at 192.168.0.3 entering state [completed][200] 2011-11-05 19:55:11.053751 [DEBUG] sofia.c:5283 Channel sofia/internal/1000 at 192.168.0.3 entering state [ready][200] 2011-11-05 19:55:11.053751 [DEBUG] switch_core_state_machine.c:362 (dingaling/gmail.com/gilles.gerlinger at gmail.com) Running State Change CS_INIT 2011-11-05 19:55:11.053751 [DEBUG] switch_core_state_machine.c:401 (dingaling/gmail.com/gilles.gerlinger at gmail.com) State INIT 2011-11-05 19:55:11.073909 [NOTICE] mod_dingaling.c:1158 Ring-Ready dingaling/gmail.com/gilles.gerlinger at gmail.com! 2011-11-05 19:55:11.073909 [DEBUG] mod_dingaling.c:1111 Don't have my codec yet here's one 2011-11-05 19:55:11.073909 [DEBUG] mod_dingaling.c:1131 Send Describe [PCMU at 8000] 2011-11-05 19:55:11.773513 [DEBUG] mod_dingaling.c:3017 using Existing session for 9602298177 2011-11-05 19:55:11.873506 [DEBUG] mod_dingaling.c:3017 using Existing session for 9602298177 2011-11-05 19:55:11.873506 [DEBUG] mod_dingaling.c:3356 1 candidates 2011-11-05 19:55:11.873506 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1408 freeswitch at wetalk> freeswitch at wetalk> 2011-11-05 19:55:21.093499 [DEBUG] mod_dingaling.c:1056 Send Candidate 82.237.228.27:27456 [9tEsX2r0J2lRtJ4V] 2011-11-05 19:55:22.493584 [DEBUG] mod_dingaling.c:3017 using Existing session for 9602298177 2011-11-05 19:55:22.493584 [DEBUG] mod_dingaling.c:3270 Already decided on a codec 2011-11-05 19:55:24.093532 [DEBUG] mod_dingaling.c:3017 using Existing session for 9602298177 2011-11-05 19:55:24.093532 [DEBUG] mod_dingaling.c:3356 2 candidates 2011-11-05 19:55:24.093532 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1408 2011-11-05 19:55:24.093532 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1425 2011-11-05 19:55:26.393902 [DEBUG] mod_dingaling.c:3017 using Existing session for 9602298177 2011-11-05 19:55:26.393902 [DEBUG] mod_dingaling.c:3356 3 candidates etc... *On the other hand I cannot call from gtalk the SIP phone registered with a gmail account (extension 1000). Below is the debugging trace: * 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3150 Creating an identity for 3626629742 gilles.gerlinger at gmail.com/Talk.v104900E9C7C 1000 2011-11-05 19:50:37.633502 [NOTICE] switch_channel.c:915 New Channel dingaling/1000 [0d146fa0-07df-11e1-8d0a-cdd18f8a8f7f] 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3178 Creating a session for 3626629742 2011-11-05 19:50:37.633502 [NOTICE] switch_channel.c:913 Rename Channel dingaling/1000->DingaLing/new [0d146fa0-07df-11e1-8d0a-cdd18f8a8f7f] 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3182 (DingaLing/new) State Change CS_NEW -> CS_INIT 2011-11-05 19:50:37.633502 [DEBUG] switch_core_session.c:1177 Send signal DingaLing/new [BREAK] 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:1396 DingaLing/new CHANNEL KILL 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3284 12 payloads 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3286 Available Payload ISAC 103 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3294 compare ISAC 103/16000 to PCMU 0/8000 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3286 Available Payload IPCMWB 97 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3294 compare IPCMWB 97/16000 to PCMU 0/8000 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3286 Available Payload speex 99 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3294 compare speex 99/16000 to PCMU 0/8000 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3286 Available Payload G723 4 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3294 compare G723 4/8000 to PCMU 0/8000 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3286 Available Payload speex 98 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3294 compare speex 98/8000 to PCMU 0/8000 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3286 Available Payload EG711U 100 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3294 compare EG711U 100/8000 to PCMU 0/8000 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3286 Available Payload EG711A 101 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3294 compare EG711A 101/8000 to PCMU 0/8000 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3286 Available Payload PCMU 0 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3294 compare PCMU 0/8000 to PCMU 0/8000 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:3305 Choosing Payload index 0 PCMU 0 2011-11-05 19:50:37.633502 [DEBUG] mod_dingaling.c:1131 Send Describe [PCMU at 8000] 2011-11-05 19:50:37.633502 [DEBUG] switch_core_state_machine.c:362 (DingaLing/new) Running State Change CS_INIT 2011-11-05 19:50:37.633502 [DEBUG] switch_core_state_machine.c:401 (DingaLing/new) State INIT 2011-11-05 19:50:37.633502 [NOTICE] mod_dingaling.c:1158 Ring-Ready DingaLing/new! 2011-11-05 19:50:37.733589 [DEBUG] mod_dingaling.c:3017 using Existing session for 3626629742 2011-11-05 19:50:37.733589 [DEBUG] mod_dingaling.c:3356 1 candidates 2011-11-05 19:50:37.733589 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1206 2011-11-05 19:50:47.633968 [DEBUG] mod_dingaling.c:1056 Send Candidate 82.237.228.27:27976 [jHNpXRZxqmR2Fdhe] 2011-11-05 19:50:48.213494 [DEBUG] mod_dingaling.c:3017 using Existing session for 3626629742 2011-11-05 19:50:48.213494 [DEBUG] mod_dingaling.c:3356 1 candidates 2011-11-05 19:50:48.213494 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1206 2011-11-05 19:50:50.213524 [DEBUG] mod_dingaling.c:3017 using Existing session for 3626629742 2011-11-05 19:50:50.213524 [DEBUG] mod_dingaling.c:3356 2 candidates 2011-11-05 19:50:50.213524 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1206 2011-11-05 19:50:50.213524 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1211 2011-11-05 19:50:52.514082 [DEBUG] mod_dingaling.c:3017 using Existing session for 3626629742 2011-11-05 19:50:52.514082 [DEBUG] mod_dingaling.c:3356 3 candidates 2011-11-05 19:50:52.514082 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1206 2011-11-05 19:50:52.514082 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1211 2011-11-05 19:50:52.514082 [DEBUG] mod_dingaling.c:3392 candidates 209.85.229.126:19294 2011-11-05 19:50:52.613655 [DEBUG] mod_dingaling.c:3017 using Existing session for 3626629742 2011-11-05 19:50:52.613655 [DEBUG] mod_dingaling.c:3356 3 candidates 2011-11-05 19:50:52.613655 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1206 2011-11-05 19:50:52.613655 [DEBUG] mod_dingaling.c:3392 candidates 192.168.0.7:1211 2011-11-05 19:50:52.613655 [DEBUG] mod_dingaling.c:3392 candidates 209.85.229.126:443 2011-11-05 19:50:57.673499 [DEBUG] mod_dingaling.c:3017 using Existing session for 3626629742 *Thank you in advance for your help. Gilles Gerlinger * -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-gtalk-tp6966503p6966503.html Sent from the freeswitch-users mailing list archive at Nabble.com. From yehavi.bourvine at gmail.com Sun Nov 6 09:29:44 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 6 Nov 2011 08:29:44 +0200 Subject: [Freeswitch-users] Automatic fax formating sending via Freeswitch In-Reply-To: <40650EFDEA0042788AF84CBE8236745A@e1705> References: <40650EFDEA0042788AF84CBE8236745A@e1705> Message-ID: Thanks, but that's the final stage. I am looking for the step befoer convert: A "language" that is textual and has decent WISIWIG editing tools; I want to take this "language" and feed it to ImageMagic to get the final image. Thanks, __Yehavi: 2011/11/3 Madovsky > ** > convert from Imagemagick > > ----- Original Message ----- > *From:* Yehavi Bourvine > *To:* FreeSWITCH Users Help > *Sent:* Thursday, November 03, 2011 10:30 AM > *Subject:* [Freeswitch-users] Automatic fax formating sending via > Freeswitch > > Hello, > > This is not really a FreeSwitch question, but I guess people here might > have answer to this... > > I have to generate fax from two files: template and list of fields to > replace in the template. I would like the template to be a tectual thing so > I can use sed to replace the fields in it, and then convert the result to > PFD/TIFF and send it via FS. > > Is there some tectual format (like Tex, SVG) that can be eaily created by > some WISIWIG software? > > 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/20111106/90979ceb/attachment-0001.html From fieldpeak at gmail.com Sun Nov 6 14:04:39 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Sun, 6 Nov 2011 19:04:39 +0800 Subject: [Freeswitch-users] Help!!! -OPTIONS sent to the private IP for Natted registeration users In-Reply-To: References: Message-ID: i fixed it by disable below param, cheers! "NDLB-received-in-nat-reg-contact" 2011/11/5 fieldpeak > Hi friends, > > my FS implemnt has only a public IP not behind NAT, and there are some > registed users behind NAT, below is configure for internal profile, to keep > the NAT mapping in remote router device, i open the keep live from FS to > remote users (), howerver, > i found the OPTIONS (from FS) sent to the private IP address of the remote > users, it should send to the public IP of users external IP (router public > IP), can we modify the configure to fix it? > > additionaly, i make a test, change configuration to " name="nat-options-ping" value="true"/>", that is enable OPTIONS only sent > to the NATted device, > it did send the OPTIONS to the Natted device's public ip correctly > that FS dectected, however, some device was not dected as a Natted device > while it is behind NAT like below status, both of them are behind NAT, > below is two registeration messages, the first one was detect as NAtted > device, but the second was not, what is the mechanism for FS detect if a > remote user behind NAT or not? Could anybody help to address this problem, > thanks a lot! > > 9065 Registered(UDP-NAT)(unknown) exp(2011-11-05 18:30:30) > expsecs(3611) > 1026 Registered(UDP)(unknown) exp(2011-11-05 17:33:33) > expsecs(194) > > > ------------------------------------------------------------------------ > recv 823 bytes from udp/[183.37.75.168]:9066 at 09:09:32.335911: > ------------------------------------------------------------------------ > REGISTER sip:124.193.106.104 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.86:9066 > ;branch=z9hG4bK-d87543-ac6cfe2f736efb21-1--d87543-;rport > Max-Forwards: 70 > Contact: ;rinstance=730e3f0e44ed8142>;expires=0 > To: "13580358068" > From: "13580358068";tag=636eb146 > Call-ID: 3c29a86eff650823 at bXlwYw.. > CSeq: 4 REGISTER > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > Supported: eventlist > User-Agent: eyeBeam release 3015c stamp 27107 > Authorization: Digest > username="13580358068",realm="124.193.106.104",nonce="a6eb2963-21fa-4875-aa62-11e67d956f64",uri="sip:124.193.106.104",response="5083e7fe5eb078ca091279d7b1b9389f",cnonce="8b9c85686cc356e7",nc=00000001,qop=auth,algorithm=MD5 > Content-Length: 0 > > ************** > recv 638 bytes from udp/[124.193.106.98]:1026 at 09:12:43.891159: > ------------------------------------------------------------------------ > REGISTER sip:124.193.106.104 SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.4:8060 > ;rport;branch=z9hG4bK3089092136;xxx-nat-type=prcone > Route: > From: ;tag=181867095 > To: > Call-ID: 1197657332 at 192.168.2.4 > CSeq: 308 REGISTER > Contact: > Authorization: Digest username="15130351737", realm="124.193.106.104", > nonce="03d8e8b2-19b5-4aa5-910f-196951870bc3", uri="sip:124.193.106.104", > response="c30374736da747eb33dc719def41ed08", algorithm=MD5 > Max-Forwards: 70 > User-Agent: YT-2.11.926.8 > Expires: 200 > Content-Length: 0 > > ******************* > > > Profile internal content: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > Regards, > Charles > > -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111106/7161f985/attachment.html From admin at blindi.net Sun Nov 6 14:55:10 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Sun, 6 Nov 2011 12:55:10 +0100 (CET) Subject: [Freeswitch-users] Problem Verifying an existing file with lua Message-ID: Hi all, I like to export a vatiable to my dialplan. The variable file_name_ok should give back the value true if the file exists My fialplan: My problem: i can.t export the variable from the luascript my scipt: -- function used to check for the existence of a file function file_exists(fname) local f = io.open(fname, "r") if (f and f:read()) then return true end end if ( session:ready() ) then session:answer( ); datei = session:getVariable("file_name_ok"); if (file_exists(datei..extension..".wav")) then session:setVariable("file_name_ok", "true"); else session:setVariable("file_name_ok", "true"); end end can your help me please? --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From govoiper at gmail.com Sun Nov 6 18:00:49 2011 From: govoiper at gmail.com (Sammy Govind) Date: Sun, 6 Nov 2011 20:00:49 +0500 Subject: [Freeswitch-users] Problem Verifying an existing file with lua In-Reply-To: References: Message-ID: Hey, What I understand you are trying to do is get a variable name from FS and check from LUA if the file exists ! file_name_ok is just a flag to return true or false depending upon existence of caal_screen_filename..if this is the case then datei = session:getVariable("file_**name_ok"); --You are getting the variable containing filename here. shouldn't it be ? datei = session:getVariable("call_screen_filename"); also, if (file_exists(datei..extension.**.".wav")) then > session:setVariable("file_**name_ok", "true"); > else > session:setVariable("file_**name_ok", "true"); > end returns true in any case file exists or not. Please see if this is right ! -- Regards, Sammy On Sun, Nov 6, 2011 at 4:55 PM, Thomas Hoellriegel wrote: > Hi all, > I like to export a vatiable to my dialplan. > The variable file_name_ok should give back the value true if the file > exists > My fialplan: > > > > My problem: i can.t export the variable from the luascript > > my scipt: > -- function used to check for the existence of a file > function file_exists(fname) > local f = io.open(fname, "r") > if (f and f:read()) then return true end > end > if ( session:ready() ) then > session:answer( ); > > datei = session:getVariable("file_**name_ok"); > if (file_exists(datei..extension.**.".wav")) then > session:setVariable("file_**name_ok", "true"); > else > session:setVariable("file_**name_ok", "true"); > end > end > > can your help me please? > > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111106/cc312d1d/attachment-0001.html From brian at freeswitch.org Mon Nov 7 02:33:30 2011 From: brian at freeswitch.org (Brian West) Date: Sun, 6 Nov 2011 17:33:30 -0600 Subject: [Freeswitch-users] External IVR and MWI In-Reply-To: <2201b1154e174794473a5792a0649a60@mail.gmail.com> References: <2201b1154e174794473a5792a0649a60@mail.gmail.com> Message-ID: <8F98DEFE-55EF-41F0-9800-60077E3C1F28@freeswitch.org> Why are you sending them to an external IVR? /b On Sep 20, 2011, at 8:03 AM, Ira Tessler wrote: > I am planning to use an external IVR with a Freeswitch system. The phones > will register with Freeswitch and forward to the external IVR for > voicemails. When the external IVR receives a voicemail, I need to send a > message to Freeswitch to turn on the message waiting indicator on the phone. > I will be using a Multi Tenant Freeswitch system. The code will need to be > programmed in C#. > > > > What is the best way to go about this? Using the ESL API? Does anyone have > any examples? > > > > Thanks, > > > > Ira Tessler > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From admin at blindi.net Mon Nov 7 02:51:14 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Mon, 7 Nov 2011 00:51:14 +0100 (CET) Subject: [Freeswitch-users] Problem Verifying an existing file with lua In-Reply-To: References: Message-ID: Hi Sammy, I become no output true or false. My Script is now: function file_exists(fname) local f = io.open(fname, "r") if (f and f:read()) then return true end end if ( session:ready() ) then session:answer( ); datei = session:getVariable("call_screen_filename"); if (file_exists(datei..extension.**.".wav")) then session:setVariable("fileex", "true"); session:setVariable(fileex, true) returns true; else session:setVariable("fileex", "false"); returns false; end end The variable fileex is emty. The variable: call_screen_filename will not be changed after to execute these script. I like to setup a callscreening. if the name exists, transfer to the given extension. If the name don.t exists, start the recording and transwer the call. Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From nestor at tiendalinux.com Mon Nov 7 05:07:17 2011 From: nestor at tiendalinux.com (Nestor A Diaz) Date: Sun, 06 Nov 2011 21:07:17 -0500 Subject: [Freeswitch-users] mod_timerfd build error on debian squeeze amd64 Message-ID: <4EB73D55.9070309@tiendalinux.com> Hi Guys, i am experiencing a problem compiling freeswitch under debian squeeze (AMD64): # tail -15 freeswitch_1.0.head-git.master.20110530.1-1_amd64.build make[6]: Entering directory `/usr/local/src/freeswitch/src/mod/timers/mod_timerfd' Compiling /usr/local/src/freeswitch/src/mod/timers/mod_timerfd/mod_timerfd.c... quiet_libtool: compile: x86_64-linux-gnu-gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -ggdb -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/timers/mod_timerfd/mod_timerfd.c -fPIC -DPIC -o .libs/mod_timerfd.o quiet_libtool: compile: x86_64-linux-gnu-gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -ggdb -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/timers/mod_timerfd/mod_timerfd.c -o mod_timerfd.o >/dev/null 2>&1 Creating mod_timerfd.la... make[6]: Leaving directory `/usr/local/src/freeswitch/src/mod/timers/mod_timerfd' make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/timers/mod_timerfd' make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[3]: Leaving directory `/usr/local/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [build-stamp] Error 2 dpkg-buildpackage: error: debian/rules build gave error exit status 2 I got the git source through: git clone git://git.freeswitch.org/freeswitch.git and install dependencies before: apt-get install autoconf automake bison build-essential debhelper devscripts git-core libasound2-dev libcurl4-openssl-dev libdb-dev libexpat1-dev libgdbm-dev libgnutls-dev libjpeg-dev libncurses5 libncurses5-dev libogg-dev libperl-dev libssl-dev libtiff4-dev libtool libvorbis-dev libx11-dev libzrtpcpp-dev make python2.6-dev python-dev unixodbc-dev uuid-dev wget zlib1g-dev gawk pkg-config and run: debuild -i -us -uc -b Thank you very much ! -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211 at tiendalinux.com Email/MSN: nestor at tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia From nestor at tiendalinux.com Mon Nov 7 06:36:52 2011 From: nestor at tiendalinux.com (Nestor A Diaz) Date: Sun, 06 Nov 2011 22:36:52 -0500 Subject: [Freeswitch-users] SOLVED Re: mod_timerfd build error on debian squeeze amd64 In-Reply-To: <4EB73D55.9070309@tiendalinux.com> References: <4EB73D55.9070309@tiendalinux.com> Message-ID: <4EB75254.4070804@tiendalinux.com> Hi, the problem was not related to the mod_timerfd, the uuid library was missing from the configure, when it tries to ldd tone2wav, so i have to change in the debian/rules file the line that begins with ./configure and add a LIBS=-luuid, and now i have the packages built ! ./configure --prefix=/opt/freeswitch --host=$(DEB_HOST_GNU_TYPE) --build=$(DEB_BUILD_GNU_TYPE) ${FEATURES} LIBS=-luuid Not sure it that is a problem of the freeswitch generated ./configure files or debian maintainers way of generating them. CC. Debian Maintainer, a.k.a. "The maintainer is always right" Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211 at tiendalinux.com Email/MSN: nestor at tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia On 11/06/2011 09:07 PM, Nestor A Diaz wrote: > Hi Guys, i am experiencing a problem compiling freeswitch under debian > squeeze (AMD64): > > # tail -15 freeswitch_1.0.head-git.master.20110530.1-1_amd64.build > make[6]: Entering directory > `/usr/local/src/freeswitch/src/mod/timers/mod_timerfd' > Compiling > /usr/local/src/freeswitch/src/mod/timers/mod_timerfd/mod_timerfd.c... > quiet_libtool: compile: x86_64-linux-gnu-gcc > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb -DHAVE_OPENSSL -ggdb -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/local/src/freeswitch/src/mod/timers/mod_timerfd/mod_timerfd.c > -fPIC -DPIC -o .libs/mod_timerfd.o > quiet_libtool: compile: x86_64-linux-gnu-gcc > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb -DHAVE_OPENSSL -ggdb -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/local/src/freeswitch/src/mod/timers/mod_timerfd/mod_timerfd.c -o > mod_timerfd.o>/dev/null 2>&1 > Creating mod_timerfd.la... > make[6]: Leaving directory > `/usr/local/src/freeswitch/src/mod/timers/mod_timerfd' > make[5]: Leaving directory > `/usr/local/src/freeswitch/src/mod/timers/mod_timerfd' > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [build-stamp] Error 2 > dpkg-buildpackage: error: debian/rules build gave error exit status 2 > > I got the git source through: > > git clone git://git.freeswitch.org/freeswitch.git > > and install dependencies before: > > apt-get install autoconf automake bison build-essential debhelper > devscripts git-core libasound2-dev libcurl4-openssl-dev libdb-dev > libexpat1-dev libgdbm-dev libgnutls-dev libjpeg-dev libncurses5 > libncurses5-dev libogg-dev libperl-dev libssl-dev libtiff4-dev libtool > libvorbis-dev libx11-dev libzrtpcpp-dev make python2.6-dev python-dev > unixodbc-dev uuid-dev wget zlib1g-dev gawk pkg-config > > and run: > > debuild -i -us -uc -b > > Thank you very much ! > > From trob at freemail.hu Mon Nov 7 07:34:34 2011 From: trob at freemail.hu (=?ISO-8859-2?Q?T=F3th_R=F3bert?=) Date: Mon, 07 Nov 2011 05:34:34 +0100 Subject: [Freeswitch-users] BLF to show something Message-ID: <4EB75FDA.5070104@freemail.hu> Hi Is it possible to use BLF to show something from either dialplan or LUA? (For example i have a dialplan-code to enable or disable the pickup from the phone. By dialing *9150 i enable or disable the possibility of pickup my ringing calls from another phone. *91 is the code of the setting and 50 is the phone's number. I store the setting in db, and it works fine. But i would like to show the user his/her actual setting by green or red lamp.) Is this possible now with some trick? Or is it possible to make a feature for this? I tried it with http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_presence but with no success. (Or am i the unskilful?) I got two idea: - sub to dp+ext where ext is the extent you want to monitor - use the new mapping feature to map it to a proto. But i can not find any informations about these. Thanks Robert From miha at softnet.si Mon Nov 7 11:14:22 2011 From: miha at softnet.si (Miha Zoubek) Date: Mon, 07 Nov 2011 09:14:22 +0100 Subject: [Freeswitch-users] h323-conf-id, not found in dictionary In-Reply-To: References: <4EB39061.7070909@softnet.si> Message-ID: <4EB7935E.1030405@softnet.si> @Tihomir thank you for that file. Now it works:) Just one question. What kind of encryption is freeswitch (radclient) using for users? I have difined share secret like that: I am getting error on freeradius side that share secret is wrong. I have defined shared secret in clinet.conf (freeradius) and mod.radius.conf.xml. Error form freeradius: [pap] Passwords don't match ++[pap] returns reject Failed to authenticate the user. WARNING: Unprintable characters in the password. Double-check the shared secret on the server and the NAS! Using Post-Auth-Type Reject This is client.conf: client xxx.xxx.xxx.xxx { secret = test1234 } Thank you! BR, Miha On 11/5/2011 11:00 AM, Tihomir Culjaga wrote: > > > On Fri, Nov 4, 2011 at 8:12 AM, Miha Zoubek > wrote: > > Hi, > > please help me with this issue as I realy do not know what should > I do next or what I am doing wrong. > > I have put all configuration files on pastbin. > http://pastebin.freeswitch.org/17673 > > Error that I am getting is: > > 2011-11-04 07:25:46.967596 [INFO] mod_dptools.c:1316 Before Auth > 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:428 Unknown > attribute: key:h323-conf-id, not found in dictionary > 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:546 An error > occured during RADIUS Authentication(RC=-1) > 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:702 An error > occured during radius authorization. > > I do not why I get this as I have this attribute defined in my > dictionary.cisco :( > > > point to only one dictionary file and at the end of it add an include > to reference your cisco dictionary. > > or use dictionary.all file i attached here. > > > Please help! > > Thank you! > > BR, > Miha > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/dfefd32b/attachment-0001.html From miha at softnet.si Mon Nov 7 11:43:06 2011 From: miha at softnet.si (Miha Zoubek) Date: Mon, 07 Nov 2011 09:43:06 +0100 Subject: [Freeswitch-users] h323-conf-id, not found in dictionary In-Reply-To: <4EB7935E.1030405@softnet.si> References: <4EB39061.7070909@softnet.si> <4EB7935E.1030405@softnet.si> Message-ID: <4EB79A1A.9000800@softnet.si> @Tihomir ignore this:) xxx.xxx.xxx.xxx:1813:test1234" my mistake:) Working just perfect:) BR, Miha On 11/7/2011 9:14 AM, Miha Zoubek wrote: > @Tihomir thank you for that file. Now it works:) > > Just one question. What kind of encryption is freeswitch (radclient) > using for users? > I have difined share secret like that: > > > I am getting error on freeradius side that share secret is wrong. I > have defined shared secret in clinet.conf (freeradius) and > mod.radius.conf.xml. > > Error form freeradius: > > [pap] Passwords don't match > ++[pap] returns reject > Failed to authenticate the user. > WARNING: Unprintable characters in the password. Double-check the > shared secret on the server and the NAS! > Using Post-Auth-Type Reject > > This is client.conf: > > client xxx.xxx.xxx.xxx { > secret = test1234 > } > > Thank you! > > BR, > Miha > > On 11/5/2011 11:00 AM, Tihomir Culjaga wrote: >> >> >> On Fri, Nov 4, 2011 at 8:12 AM, Miha Zoubek > > wrote: >> >> Hi, >> >> please help me with this issue as I realy do not know what should >> I do next or what I am doing wrong. >> >> I have put all configuration files on pastbin. >> http://pastebin.freeswitch.org/17673 >> >> Error that I am getting is: >> >> 2011-11-04 07:25:46.967596 [INFO] mod_dptools.c:1316 Before Auth >> 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:428 Unknown >> attribute: key:h323-conf-id, not found in dictionary >> 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:546 An error >> occured during RADIUS Authentication(RC=-1) >> 2011-11-04 07:25:48.967589 [ERR] mod_rad_auth.c:702 An error >> occured during radius authorization. >> >> I do not why I get this as I have this attribute defined in my >> dictionary.cisco :( >> >> >> point to only one dictionary file and at the end of it add an include >> to reference your cisco dictionary. >> >> or use dictionary.all file i attached here. >> >> >> Please help! >> >> Thank you! >> >> BR, >> Miha >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/10cf8d1a/attachment-0001.html From miha at softnet.si Mon Nov 7 11:47:12 2011 From: miha at softnet.si (Miha Zoubek) Date: Mon, 07 Nov 2011 09:47:12 +0100 Subject: [Freeswitch-users] SIP trunk without password and username Message-ID: <4EB79B10.60608@softnet.si> Hi, we are having all trunks on SBC without usr/pass. If i left param name and password empty I get ERROR: username param is REQUIRED! and also same if I delete this param usr/pass. How can I configure GW sip trunk without password and username. I would like to have only IP. Thank you! BR, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/d915d4ac/attachment.html From fdelawarde at wirelessmundi.com Mon Nov 7 12:17:45 2011 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 07 Nov 2011 10:17:45 +0100 Subject: [Freeswitch-users] BLF to show something In-Reply-To: <4EB75FDA.5070104@freemail.hu> References: <4EB75FDA.5070104@freemail.hu> Message-ID: <1320657465.4842.31.camel@luna.madrid.commsmundi.com> On Mon, 2011-11-07 at 05:34 +0100, T?th R?bert wrote: > I got two idea: > - sub to dp+ext where ext is the extent you want to monitor > - use the new mapping feature to map it to a proto. A really cool feature would be to have some type of "presenceplan" (like dialplan or chatplan) called upon SUBSCRIBE. It would allow to setup mappings to local extensions/queues/..., given any type of condition on the origin and destination, refuse the subscription given ACLs, send a status fetched from ODBC, or why not even call an external program to fetch a status every time we send a NOTIFY. More or less, "The Ultimate Presence Server"... I would note that the new mapping feature also seems quite powerful and fun! Fran?ois. From spautz2 at telefaks.biz Mon Nov 7 12:23:22 2011 From: spautz2 at telefaks.biz (David) Date: Mon, 07 Nov 2011 10:23:22 +0100 Subject: [Freeswitch-users] CHANNEL_CALLSTATE 'switch_channel_perform_set_callstate' - Informations Message-ID: <4EB7A38A.7070904@telefaks.biz> Hi, If i create a conference via dialplan tool 'conference' like ... , ... the caller_caller_id_name and caller_caller_id_number in the CHANNEL_CALLSTATE Event for Event calling function 'switch_channel_perform_set_callstate' for the OUTBOUND Channel state 'RINGING' was set to 'FreeSWITCH' and '00000000' I cannot set these variables to another value. No set of channel variables via dialplan tools 'set' or via dialstring variables like '{outbound_caller_id=555}555 at default 555 TestConf' or with other variables. How can i set these variables to another value by the dialplan or conference configuration dynamicaly? How can i not ? Gernerally, How can i push more informations to these Event function? Thanks David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/2d9808e5/attachment.html From avi at avimarcus.net Mon Nov 7 13:42:38 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 7 Nov 2011 12:42:38 +0200 Subject: [Freeswitch-users] SIP trunk without password and username In-Reply-To: <4EB79B10.60608@softnet.si> References: <4EB79B10.60608@softnet.si> Message-ID: You are trying to register to somewhere? Then just set whatever for the username/pass, if the other side doesn't ask for authentication then you'll never provide the username/pass. -Avi On Mon, Nov 7, 2011 at 10:47 AM, Miha Zoubek wrote: > Hi, > > we are having all trunks on SBC without usr/pass. > If i left param name and password empty I get ERROR: username param is > REQUIRED! and also same if I delete this param usr/pass. > > How can I configure GW sip trunk without password and username. I would > like to have only IP. > > Thank you! > BR, > Miha > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/6e1eb5fb/attachment.html From miha at softnet.si Mon Nov 7 14:23:37 2011 From: miha at softnet.si (Miha Zoubek) Date: Mon, 07 Nov 2011 12:23:37 +0100 Subject: [Freeswitch-users] SIP trunk without password and username In-Reply-To: References: <4EB79B10.60608@softnet.si> Message-ID: <4EB7BFB9.5090300@softnet.si> Hi @Avi, now I am looking about this gw thing. This is trunk and we have only option without authentication (just ip). Do I need to set up in sip_profiles gw? Or must I just allow it in ACL list and that with dialplan redirect to right phone number? I have put this in acl.conf.xml and I get this after starting freeswitch, so I guess this is ok. 2011-11-07 11:39:30.062345 [NOTICE] switch_utils.c:248 Adding 172.31.1.90/29(allow) [] to list strict 2011-11-07 11:39:30.062351 [NOTICE] switch_core.c:1233 Adding 172.31.1.90/29 (allow) to list strict Freeswitch log (why this ip is rejacted if i put it in ACL?): 2011-11-07 11:32:17.662016 [NOTICE] switch_loadable_module.c:254 Adding Application 'auth_function' 2011-11-07 11:32:30.738908 [DEBUG] sofia.c:7269 IP 172.31.1.90 Rejected by acl "domains". Falling back to Digest auth. P.s.: I tried with some usr/pass but also get error with connection to gw. Thank you for your help! BR, Miha On 11/7/2011 11:42 AM, Avi Marcus wrote: > You are trying to register to somewhere? Then just set whatever for > the username/pass, if the other side doesn't ask for authentication > then you'll never provide the username/pass. > -Avi > > > On Mon, Nov 7, 2011 at 10:47 AM, Miha Zoubek > wrote: > > Hi, > > we are having all trunks on SBC without usr/pass. > If i left param name and password empty I get ERROR: username > param is REQUIRED! and also same if I delete this param usr/pass. > > How can I configure GW sip trunk without password and username. I > would like to have only IP. > > Thank you! > BR, > Miha > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/b39998af/attachment-0001.html From pkelly at gmail.com Mon Nov 7 16:05:23 2011 From: pkelly at gmail.com (Pete Kelly) Date: Mon, 7 Nov 2011 13:05:23 +0000 Subject: [Freeswitch-users] Transcoding iLBC/GSM In-Reply-To: References: Message-ID: Thanks for the pointers... some CLI sessions with DEBUG on and digging round the docs got it sorted. I needed to specify iLBC at 30i in the codec list. Pete On 4 November 2011 16:17, Anthony Minessale wrote: > not enough data, do you have iLBC in your configs and mod_ilbc loaded? > you need more data like a trace "console loglevel debug" "sofia global > siptrace on" > > On Fri, Nov 4, 2011 at 9:47 AM, Pete Kelly wrote: >> >> Hi >> >> I am trying to get freeswitch to bridge 2 calls while maintaining iLBC >> or GSM for legA, and allowing legB to choose its own codec. >> >> This is because legA is on a low bandwidth connection so allowing >> PCMA/PCMU is unwise. >> >> This works fine on a freeswitch install where there is a Sangoma >> transcoding card installed, however when I try it on Freeswitch with >> no Transcode card, it just responds with a 488. >> >> I read in the docs that iLBC is only transcodable at ptime=30, do you >> think this could be the reason the transcode is not happening. The SDP >> looks like this from legA: >> >> v=0. >> o=- 3529404938 3529404938 IN IP4 10.15.20.104. >> s=pjmedia. >> c=IN IP4 188.39.51.2. >> t=0 0. >> m=audio 4213 RTP/AVP 104 3 98 97 99 8 0 96. >> a=rtcp:4001. >> a=rtpmap:104 iLBC/8000. >> a=rtpmap:3 GSM/8000. >> a=rtpmap:98 speex/16000. >> a=rtpmap:97 speex/8000. >> a=rtpmap:99 speex/32000. >> a=rtpmap:8 PCMA/8000. >> a=rtpmap:0 PCMU/8000. >> a=sendrecv. >> a=rtpmap:96 telephone-event/8000. >> a=fmtp:96 0-15. >> >> Anyone got any ideas/pointers I can look at? >> >> Thanks >> >> Pete >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From vipkilla at gmail.com Mon Nov 7 17:08:08 2011 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 Nov 2011 09:08:08 -0500 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: <07BF4904977CC645B485E970424193AD0E689458B8@localhost> References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> <07BF4904977CC645B485E970424193AD0E689458B8@localhost> Message-ID: was FS 3277: http://jira.freeswitch.org/browse/FS-3277 actually committed? If not, why? Seems like a good addition. On Fri, Nov 4, 2011 at 7:24 AM, wrote: > Hi, > > That is the reason. I think FS is able to act as a cluster. So many machines act as one and should have the same name... > Just an idea. > > greets. > > > >>Can I ask when exactly the "switchname" should be used then? >> >>FS 3277: http://jira.freeswitch.org/browse/FS-3277 >> >>Suggests it was implemented for the exact reason of keeping the names of two switches in an HA configuration identical, but it seems they are best >left separate. >> >>I don't mind cleaning up some of the detail in the HA Wiki but would just like to know when the names should be the same and when they should be >different i.e. shoudl the names be the same for kind kind of active / active 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 > From beppe.grillo at gmail.com Mon Nov 7 17:29:07 2011 From: beppe.grillo at gmail.com (Beppe Grillo) Date: Mon, 7 Nov 2011 15:29:07 +0100 Subject: [Freeswitch-users] SIP statistic Message-ID: Hy, I need to retrieve the statistical data of the SIP Protocol. Show me some documentation that you know is this information ? How can I view SIP traffic counters? Thanks, Giuseppe. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/ce4aac33/attachment.html From miha at softnet.si Mon Nov 7 18:03:31 2011 From: miha at softnet.si (Miha Zoubek) Date: Mon, 07 Nov 2011 16:03:31 +0100 Subject: [Freeswitch-users] SIP trunk without password and username In-Reply-To: <4EB7BFB9.5090300@softnet.si> References: <4EB79B10.60608@softnet.si> <4EB7BFB9.5090300@softnet.si> Message-ID: <4EB7F343.7090803@softnet.si> I deleted gw trunk profil sip_profiles. I have add in acl.conf.xml my domain: For xxx.xxx.xxx.xxx I tried to put ip and network but the error was always the same: 2011-11-07 15:18:57.966016 [CONSOLE] switch_core.c:1164 Created ip list domains default (deny) 2011-11-07 15:18:57.966047 [WARNING] switch_core.c:1193 Cannot locate domain xxx.xxx.xxx.xxx/29 P.s.: also add in public folder for dialplan. Please help me out! BR, Miha On 11/7/2011 12:23 PM, Miha Zoubek wrote: > Hi @Avi, > > now I am looking about this gw thing. This is trunk and we have only > option without authentication (just ip). > Do I need to set up in sip_profiles gw? Or must I just allow it in ACL > list and that with dialplan redirect to right phone number? > > I have put this in acl.conf.xml > > > > > > and I get this after starting freeswitch, so I guess this is ok. > > 2011-11-07 11:39:30.062345 [NOTICE] switch_utils.c:248 Adding > 172.31.1.90/29(allow) [] to list strict > 2011-11-07 11:39:30.062351 [NOTICE] switch_core.c:1233 Adding > 172.31.1.90/29 (allow) to list strict > > Freeswitch log (why this ip is rejacted if i put it in ACL?): > > 2011-11-07 11:32:17.662016 [NOTICE] switch_loadable_module.c:254 > Adding Application 'auth_function' > 2011-11-07 11:32:30.738908 [DEBUG] sofia.c:7269 IP 172.31.1.90 > Rejected by acl "domains". Falling back to Digest auth. > > P.s.: I tried with some usr/pass but also get error with connection to gw. > > Thank you for your help! > > BR, > Miha > > > On 11/7/2011 11:42 AM, Avi Marcus wrote: >> You are trying to register to somewhere? Then just set whatever for >> the username/pass, if the other side doesn't ask for authentication >> then you'll never provide the username/pass. >> -Avi >> >> >> On Mon, Nov 7, 2011 at 10:47 AM, Miha Zoubek > > wrote: >> >> Hi, >> >> we are having all trunks on SBC without usr/pass. >> If i left param name and password empty I get ERROR: username >> param is REQUIRED! and also same if I delete this param usr/pass. >> >> How can I configure GW sip trunk without password and username. I >> would like to have only IP. >> >> Thank you! >> BR, >> Miha >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/ff039763/attachment.html From eagle.antonio at gmail.com Mon Nov 7 18:10:50 2011 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Mon, 7 Nov 2011 15:10:50 +0000 Subject: [Freeswitch-users] SIP trunk without password and username In-Reply-To: <4EB7F343.7090803@softnet.si> References: <4EB79B10.60608@softnet.si> <4EB7BFB9.5090300@softnet.si> <4EB7F343.7090803@softnet.si> Message-ID: Miha Add register = false in the gateways params or if your FS is older just add a blank username and password params , the remote SBC will ingore it. 2011/11/7 Miha Zoubek > I deleted gw trunk profil sip_profiles. > > I have add in acl.conf.xml my domain: > > > > > > > > > > For xxx.xxx.xxx.xxx I tried to put ip and network but the error was always > the same: > > 2011-11-07 15:18:57.966016 [CONSOLE] switch_core.c:1164 Created ip list > domains default (deny) > 2011-11-07 15:18:57.966047 [WARNING] switch_core.c:1193 Cannot locate > domain xxx.xxx.xxx.xxx/29 > > P.s.: also add in public folder for dialplan. > > Please help me out! > > BR, > Miha > > > > On 11/7/2011 12:23 PM, Miha Zoubek wrote: > > Hi @Avi, > > now I am looking about this gw thing. This is trunk and we have only > option without authentication (just ip). > Do I need to set up in sip_profiles gw? Or must I just allow it in ACL > list and that with dialplan redirect to right phone number? > > I have put this in acl.conf.xml > > > > > > and I get this after starting freeswitch, so I guess this is ok. > > 2011-11-07 11:39:30.062345 [NOTICE] switch_utils.c:248 Adding > 172.31.1.90/29(allow) [] to list strict > 2011-11-07 11:39:30.062351 [NOTICE] switch_core.c:1233 Adding > 172.31.1.90/29 (allow) to list strict > > Freeswitch log (why this ip is rejacted if i put it in ACL?): > > 2011-11-07 11:32:17.662016 [NOTICE] switch_loadable_module.c:254 Adding > Application 'auth_function' > 2011-11-07 11:32:30.738908 [DEBUG] sofia.c:7269 IP 172.31.1.90 Rejected > by acl "domains". Falling back to Digest auth. > > P.s.: I tried with some usr/pass but also get error with connection to gw. > > Thank you for your help! > > BR, > Miha > > > On 11/7/2011 11:42 AM, Avi Marcus wrote: > > You are trying to register to somewhere? Then just set whatever for the > username/pass, if the other side doesn't ask for authentication then you'll > never provide the username/pass. > -Avi > > > On Mon, Nov 7, 2011 at 10:47 AM, Miha Zoubek wrote: > >> Hi, >> >> we are having all trunks on SBC without usr/pass. >> If i left param name and password empty I get ERROR: username param is >> REQUIRED! and also same if I delete this param usr/pass. >> >> How can I configure GW sip trunk without password and username. I would >> like to have only IP. >> >> Thank you! >> BR, >> Miha >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/f69925df/attachment-0001.html From victor.chukalovskiy at utoronto.ca Mon Nov 7 18:17:06 2011 From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy) Date: Mon, 07 Nov 2011 10:17:06 -0500 Subject: [Freeswitch-users] SIP trunk without password and username In-Reply-To: <4EB7F343.7090803@softnet.si> References: <4EB79B10.60608@softnet.si> <4EB7BFB9.5090300@softnet.si> <4EB7F343.7090803@softnet.si> Message-ID: <4EB7F672.6020902@utoronto.ca> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/72ce7560/attachment.html From crash at klamzi.hu Sun Nov 6 22:16:15 2011 From: crash at klamzi.hu (Kiss Karoly) Date: Sun, 6 Nov 2011 20:16:15 +0100 Subject: [Freeswitch-users] RTP SSRC changes mid session Message-ID: <20111106191615.GA9667@alcor.klamzi.hu> Hello all, I ran into a problem where freeswitch stopped relaying media in bridge mode from leg A to leg B when the endpoint on leg A (Cisco SIP gateway) changed the SSRC in the RTP stream. The first RTP packet with the new SSRC had the Mark bit set. The closest I got when reading up on this issue was RFC3550 Section 8.2, which states that and endpoint may change SSRC but must send an RTCP Bye before doing so. In my case there was no RTCP involved. Am I right to assume that FS stopped relaying media because the SSRC changed ? Is there a magical switch that will tell FS not to stop relaying media in this case ? If there is, are there any caveats ??? Using: FreeSWITCH Version 1.0.head (git-3aeaec7 2011-09-02 23-45-57 +0200) Thanks, Kiss Karoly From trap at 2dayhost.com Mon Nov 7 00:17:59 2011 From: trap at 2dayhost.com (Max Machula) Date: Sun, 06 Nov 2011 21:17:59 +0000 Subject: [Freeswitch-users] date/time condition does not work Message-ID: <4EB6F987.7080909@2dayhost.com> Hello, Hope someone can help me. I have a strange problem with date and time on my FreeSwitch. I have dialplan: ... skip ... Server time: # date Thu Nov 3 21:34:22 GMT 2011 However FreeSwitch can not confirm the year 2011. Log file: Dialplan: sofia/internal/07711567890 at xx.xx.xx.xx parsing [public->WW26] continue=true Dialplan: sofia/internal/07711567890 at xx.xx.xx.xxRegex (PASS) [WW26] destination_number(26) =~ /26/ break=on-false Dialplan: sofia/internal/07711567890 at xx.xx.xx.xxRegex (FAIL) [WW26] year() =~ /2011/ break=on-false I get the same problem if I change field year to wday which set to 1-6 for example. Any ideas would be much appreciated. Thanks, Max -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111106/1643e879/attachment-0001.html From roman at dissauer.net Mon Nov 7 06:33:33 2011 From: roman at dissauer.net (Roman Dissauer) Date: Mon, 7 Nov 2011 11:33:33 +0800 Subject: [Freeswitch-users] one gateway two registrations In-Reply-To: References: <2F46CC4C-70D7-4B94-A229-9E22834D5B64@dissauer.net> Message-ID: <3865297A-2DE6-4E8F-BBC2-56D92652F8A6@dissauer.net> I disabled in my external sip profile, now i have just one registration of the gateway. But now the gateway defined in the directory also regilters over the internal profile. Is there a way to define a gateway in the directory to register over the external sip profile, but the clients from the same domain register over the internal profile? Seems that i have to define my gateways in the sip profiles and not in the directory?! Thanks, Roman On 05.11.2011, at 00:35, Anthony Minessale wrote: > you must have the param to parse the domain set in both gateways > > Comment out the section in the one where you do not want to register. > > > On Wed, Nov 2, 2011 at 10:20 PM, Roman Dissauer wrote: > I have a gateway defined in directory, this works fine but it ties to register over both internal and external sip_profile. How can I restrict it to just the external profile? > > Thanks! > Roman > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/b1258853/attachment-0001.html From jsun at junsun.net Mon Nov 7 08:16:58 2011 From: jsun at junsun.net (Jun Sun) Date: Sun, 06 Nov 2011 21:16:58 -0800 Subject: [Freeswitch-users] possible to run conference calls on ec2 instance? Message-ID: <4EB769CA.7010906@junsun.net> We are thinking to set up a dial-out conference system on aws ec2 instances. The number of lines could range from 3 to 30. Is this feasible? The freeswitch wiki page on ec2 gives very contradicting hints. Anybody has a real first hand experience with this? Also, I did search of public AMI's and could not find any freeswitch AMI. Is this expected? Cheers. Jun From 123rpv at gmail.com Mon Nov 7 11:09:45 2011 From: 123rpv at gmail.com (=?KOI8-R?B?8MHXxcwg4s/OxMHS2A==?=) Date: Mon, 7 Nov 2011 10:09:45 +0200 Subject: [Freeswitch-users] I Have some problems with "sudo make current" Message-ID: I Have some problems with "sudo make current" Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_serialize.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_abyss.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_cgi.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_string.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_struct.c... Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c... Compiling /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c... quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC -o .libs/mod_xml_rpc.o quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o mod_xml_rpc.o >/dev/null 2>&1 Creating mod_xml_rpc.la... make[6]: ????? ?? ???????? `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' make[5]: ????? ?? ???????? `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' make[4]: ????? ?? ???????? `/usr/src/freeswitch/src/mod' make[3]: ????? ?? ???????? `/usr/src/freeswitch/src' make[2]: *** [all-recursive] ?????? 1 make[2]: ????? ?? ???????? `/usr/src/freeswitch' make[1]: *** [all] ?????? 2 make[1]: ????? ?? ???????? `/usr/src/freeswitch' make: *** [current] ?????? 2 How can I fix it????? Ubuntu 10.04 x64 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/b322aeaf/attachment.html From brian at freeswitch.org Mon Nov 7 19:59:05 2011 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Nov 2011 10:59:05 -0600 Subject: [Freeswitch-users] RTP SSRC changes mid session In-Reply-To: <20111106191615.GA9667@alcor.klamzi.hu> References: <20111106191615.GA9667@alcor.klamzi.hu> Message-ID: <3CA74899-058F-47B5-AEE2-A1CF4826E825@freeswitch.org> On Nov 6, 2011, at 1:16 PM, Kiss Karoly wrote: > Am I right to assume that FS stopped relaying media because the SSRC changed ? You assume wrong I think. > > Is there a magical switch that will tell FS not to stop relaying media in this case ? > If there is, are there any caveats ??? > > Using: FreeSWITCH Version 1.0.head (git-3aeaec7 2011-09-02 23-45-57 +0200) You have to update. > > Thanks, > Kiss Karoly -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/13ac104b/attachment.html From msc at freeswitch.org Mon Nov 7 20:05:08 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Nov 2011 09:05:08 -0800 Subject: [Freeswitch-users] Automatic fax formating sending via Freeswitch In-Reply-To: References: <40650EFDEA0042788AF84CBE8236745A@e1705> Message-ID: There are WYSIWYG editors for Tex, and there are also command line tools to convert Tex to PDF. Just understand that Tex is a bit unforgiving so plan on spending some time getting to know it. However, once you are familiar with it you'll probably find that it is quite powerful and handy. -MC On Sat, Nov 5, 2011 at 11:29 PM, Yehavi Bourvine wrote: > Thanks, but that's the final stage. I am looking for the step befoer > convert: A "language" that is textual and has decent WISIWIG editing tools; > I want to take this "language" and feed it to ImageMagic to get the final > image. > > Thanks, __Yehavi: > > 2011/11/3 Madovsky > >> ** >> convert from Imagemagick >> >> ----- Original Message ----- >> *From:* Yehavi Bourvine >> *To:* FreeSWITCH Users Help >> *Sent:* Thursday, November 03, 2011 10:30 AM >> *Subject:* [Freeswitch-users] Automatic fax formating sending via >> Freeswitch >> >> Hello, >> >> This is not really a FreeSwitch question, but I guess people here might >> have answer to this... >> >> I have to generate fax from two files: template and list of fields to >> replace in the template. I would like the template to be a tectual thing so >> I can use sed to replace the fields in it, and then convert the result to >> PFD/TIFF and send it via FS. >> >> Is there some tectual format (like Tex, SVG) that can be eaily created by >> some WISIWIG software? >> >> 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/20111107/88167fbc/attachment.html From freeswitch at peely.com Mon Nov 7 20:07:00 2011 From: freeswitch at peely.com (peely) Date: Mon, 7 Nov 2011 09:07:00 -0800 (PST) Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> <07BF4904977CC645B485E970424193AD0E689458B8@localhost> Message-ID: <1320685620061-6971178.post@n2.nabble.com> It was committed and works, but the point is that using it as described in the Jira commit (i.e. using it in a HA failover cluster) has strange behaviour. As Anthony described, it's actually best not using this and using separate hostnames for each half of the cluster, the failover still works perfectly as both switches read pertinent data from a shared database even though their hostnames are different. There may be other reasons for using a switchname which is different to the machine's hostname and if so, then switchname in switch.conf.xml works as described. I've gone back to my test setup and am re-testing failover in test conditions, when I've put this into a live scenario i.e. everything works I'll look at the Wiki article and see if anything needs updating, although Avi has rightly noted there that the failover works with different hostnames. Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6971178.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at peely.com Mon Nov 7 20:07:11 2011 From: freeswitch at peely.com (peely) Date: Mon, 7 Nov 2011 09:07:11 -0800 (PST) Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> <07BF4904977CC645B485E970424193AD0E689458B8@localhost> Message-ID: <1320685631151-6971180.post@n2.nabble.com> It was committed and works, but the point is that using it as described in the Jira commit (i.e. using it in a HA failover cluster) has strange behaviour. As Anthony described, it's actually best not using this and using separate hostnames for each half of the cluster, the failover still works perfectly as both switches read pertinent data from a shared database even though their hostnames are different. There may be other reasons for using a switchname which is different to the machine's hostname and if so, then switchname in switch.conf.xml works as described. I've gone back to my test setup and am re-testing failover in test conditions, when I've put this into a live scenario i.e. everything works I'll look at the Wiki article and see if anything needs updating, although Avi has rightly noted there that the failover works with different hostnames. Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6971180.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at peely.com Mon Nov 7 20:07:27 2011 From: freeswitch at peely.com (peely) Date: Mon, 7 Nov 2011 09:07:27 -0800 (PST) Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> <07BF4904977CC645B485E970424193AD0E689458B8@localhost> Message-ID: <1320685647104-6971181.post@n2.nabble.com> It was committed and works, but the point is that using it as described in the Jira commit (i.e. using it in a HA failover cluster) has strange behaviour. As Anthony described, it's actually best not using this and using separate hostnames for each half of the cluster, the failover still works perfectly as both switches read pertinent data from a shared database even though their hostnames are different. There may be other reasons for using a switchname which is different to the machine's hostname and if so, then switchname in switch.conf.xml works as described. I've gone back to my test setup and am re-testing failover in test conditions, when I've put this into a live scenario i.e. everything works I'll look at the Wiki article and see if anything needs updating, although Avi has rightly noted there that the failover works with different hostnames. Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6971181.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at peely.com Mon Nov 7 20:07:46 2011 From: freeswitch at peely.com (peely) Date: Mon, 7 Nov 2011 09:07:46 -0800 (PST) Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> <07BF4904977CC645B485E970424193AD0E689458B8@localhost> Message-ID: <1320685666394-6971182.post@n2.nabble.com> It was committed and works, but the point is that using it as described in the Jira commit (i.e. using it in a HA failover cluster) has strange behaviour. As Anthony described, it's actually best not using this and using separate hostnames for each half of the cluster, the failover still works perfectly as both switches read pertinent data from a shared database even though their hostnames are different. There may be other reasons for using a switchname which is different to the machine's hostname and if so, then switchname in switch.conf.xml works as described. I've gone back to my test setup and am re-testing failover in test conditions, when I've put this into a live scenario i.e. everything works I'll look at the Wiki article and see if anything needs updating, although Avi has rightly noted there that the failover works with different hostnames. Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6971182.html Sent from the freeswitch-users mailing list archive at Nabble.com. From infos at madovsky.org Mon Nov 7 20:24:36 2011 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Nov 2011 12:24:36 -0500 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the callstable? References: <1320324355799-6958891.post@n2.nabble.com><1320399957355-6962201.post@n2.nabble.com><1320404772101-6962394.post@n2.nabble.com><07BF4904977CC645B485E970424193AD0E689458B8@localhost> Message-ID: <851FD28188EF43AF8D2FA23BDB02729A@e1705> cluster principle is a group of nodes. each node has a different name of course.. ----- Original Message ----- From: "Vik Killa" To: "FreeSWITCH Users Help" Sent: Monday, November 07, 2011 9:08 AM Subject: Re: [Freeswitch-users] HA: Starting FreeSWITCH empties the callstable? > was FS 3277: http://jira.freeswitch.org/browse/FS-3277 actually committed? > If not, why? Seems like a good addition. > > On Fri, Nov 4, 2011 at 7:24 AM, wrote: >> Hi, >> >> That is the reason. I think FS is able to act as a cluster. So many >> machines act as one and should have the same name... >> Just an idea. >> >> greets. >> >> >> >>>Can I ask when exactly the "switchname" should be used then? >>> >>>FS 3277: http://jira.freeswitch.org/browse/FS-3277 >>> >>>Suggests it was implemented for the exact reason of keeping the names of >>>two switches in an HA configuration identical, but it seems they are best >>> >left separate. >>> >>>I don't mind cleaning up some of the detail in the HA Wiki but would just >>>like to know when the names should be the same and when they should be >>> >different i.e. shoudl the names be the same for kind kind of active / >>>active 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 >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 7 20:26:47 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Nov 2011 09:26:47 -0800 Subject: [Freeswitch-users] Problem Verifying an existing file with lua In-Reply-To: References: Message-ID: Thomas, I suspect there is a problem elsewhere in your dialplan. Can you pastebin the whole extension and also a debug log of a call flowing through this extension? I have a feeling that will help us figure out what is going on. Please use pastebin.freeswitch.org and "FreeSWITCH Log" for syntax highlighting. -MC On Sun, Nov 6, 2011 at 3:51 PM, Thomas Hoellriegel wrote: > Hi Sammy, > I become no output true or false. > My Script is now: > > function file_exists(fname) > local f = io.open(fname, "r") > if (f and f:read()) then return true end > end > if ( session:ready() ) then > session:answer( ); > datei = session:getVariable("call_**screen_filename"); > if (file_exists(datei..extension.****.".wav")) then > session:setVariable("fileex", "true"); > session:setVariable(fileex, true) > returns true; > else > session:setVariable("fileex", "false"); > returns false; > end > end > > The variable fileex is emty. > The variable: > call_screen_filename > will not be changed after to execute these script. > > I like to setup a callscreening. > if the name exists, transfer to the given extension. > If the name don.t exists, start the recording and transwer the call. > > Thanks. > > > > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/7a9aa332/attachment.html From avi at avimarcus.net Mon Nov 7 20:28:24 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 7 Nov 2011 19:28:24 +0200 Subject: [Freeswitch-users] date/time condition does not work In-Reply-To: <4EB6F987.7080909@2dayhost.com> References: <4EB6F987.7080909@2dayhost.com> Message-ID: According to http://wiki.freeswitch.org/wiki/Dialplan_XML#Built-In_Variables that is correct.. but, the examples show accessing wday and year by doing: wrote: > Hello, > > Hope someone can help me. > > I have a strange problem with date and time on my FreeSwitch. > > I have dialplan: > > > > > ... skip ... > > > > Server time: > # date > Thu Nov 3 21:34:22 GMT 2011 > > However FreeSwitch can not confirm the year 2011. Log file: > Dialplan: sofia/internal/07711567890 at xx.xx.xx.xx parsing [public->WW26] > continue=true > Dialplan: sofia/internal/07711567890 at xx.xx.xx.xx Regex (PASS) [WW26] > destination_number(26) =~ /26/ break=on-false > Dialplan: sofia/internal/07711567890 at xx.xx.xx.xx Regex (FAIL) [WW26] > year() =~ /2011/ break=on-false > > I get the same problem if I change field year to wday which set to 1-6 for > example. > > Any ideas would be much appreciated. > > Thanks, > Max > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/65c3ab99/attachment.html From vipkilla at gmail.com Mon Nov 7 20:37:20 2011 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 Nov 2011 12:37:20 -0500 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the calls table? In-Reply-To: <1320685620061-6971178.post@n2.nabble.com> References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> <07BF4904977CC645B485E970424193AD0E689458B8@localhost> <1320685620061-6971178.post@n2.nabble.com> Message-ID: Is it recommended to set the hostname the same with 2 active FS servers? Where/how do i set this variable? On Mon, Nov 7, 2011 at 12:07 PM, peely wrote: > It was committed and works, but the point is that using it as described in > the Jira commit (i.e. using it in a HA failover cluster) has strange > behaviour. > > As Anthony described, it's actually best not using this and using separate > hostnames for each half of the cluster, the failover still works perfectly > as both switches read pertinent data from a shared database even though > their hostnames are different. There may be other reasons for using a > switchname which is different to the machine's hostname and if so, then > switchname in switch.conf.xml works as described. > > I've gone back to my test setup and am re-testing failover in test > conditions, when I've put this into a live scenario i.e. everything works > I'll look at the Wiki article and see if anything needs updating, although > Avi has rightly noted there that the failover works with different > hostnames. > > > Neil. > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6971178.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 infos at madovsky.org Mon Nov 7 20:53:15 2011 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Nov 2011 12:53:15 -0500 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the callstable? References: <1320324355799-6958891.post@n2.nabble.com><1320399957355-6962201.post@n2.nabble.com><1320404772101-6962394.post@n2.nabble.com><07BF4904977CC645B485E970424193AD0E689458B8@localhost><1320685620061-6971178.post@n2.nabble.com> Message-ID: <4915D11A76D14084A50132739870EE50@e1705> depend what you call "set the hostname". in FS config, yes, outside FS, no ----- Original Message ----- From: "Vik Killa" To: "FreeSWITCH Users Help" Sent: Monday, November 07, 2011 12:37 PM Subject: Re: [Freeswitch-users] HA: Starting FreeSWITCH empties the callstable? > Is it recommended to set the hostname the same with 2 active FS servers? > Where/how do i set this variable? > > On Mon, Nov 7, 2011 at 12:07 PM, peely wrote: >> It was committed and works, but the point is that using it as described >> in >> the Jira commit (i.e. using it in a HA failover cluster) has strange >> behaviour. >> >> As Anthony described, it's actually best not using this and using >> separate >> hostnames for each half of the cluster, the failover still works >> perfectly >> as both switches read pertinent data from a shared database even though >> their hostnames are different. There may be other reasons for using a >> switchname which is different to the machine's hostname and if so, then >> switchname in switch.conf.xml works as described. >> >> I've gone back to my test setup and am re-testing failover in test >> conditions, when I've put this into a live scenario i.e. everything works >> I'll look at the Wiki article and see if anything needs updating, >> although >> Avi has rightly noted there that the failover works with different >> hostnames. >> >> >> Neil. >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6971178.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 msc at freeswitch.org Mon Nov 7 20:58:23 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Nov 2011 09:58:23 -0800 Subject: [Freeswitch-users] CHANNEL_CALLSTATE 'switch_channel_perform_set_callstate' - Informations In-Reply-To: <4EB7A38A.7070904@telefaks.biz> References: <4EB7A38A.7070904@telefaks.biz> Message-ID: I believe you want to set the various conference_auto_outcall_xxx chan vars. See this page: http://wiki.freeswitch.org/wiki/Conference_set_auto_outcall -MC On Mon, Nov 7, 2011 at 1:23 AM, David wrote: > ** > Hi, > > If i create a conference via dialplan tool 'conference' like > > data="{sip_auto_answer=false}sofia/internal/220"/> > ... > , > ... > > the caller_caller_id_name and caller_caller_id_number in the > CHANNEL_CALLSTATE Event for Event calling function > 'switch_channel_perform_set_callstate' for the OUTBOUND Channel state > 'RINGING' was set to 'FreeSWITCH' and '00000000' > > I cannot set these variables to another value. No set of channel variables > via dialplan tools 'set' or via dialstring variables like > '{outbound_caller_id=555}555 at default 555 TestConf' or with other > variables. > How can i set these variables to another value by the dialplan or > conference configuration dynamicaly? How can i not ? > > Gernerally, How can i push more informations to these Event function? > > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/40a1852f/attachment.html From vipkilla at gmail.com Mon Nov 7 20:59:09 2011 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 Nov 2011 12:59:09 -0500 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties the callstable? In-Reply-To: <4915D11A76D14084A50132739870EE50@e1705> References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> <07BF4904977CC645B485E970424193AD0E689458B8@localhost> <1320685620061-6971178.post@n2.nabble.com> <4915D11A76D14084A50132739870EE50@e1705> Message-ID: i mean in the FS config I know the server's hostnames should different How do you set the hostname inside FS? On Mon, Nov 7, 2011 at 12:53 PM, Madovsky wrote: > depend what you call "set the hostname". > in FS config, yes, outside FS, no > > ----- Original Message ----- > From: "Vik Killa" > To: "FreeSWITCH Users Help" > Sent: Monday, November 07, 2011 12:37 PM > Subject: Re: [Freeswitch-users] HA: Starting FreeSWITCH empties the > callstable? > > >> Is it recommended to set the hostname the same with 2 active FS servers? >> Where/how do i set this variable? >> >> On Mon, Nov 7, 2011 at 12:07 PM, peely wrote: >>> It was committed and works, but the point is that using it as described >>> in >>> the Jira commit (i.e. using it in a HA failover cluster) has strange >>> behaviour. >>> >>> As Anthony described, it's actually best not using this and using >>> separate >>> hostnames for each half of the cluster, the failover still works >>> perfectly >>> as both switches read pertinent data from a shared database even though >>> their hostnames are different. There may be other reasons for using a >>> switchname which is different to the machine's hostname and if so, then >>> switchname in switch.conf.xml works as described. >>> >>> I've gone back to my test setup and am re-testing failover in test >>> conditions, when I've put this into a live scenario i.e. everything works >>> I'll look at the Wiki article and see if anything needs updating, >>> although >>> Avi has rightly noted there that the failover works with different >>> hostnames. >>> >>> >>> Neil. >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6971178.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 > From msc at freeswitch.org Mon Nov 7 21:03:42 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Nov 2011 10:03:42 -0800 Subject: [Freeswitch-users] I Have some problems with "sudo make current" In-Reply-To: References: Message-ID: I'm afraid my Russian isn't what it used to be. Can someone translate this? -MC 2011/11/7 ????? ??????? <123rpv at gmail.com> > I Have some problems with "sudo make current" > > Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_serialize.c... > Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_abyss.c... > Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_cgi.c... > Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_string.c... > Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_struct.c... > Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c... > Compiling /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c... > quiet_libtool: compile: gcc -w > -I../../../../libs/xmlrpc-c/lib/expat/xmlparse > -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c > -I../../../../libs/xmlrpc-c/include > -I../../../../libs/xmlrpc-c/lib/abyss/src > -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC > -o .libs/mod_xml_rpc.o > quiet_libtool: compile: gcc -w > -I../../../../libs/xmlrpc-c/lib/expat/xmlparse > -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c > -I../../../../libs/xmlrpc-c/include > -I../../../../libs/xmlrpc-c/lib/abyss/src > -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o > mod_xml_rpc.o >/dev/null 2>&1 > Creating mod_xml_rpc.la... > make[6]: ????? ?? ???????? > `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' > make[5]: ????? ?? ???????? > `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' > make[4]: ????? ?? ???????? `/usr/src/freeswitch/src/mod' > make[3]: ????? ?? ???????? `/usr/src/freeswitch/src' > make[2]: *** [all-recursive] ?????? 1 > make[2]: ????? ?? ???????? `/usr/src/freeswitch' > make[1]: *** [all] ?????? 2 > make[1]: ????? ?? ???????? `/usr/src/freeswitch' > make: *** [current] ?????? 2 > > How can I fix it????? > > Ubuntu 10.04 x64 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/dbbc3c0c/attachment.html From vetali100 at gmail.com Mon Nov 7 21:07:40 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Mon, 7 Nov 2011 10:07:40 -0800 Subject: [Freeswitch-users] I Have some problems with "sudo make current" In-Reply-To: References: Message-ID: Just regular compilation errors at the end. make[6]: Exit from directory `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' make[5]: Exit from directory `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' make[4]: Exit from directory `/usr/src/freeswitch/src/mod' make[3]: Exit from directory `/usr/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Exit from directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Exit from directory `/usr/src/freeswitch' make: *** [current] Error 2 Regards, Vitalie 2011/11/7 ????? ??????? <123rpv at gmail.com> > I Have some problems with "sudo make current" > > Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_serialize.c... > Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_abyss.c... > Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_cgi.c... > Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_string.c... > Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_struct.c... > Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c... > Compiling /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c... > quiet_libtool: compile: gcc -w > -I../../../../libs/xmlrpc-c/lib/expat/xmlparse > -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c > -I../../../../libs/xmlrpc-c/include > -I../../../../libs/xmlrpc-c/lib/abyss/src > -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC > -o .libs/mod_xml_rpc.o > quiet_libtool: compile: gcc -w > -I../../../../libs/xmlrpc-c/lib/expat/xmlparse > -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c > -I../../../../libs/xmlrpc-c/include > -I../../../../libs/xmlrpc-c/lib/abyss/src > -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o > mod_xml_rpc.o >/dev/null 2>&1 > Creating mod_xml_rpc.la... > make[6]: ????? ?? ???????? > `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' > make[5]: ????? ?? ???????? > `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' > make[4]: ????? ?? ???????? `/usr/src/freeswitch/src/mod' > make[3]: ????? ?? ???????? `/usr/src/freeswitch/src' > make[2]: *** [all-recursive] ?????? 1 > make[2]: ????? ?? ???????? `/usr/src/freeswitch' > make[1]: *** [all] ?????? 2 > make[1]: ????? ?? ???????? `/usr/src/freeswitch' > make: *** [current] ?????? 2 > > How can I fix it????? > > Ubuntu 10.04 x64 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/fa859d9c/attachment-0001.html From msc at freeswitch.org Mon Nov 7 21:08:16 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Nov 2011 10:08:16 -0800 Subject: [Freeswitch-users] SIP statistic In-Reply-To: References: Message-ID: Ciao Giuseppe! FreeSWITCH does not store SIP statistical data. You'd need a 3rd party application to do that. I know that Homer is one tool for gathering SIP information but it's not really designed for statistics. Anyone else have a suggestion for Giuseppe? -MC On Mon, Nov 7, 2011 at 6:29 AM, Beppe Grillo wrote: > Hy, > I need to retrieve the statistical data of the SIP Protocol. > Show me some documentation that you know is this information ? > How can I view SIP traffic counters? > > Thanks, > Giuseppe. > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/1ef6164c/attachment.html From infos at madovsky.org Mon Nov 7 21:10:30 2011 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Nov 2011 13:10:30 -0500 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties thecallstable? References: <1320324355799-6958891.post@n2.nabble.com><1320399957355-6962201.post@n2.nabble.com><1320404772101-6962394.post@n2.nabble.com><07BF4904977CC645B485E970424193AD0E689458B8@localhost><1320685620061-6971178.post@n2.nabble.com><4915D11A76D14084A50132739870EE50@e1705> Message-ID: use xml flags like the example in internal.conf.xml ----- Original Message ----- From: "Vik Killa" To: "FreeSWITCH Users Help" Sent: Monday, November 07, 2011 12:59 PM Subject: Re: [Freeswitch-users] HA: Starting FreeSWITCH empties thecallstable? >i mean in the FS config > I know the server's hostnames should different > > How do you set the hostname inside FS? > > On Mon, Nov 7, 2011 at 12:53 PM, Madovsky wrote: >> depend what you call "set the hostname". >> in FS config, yes, outside FS, no >> >> ----- Original Message ----- >> From: "Vik Killa" >> To: "FreeSWITCH Users Help" >> Sent: Monday, November 07, 2011 12:37 PM >> Subject: Re: [Freeswitch-users] HA: Starting FreeSWITCH empties the >> callstable? >> >> >>> Is it recommended to set the hostname the same with 2 active FS servers? >>> Where/how do i set this variable? >>> >>> On Mon, Nov 7, 2011 at 12:07 PM, peely wrote: >>>> It was committed and works, but the point is that using it as described >>>> in >>>> the Jira commit (i.e. using it in a HA failover cluster) has strange >>>> behaviour. >>>> >>>> As Anthony described, it's actually best not using this and using >>>> separate >>>> hostnames for each half of the cluster, the failover still works >>>> perfectly >>>> as both switches read pertinent data from a shared database even though >>>> their hostnames are different. There may be other reasons for using a >>>> switchname which is different to the machine's hostname and if so, then >>>> switchname in switch.conf.xml works as described. >>>> >>>> I've gone back to my test setup and am re-testing failover in test >>>> conditions, when I've put this into a live scenario i.e. everything >>>> works >>>> I'll look at the Wiki article and see if anything needs updating, >>>> although >>>> Avi has rightly noted there that the failover works with different >>>> hostnames. >>>> >>>> >>>> Neil. >>>> >>>> -- >>>> View this message in context: >>>> http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6971178.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 From msc at freeswitch.org Mon Nov 7 21:14:44 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Nov 2011 10:14:44 -0800 Subject: [Freeswitch-users] I Have some problems with "sudo make current" In-Reply-To: References: Message-ID: Ah, well, I don't see the actual error in the snippet that was included. Can you pastebin the whole output? That will be easier. Use pastebin.freeswitch.org and put the URL in this email thread. -MC 2011/11/7 Vitalie Colosov > Just regular compilation errors at the end. > > make[6]: Exit from directory > `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' > make[5]: Exit from > directory `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' > make[4]: Exit from directory `/usr/src/freeswitch/src/mod' > make[3]: Exit from directory `/usr/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Exit from directory `/usr/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Exit from directory `/usr/src/freeswitch' > make: *** [current] Error 2 > > Regards, > Vitalie > > > 2011/11/7 ????? ??????? <123rpv at gmail.com> > >> I Have some problems with "sudo make current" >> >> Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_serialize.c... >> Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_abyss.c... >> Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_cgi.c... >> Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_string.c... >> Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_struct.c... >> Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c... >> Compiling /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c... >> quiet_libtool: compile: gcc -w >> -I../../../../libs/xmlrpc-c/lib/expat/xmlparse >> -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c >> -I../../../../libs/xmlrpc-c/include >> -I../../../../libs/xmlrpc-c/lib/abyss/src >> -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ >> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror >> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >> -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic >> -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC >> -o .libs/mod_xml_rpc.o >> quiet_libtool: compile: gcc -w >> -I../../../../libs/xmlrpc-c/lib/expat/xmlparse >> -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c >> -I../../../../libs/xmlrpc-c/include >> -I../../../../libs/xmlrpc-c/lib/abyss/src >> -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ >> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror >> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >> -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic >> -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o >> mod_xml_rpc.o >/dev/null 2>&1 >> Creating mod_xml_rpc.la... >> make[6]: ????? ?? ???????? >> `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' >> make[5]: ????? ?? ???????? >> `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' >> make[4]: ????? ?? ???????? `/usr/src/freeswitch/src/mod' >> make[3]: ????? ?? ???????? `/usr/src/freeswitch/src' >> make[2]: *** [all-recursive] ?????? 1 >> make[2]: ????? ?? ???????? `/usr/src/freeswitch' >> make[1]: *** [all] ?????? 2 >> make[1]: ????? ?? ???????? `/usr/src/freeswitch' >> make: *** [current] ?????? 2 >> >> How can I fix it????? >> >> Ubuntu 10.04 x64 >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/8955fbe7/attachment.html From vipkilla at gmail.com Mon Nov 7 21:16:23 2011 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 Nov 2011 13:16:23 -0500 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties thecallstable? In-Reply-To: References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> <07BF4904977CC645B485E970424193AD0E689458B8@localhost> <1320685620061-6971178.post@n2.nabble.com> <4915D11A76D14084A50132739870EE50@e1705> Message-ID: I'm using FS in a multi-tenant env, so i have disabled On Mon, Nov 7, 2011 at 1:10 PM, Madovsky wrote: > use > xml flags like the example in internal.conf.xml > > > ----- Original Message ----- > From: "Vik Killa" > To: "FreeSWITCH Users Help" > Sent: Monday, November 07, 2011 12:59 PM > Subject: Re: [Freeswitch-users] HA: Starting FreeSWITCH empties > thecallstable? > > >>i mean in the FS config >> I know the server's hostnames should different >> >> How do you set the hostname inside FS? >> >> On Mon, Nov 7, 2011 at 12:53 PM, Madovsky wrote: >>> depend what you call "set the hostname". >>> in FS config, yes, outside FS, no >>> >>> ----- Original Message ----- >>> From: "Vik Killa" >>> To: "FreeSWITCH Users Help" >>> Sent: Monday, November 07, 2011 12:37 PM >>> Subject: Re: [Freeswitch-users] HA: Starting FreeSWITCH empties the >>> callstable? >>> >>> >>>> Is it recommended to set the hostname the same with 2 active FS servers? >>>> Where/how do i set this variable? >>>> >>>> On Mon, Nov 7, 2011 at 12:07 PM, peely wrote: >>>>> It was committed and works, but the point is that using it as described >>>>> in >>>>> the Jira commit (i.e. using it in a HA failover cluster) has strange >>>>> behaviour. >>>>> >>>>> As Anthony described, it's actually best not using this and using >>>>> separate >>>>> hostnames for each half of the cluster, the failover still works >>>>> perfectly >>>>> as both switches read pertinent data from a shared database even though >>>>> their hostnames are different. There may be other reasons for using a >>>>> switchname which is different to the machine's hostname and if so, then >>>>> switchname in switch.conf.xml works as described. >>>>> >>>>> I've gone back to my test setup and am re-testing failover in test >>>>> conditions, when I've put this into a live scenario i.e. everything >>>>> works >>>>> I'll look at the Wiki article and see if anything needs updating, >>>>> although >>>>> Avi has rightly noted there that the failover works with different >>>>> hostnames. >>>>> >>>>> >>>>> Neil. >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6971178.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 > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gabe at gundy.org Mon Nov 7 21:21:11 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 7 Nov 2011 11:21:11 -0700 Subject: [Freeswitch-users] SIP statistic In-Reply-To: References: Message-ID: On Mon, Nov 7, 2011 at 11:08 AM, Michael Collins wrote: > FreeSWITCH does not store SIP statistical data. You'd need a 3rd party > application to do that. I know that Homer is one tool for gathering SIP > information but it's not really designed for statistics. > Anyone else have a suggestion for Giuseppe? http://code.google.com/p/homer/ ? It doesn't sound like it's a perfect fit, but you might be able to pull the stats you need out of it. Gabe From gabe at gundy.org Mon Nov 7 21:39:33 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 7 Nov 2011 11:39:33 -0700 Subject: [Freeswitch-users] Lua time condition In-Reply-To: <20111104131334.28165.qmail@community1.interfree.it> References: <20111104131334.28165.qmail@community1.interfree.it> Message-ID: On Fri, Nov 4, 2011 at 7:13 AM, wrote: > I'm a Lua newbie. I'd like to route an incoming call using a Lua script, > according to time. You might consider routing based on the current time before it hits your script. I don't know if that's a good fit for your application, but it's right up the dialplan's ally. http://wiki.freeswitch.org/wiki/Dialplan_XML#Complex_Condition.2FAction_Rules Good luck, Gabe From anthony.minessale at gmail.com Mon Nov 7 22:11:37 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Nov 2011 13:11:37 -0600 Subject: [Freeswitch-users] one gateway two registrations In-Reply-To: <3865297A-2DE6-4E8F-BBC2-56D92652F8A6@dissauer.net> References: <2F46CC4C-70D7-4B94-A229-9E22834D5B64@dissauer.net> <3865297A-2DE6-4E8F-BBC2-56D92652F8A6@dissauer.net> Message-ID: yes you can only put the gateway in the sip profiles too On Sun, Nov 6, 2011 at 9:33 PM, Roman Dissauer wrote: > I disabled in my external sip profile, > now i have just one registration of the gateway. But now the gateway > defined in the directory also regilters over the internal profile. Is there > a way to define a gateway in the directory to register over the external > sip profile, but the clients from the same domain register over the > internal profile? > Seems that i have to define my gateways in the sip profiles and not in the > directory?! > > Thanks, > Roman > > > > On 05.11.2011, at 00:35, Anthony Minessale > wrote: > > you must have the param to parse the domain set in both gateways > > Comment out the section in the one where you do not want to > register. > > > On Wed, Nov 2, 2011 at 10:20 PM, Roman Dissauer wrote: > >> I have a gateway defined in directory, this works fine but it ties to >> register over both internal and external sip_profile. How can I restrict it >> to just the external profile? >> >> Thanks! >> Roman >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/6c6c0624/attachment.html From anthony.minessale at gmail.com Mon Nov 7 22:19:01 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Nov 2011 13:19:01 -0600 Subject: [Freeswitch-users] BLF to show something In-Reply-To: <1320657465.4842.31.camel@luna.madrid.commsmundi.com> References: <4EB75FDA.5070104@freemail.hu> <1320657465.4842.31.camel@luna.madrid.commsmundi.com> Message-ID: you can sit in ESL and listen for PRESENCE_PROBE events and respond with PRESENCE_IN all you want FYI On Mon, Nov 7, 2011 at 3:17 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > On Mon, 2011-11-07 at 05:34 +0100, T?th R?bert wrote: > > I got two idea: > > - sub to dp+ext where ext is the extent you want to monitor > > - use the new mapping feature to map it to a proto. > > A really cool feature would be to have some type of "presenceplan" (like > dialplan or chatplan) called upon SUBSCRIBE. > > It would allow to setup mappings to local extensions/queues/..., given > any type of condition on the origin and destination, refuse the > subscription given ACLs, send a status fetched from ODBC, or why not > even call an external program to fetch a status every time we send a > NOTIFY. > > More or less, "The Ultimate Presence Server"... > > I would note that the new mapping feature also seems quite powerful and > fun! > > 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/31b69970/attachment.html From msc at freeswitch.org Mon Nov 7 22:32:16 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Nov 2011 11:32:16 -0800 Subject: [Freeswitch-users] Timezone with time conditions In-Reply-To: References: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com> Message-ID: How are you passing the timezone info to the dialplan? Are you using an offset? Are you using the timezone name? These are important questions. If you use the timezone name then the strftime_tz API command will let you compute the local time for the given time zone using the time zone database, including any known DST values. From there it's just some basic string manipulation and calculations. Note: The dialplan TOD conditions do not account for TZ so you'd need to do your own calculations. It's completely doable, though, as long as there is a valid TZ name supplied somehow. -MC On Sat, Nov 5, 2011 at 9:12 AM, Avi Marcus wrote: > The general thing is a GUI that allows users to create a time condition. > If you're running an ITSP, rather than a PBX for one company, then your > customers will likely be spanning across time zones. > It's one thing to normalize the times to your local time zones.. But that > doesn't take DST into account. > So basically, the best way would seem to be to pass the time zone to the > condition that evaluates TOD. > When I looked at this a while ago, I couldn't figure out how to do it.. > > -Avi > > > On Fri, Nov 4, 2011 at 7:20 PM, Michael Collins wrote: > >> Can you expand upon this question a bit? My guess is that "anything's >> possible" but I'd like to know more about what you're doing before making >> any suggestions. >> >> -MC >> >> >> On Thu, Nov 3, 2011 at 2:11 AM, Spencer Thomason < >> spencer at 5ninesolutions.com> wrote: >> >>> Hello all, >>> Has anyone found an elegant solution to using time conditions with >>> multiple locations? For example suppose you have a multi tenant system >>> with time conditions, is there any way to set the call time zone on a per >>> call basis? >>> >>> Thanks, >>> Spencer >>> >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/e5d64124/attachment.html From avi at avimarcus.net Mon Nov 7 22:37:37 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 7 Nov 2011 21:37:37 +0200 Subject: [Freeswitch-users] Timezone with time conditions In-Reply-To: References: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com> Message-ID: > > Note: The dialplan TOD conditions do not account for TZ I think that's the point of this thread. Is there a point to re-invent the wheel doing TOD conditions manually? -Avi On Mon, Nov 7, 2011 at 9:32 PM, Michael Collins wrote: > How are you passing the timezone info to the dialplan? Are you using an > offset? Are you using the timezone name? These are important questions. If > you use the timezone name then the strftime_tz API command will let you > compute the local time for the given time zone using the time zone database, > including any known DST values. From there it's just some basic string > manipulation and calculations. Note: The dialplan TOD conditions do not > account for TZ so you'd need to do your own calculations. It's completely > doable, though, as long as there is a valid TZ name supplied somehow. > -MC > > > > On Sat, Nov 5, 2011 at 9:12 AM, Avi Marcus wrote: >> >> The general thing is a GUI that allows users to create a time condition. >> If you're running an ITSP, rather than a PBX for one company, then your >> customers will likely be spanning across time zones. >> It's one thing to normalize the times to your local time zones.. But that >> doesn't take DST into account. >> So basically, the best way would seem to be to pass the time zone to the >> condition that evaluates TOD. >> When I looked at this a while ago, I couldn't figure out how to do it.. >> -Avi >> >> On Fri, Nov 4, 2011 at 7:20 PM, Michael Collins >> wrote: >>> >>> Can you expand upon this question a bit? My guess is that "anything's >>> possible" but I'd like to know more about what you're doing before making >>> any suggestions. >>> -MC >>> >>> On Thu, Nov 3, 2011 at 2:11 AM, Spencer Thomason >>> wrote: >>>> >>>> Hello all, >>>> Has anyone found an elegant solution to using time conditions with >>>> multiple locations? For example suppose you have a multi tenant system with >>>> time conditions, is there any way to set the call time zone on a per call >>>> basis? >>>> >>>> Thanks, >>>> Spencer >>>> >>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/f8cde679/attachment-0001.html From anthony.minessale at gmail.com Mon Nov 7 22:47:56 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Nov 2011 13:47:56 -0600 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties thecallstable? In-Reply-To: References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> <07BF4904977CC645B485E970424193AD0E689458B8@localhost> <1320685620061-6971178.post@n2.nabble.com> <4915D11A76D14084A50132739870EE50@e1705> Message-ID: switchname param in switch.conf.xml as explained above............... On Mon, Nov 7, 2011 at 12:16 PM, Vik Killa wrote: > I'm using FS in a multi-tenant env, so i have name="force-register-domain" value="$${domain}"/> disabled > > On Mon, Nov 7, 2011 at 1:10 PM, Madovsky wrote: > > use > > xml flags like the example in internal.conf.xml > > > > > > ----- Original Message ----- > > From: "Vik Killa" > > To: "FreeSWITCH Users Help" > > Sent: Monday, November 07, 2011 12:59 PM > > Subject: Re: [Freeswitch-users] HA: Starting FreeSWITCH empties > > thecallstable? > > > > > >>i mean in the FS config > >> I know the server's hostnames should different > >> > >> How do you set the hostname inside FS? > >> > >> On Mon, Nov 7, 2011 at 12:53 PM, Madovsky wrote: > >>> depend what you call "set the hostname". > >>> in FS config, yes, outside FS, no > >>> > >>> ----- Original Message ----- > >>> From: "Vik Killa" > >>> To: "FreeSWITCH Users Help" > >>> Sent: Monday, November 07, 2011 12:37 PM > >>> Subject: Re: [Freeswitch-users] HA: Starting FreeSWITCH empties the > >>> callstable? > >>> > >>> > >>>> Is it recommended to set the hostname the same with 2 active FS > servers? > >>>> Where/how do i set this variable? > >>>> > >>>> On Mon, Nov 7, 2011 at 12:07 PM, peely wrote: > >>>>> It was committed and works, but the point is that using it as > described > >>>>> in > >>>>> the Jira commit (i.e. using it in a HA failover cluster) has strange > >>>>> behaviour. > >>>>> > >>>>> As Anthony described, it's actually best not using this and using > >>>>> separate > >>>>> hostnames for each half of the cluster, the failover still works > >>>>> perfectly > >>>>> as both switches read pertinent data from a shared database even > though > >>>>> their hostnames are different. There may be other reasons for using a > >>>>> switchname which is different to the machine's hostname and if so, > then > >>>>> switchname in switch.conf.xml works as described. > >>>>> > >>>>> I've gone back to my test setup and am re-testing failover in test > >>>>> conditions, when I've put this into a live scenario i.e. everything > >>>>> works > >>>>> I'll look at the Wiki article and see if anything needs updating, > >>>>> although > >>>>> Avi has rightly noted there that the failover works with different > >>>>> hostnames. > >>>>> > >>>>> > >>>>> Neil. > >>>>> > >>>>> -- > >>>>> View this message in context: > >>>>> > http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-empties-the-calls-table-tp6958891p6971178.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 > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/ce956653/attachment.html From vipkilla at gmail.com Mon Nov 7 22:56:48 2011 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 Nov 2011 14:56:48 -0500 Subject: [Freeswitch-users] HA: Starting FreeSWITCH empties thecallstable? In-Reply-To: References: <1320324355799-6958891.post@n2.nabble.com> <1320399957355-6962201.post@n2.nabble.com> <1320404772101-6962394.post@n2.nabble.com> <07BF4904977CC645B485E970424193AD0E689458B8@localhost> <1320685620061-6971178.post@n2.nabble.com> <4915D11A76D14084A50132739870EE50@e1705> Message-ID: I didn't see an explanation or example of how to set it anywhere but i added to autoload_configs/switch.conf.xml under and it set the FS hostname to yourmom.com Thanks! On Mon, Nov 7, 2011 at 2:47 PM, Anthony Minessale wrote: > switchname param in switch.conf.xml as?explained?above............... > From msc at freeswitch.org Tue Nov 8 01:10:38 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Nov 2011 14:10:38 -0800 Subject: [Freeswitch-users] Timezone with time conditions In-Reply-To: References: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com> Message-ID: On Mon, Nov 7, 2011 at 11:37 AM, Avi Marcus wrote: > Note: The dialplan TOD conditions do not account for TZ > > I think that's the point of this thread. Is there a point to re-invent the > wheel doing TOD conditions manually? > No, the issue is a matter of supply and demand. The skills needed to add TZ directly into the dialplan are in short supply, whereas doing TZ stuff manually in the dialplan is all but trivial and does not require C coding skills. Also, the demand for TZ checking is relatively low. If it's really desired then a bounty should be put up. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/67925cfe/attachment.html From anthony.minessale at gmail.com Tue Nov 8 01:20:40 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Nov 2011 16:20:40 -0600 Subject: [Freeswitch-users] Timezone with time conditions In-Reply-To: References: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com> Message-ID: on multi home you are better off exploring xml_curl to do time of day stuff, what you suggest would require some new code for sure. On Mon, Nov 7, 2011 at 4:10 PM, Michael Collins wrote: > > On Mon, Nov 7, 2011 at 11:37 AM, Avi Marcus wrote: > >> Note: The dialplan TOD conditions do not account for TZ >> >> I think that's the point of this thread. Is there a point to re-invent >> the wheel doing TOD conditions manually? >> > > No, the issue is a matter of supply and demand. The skills needed to add > TZ directly into the dialplan are in short supply, whereas doing TZ stuff > manually in the dialplan is all but trivial and does not require C coding > skills. Also, the demand for TZ checking is relatively low. > > If it's really desired then a bounty should be put up. > > -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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/4ec31089/attachment-0001.html From anthony.minessale at gmail.com Tue Nov 8 01:39:03 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Nov 2011 16:39:03 -0600 Subject: [Freeswitch-users] Timezone with time conditions In-Reply-To: References: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com> Message-ID: I looked into what the code would require and decided to push a patch, now you get to test and document it. if you get the variable tod_tz_offset set on the channel before it hits the dialplan extension in question it will apply that offset to gmt so you can set it to say, -6 for central time. Note, you must have it set first so you either need to use inline set or maybe it will work if you have already defined it in your user directory. also on a per-tag basis, you can do tz-offset attr ... On Mon, Nov 7, 2011 at 4:20 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > on multi home you are better off exploring xml_curl to do time of day > stuff, what you suggest would require some new code for sure. > > > On Mon, Nov 7, 2011 at 4:10 PM, Michael Collins wrote: > >> >> On Mon, Nov 7, 2011 at 11:37 AM, Avi Marcus wrote: >> >>> Note: The dialplan TOD conditions do not account for TZ >>> >>> I think that's the point of this thread. Is there a point to re-invent >>> the wheel doing TOD conditions manually? >>> >> >> No, the issue is a matter of supply and demand. The skills needed to add >> TZ directly into the dialplan are in short supply, whereas doing TZ stuff >> manually in the dialplan is all but trivial and does not require C coding >> skills. Also, the demand for TZ checking is relatively low. >> >> If it's really desired then a bounty should be put up. >> >> -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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/9b211fbe/attachment.html From trap at 2dayhost.com Tue Nov 8 02:26:27 2011 From: trap at 2dayhost.com (Max Machula) Date: Mon, 07 Nov 2011 23:26:27 +0000 Subject: [Freeswitch-users] date/time condition does not work In-Reply-To: References: <4EB6F987.7080909@2dayhost.com> Message-ID: <4EB86923.4060402@2dayhost.com> Thank you Avi! You were absolutely right! Max On 07/11/2011 17:28, Avi Marcus wrote: > According to > http://wiki.freeswitch.org/wiki/Dialplan_XML#Built-In_Variables that > is correct.. but, the examples show accessing wday and year by doing: > rather than as a "field". > > -Avi > > > On Sun, Nov 6, 2011 at 11:17 PM, Max Machula > wrote: > > Hello, > > Hope someone can help me. > > I have a strange problem with date and time on my FreeSwitch. > > I have dialplan: > > > > > ... skip ... > > > > Server time: > # date > Thu Nov 3 21:34:22 GMT 2011 > > However FreeSwitch can not confirm the year 2011. Log file: > Dialplan: sofia/internal/07711567890 at xx.xx.xx.xx > parsing > [public->WW26] continue=true > Dialplan: sofia/internal/07711567890 at xx.xx.xx.xxRegex (PASS) > [WW26] destination_number(26) =~ /26/ break=on-false > Dialplan: sofia/internal/07711567890 at xx.xx.xx.xxRegex (FAIL) > [WW26] year() =~ /2011/ break=on-false > > I get the same problem if I change field year to wday which set to > 1-6 for example. > > Any ideas would be much appreciated. > > Thanks, > Max > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/52410e78/attachment.html From tyler at phone.com Tue Nov 8 02:40:21 2011 From: tyler at phone.com (Tyler Winter) Date: Mon, 7 Nov 2011 15:40:21 -0800 Subject: [Freeswitch-users] Problem 'adding' video to an audio initiated call, fmtp In-Reply-To: <4EB2A33A.8080504@ewetel.de> References: <539AC2B9-E137-4CAF-BFDC-DED583F66D7F@phone.com> <5544517E-9301-404C-A55E-31B49719E14B@phone.com> <4EB2A33A.8080504@ewetel.de> Message-ID: <6D861557-1C2C-48C0-9E68-9EE3D2C107E4@phone.com> Have you grabbed the pcaps? What did codec negotiation look like before/after you updated your config? I would compare that to your video calls that do work, etc. -Tyler On Nov 3, 2011, at 7:20 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi tyler, > > I tried this tpday, but no success. No video codec processing - only > audio. > > Am 02.11.2011 23:03, schrieb Tyler Winter: >> I was able to add video to a non-video initiated call by using the >> *sip_renegotiate_codec_on_reinvite* setting which ended up being >> elusive. >> >> My understanding of this setting from reviewing the code, is: - the >> default behavior of FS is that once the codec is negotiated, >> re-INVITEs skip the codec negotiation logic and only process >> changes in media IP/port/etc. - by turning this setting on, every >> re-INVITE goes through the same codec negotiation logic as happens >> during the initial INVITE. >> >> Hope this helps someone else too. > > regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.10 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ > > iEYEARECAAYFAk6yozoACgkQ4tZeNddg3dzDtQCbBV8ZoCrOf8wAT5hZg1I52IgW > hFIAn2TYCfU4BjTZZ54AyTwKkujbmsa6 > =JPSB > -----END PGP SIGNATURE----- > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/7893e0d0/attachment.html From spautz2 at telefaks.biz Tue Nov 8 03:18:06 2011 From: spautz2 at telefaks.biz (David) Date: Tue, 08 Nov 2011 01:18:06 +0100 Subject: [Freeswitch-users] CHANNEL_CALLSTATE 'switch_channel_perform_set_callstate' - Informations In-Reply-To: References: <4EB7A38A.7070904@telefaks.biz> Message-ID: <4EB8753E.4090404@telefaks.biz> Thanks. That's what i want Am 07.11.2011 18:58, schrieb Michael Collins: > I believe you want to set the various conference_auto_outcall_xxx chan > vars. See this page: > > http://wiki.freeswitch.org/wiki/Conference_set_auto_outcall > > -MC > > On Mon, Nov 7, 2011 at 1:23 AM, David > wrote: > > Hi, > > If i create a conference via dialplan tool 'conference' like > > data="{sip_auto_answer=false}sofia/internal/220"/> > ... > , > ... > > the caller_caller_id_name and caller_caller_id_number in the > CHANNEL_CALLSTATE Event for Event calling function > 'switch_channel_perform_set_callstate' for the OUTBOUND Channel > state 'RINGING' was set to 'FreeSWITCH' and '00000000' > > I cannot set these variables to another value. No set of channel > variables via dialplan tools 'set' or via dialstring variables > like '{outbound_caller_id=555}555 at default 555 TestConf' or with > other variables. > How can i set these variables to another value by the dialplan or > conference configuration dynamicaly? How can i not ? > > Gernerally, How can i push more informations to these Event function? > > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/6db8c755/attachment-0001.html From adminjew at gmail.com Tue Nov 8 05:12:34 2011 From: adminjew at gmail.com (Yitzchok) Date: Mon, 7 Nov 2011 21:12:34 -0500 Subject: [Freeswitch-users] Automatic fax formating sending via Freeswitch In-Reply-To: References: <40650EFDEA0042788AF84CBE8236745A@e1705> Message-ID: TeX is very powerful but isn't so easy to get started why don't you just use HTML. Yitzchok On Mon, Nov 7, 2011 at 12:05 PM, Michael Collins wrote: > There are WYSIWYG editors for Tex, and there are also command line tools > to convert Tex to PDF. Just understand that Tex is a bit unforgiving so > plan on spending some time getting to know it. However, once you are > familiar with it you'll probably find that it is quite powerful and handy. > > -MC > > > On Sat, Nov 5, 2011 at 11:29 PM, Yehavi Bourvine < > yehavi.bourvine at gmail.com> wrote: > >> Thanks, but that's the final stage. I am looking for the step befoer >> convert: A "language" that is textual and has decent WISIWIG editing tools; >> I want to take this "language" and feed it to ImageMagic to get the final >> image. >> >> Thanks, __Yehavi: >> >> 2011/11/3 Madovsky >> >>> ** >>> convert from Imagemagick >>> >>> ----- Original Message ----- >>> *From:* Yehavi Bourvine >>> *To:* FreeSWITCH Users Help >>> *Sent:* Thursday, November 03, 2011 10:30 AM >>> *Subject:* [Freeswitch-users] Automatic fax formating sending via >>> Freeswitch >>> >>> Hello, >>> >>> This is not really a FreeSwitch question, but I guess people here >>> might have answer to this... >>> >>> I have to generate fax from two files: template and list of fields to >>> replace in the template. I would like the template to be a tectual thing so >>> I can use sed to replace the fields in it, and then convert the result to >>> PFD/TIFF and send it via FS. >>> >>> Is there some tectual format (like Tex, SVG) that can be eaily created >>> by some WISIWIG software? >>> >>> 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 >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111107/23b969e7/attachment.html From yehavi.bourvine at gmail.com Tue Nov 8 08:12:45 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Nov 2011 07:12:45 +0200 Subject: [Freeswitch-users] Automatic fax formating sending via Freeswitch In-Reply-To: References: <40650EFDEA0042788AF84CBE8236745A@e1705> Message-ID: I tried HTML but did not find a decent Linux command line tool to convert it to PDF/TIFF... Thanks! __Yehavi: 2011/11/8 Yitzchok > TeX is very powerful but isn't so easy to get started why don't you just > use HTML. > > Yitzchok > > > > On Mon, Nov 7, 2011 at 12:05 PM, Michael Collins wrote: > >> There are WYSIWYG editors for Tex, and there are also command line tools >> to convert Tex to PDF. Just understand that Tex is a bit unforgiving so >> plan on spending some time getting to know it. However, once you are >> familiar with it you'll probably find that it is quite powerful and handy. >> >> -MC >> >> >> On Sat, Nov 5, 2011 at 11:29 PM, Yehavi Bourvine < >> yehavi.bourvine at gmail.com> wrote: >> >>> Thanks, but that's the final stage. I am looking for the step befoer >>> convert: A "language" that is textual and has decent WISIWIG editing tools; >>> I want to take this "language" and feed it to ImageMagic to get the final >>> image. >>> >>> Thanks, __Yehavi: >>> >>> 2011/11/3 Madovsky >>> >>>> ** >>>> convert from Imagemagick >>>> >>>> ----- Original Message ----- >>>> *From:* Yehavi Bourvine >>>> *To:* FreeSWITCH Users Help >>>> *Sent:* Thursday, November 03, 2011 10:30 AM >>>> *Subject:* [Freeswitch-users] Automatic fax formating sending via >>>> Freeswitch >>>> >>>> Hello, >>>> >>>> This is not really a FreeSwitch question, but I guess people here >>>> might have answer to this... >>>> >>>> I have to generate fax from two files: template and list of fields to >>>> replace in the template. I would like the template to be a tectual thing so >>>> I can use sed to replace the fields in it, and then convert the result to >>>> PFD/TIFF and send it via FS. >>>> >>>> Is there some tectual format (like Tex, SVG) that can be eaily created >>>> by some WISIWIG software? >>>> >>>> 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 >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/5e06a437/attachment.html From infos at madovsky.org Tue Nov 8 08:19:00 2011 From: infos at madovsky.org (Madovsky) Date: Tue, 8 Nov 2011 00:19:00 -0500 Subject: [Freeswitch-users] Automatic fax formating sending viaFreeswitch References: <40650EFDEA0042788AF84CBE8236745A@e1705> Message-ID: <2DC019D73EE840CA975E354C32470438@e1705> do you know linux ghost ? gs as command line ----- Original Message ----- From: Yehavi Bourvine To: FreeSWITCH Users Help Sent: Tuesday, November 08, 2011 12:12 AM Subject: Re: [Freeswitch-users] Automatic fax formating sending viaFreeswitch I tried HTML but did not find a decent Linux command line tool to convert it to PDF/TIFF... Thanks! __Yehavi: 2011/11/8 Yitzchok TeX is very powerful but isn't so easy to get started why don't you just use HTML. Yitzchok On Mon, Nov 7, 2011 at 12:05 PM, Michael Collins wrote: There are WYSIWYG editors for Tex, and there are also command line tools to convert Tex to PDF. Just understand that Tex is a bit unforgiving so plan on spending some time getting to know it. However, once you are familiar with it you'll probably find that it is quite powerful and handy. -MC On Sat, Nov 5, 2011 at 11:29 PM, Yehavi Bourvine wrote: Thanks, but that's the final stage. I am looking for the step befoer convert: A "language" that is textual and has decent WISIWIG editing tools; I want to take this "language" and feed it to ImageMagic to get the final image. Thanks, __Yehavi: 2011/11/3 Madovsky convert from Imagemagick ----- Original Message ----- From: Yehavi Bourvine To: FreeSWITCH Users Help Sent: Thursday, November 03, 2011 10:30 AM Subject: [Freeswitch-users] Automatic fax formating sending via Freeswitch Hello, This is not really a FreeSwitch question, but I guess people here might have answer to this... I have to generate fax from two files: template and list of fields to replace in the template. I would like the template to be a tectual thing so I can use sed to replace the fields in it, and then convert the result to PFD/TIFF and send it via FS. Is there some tectual format (like Tex, SVG) that can be eaily created by some WISIWIG software? 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/5ee1d80e/attachment-0001.html From 12ukwn at gmail.com Tue Nov 8 08:35:15 2011 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Tue, 8 Nov 2011 06:35:15 +0100 Subject: [Freeswitch-users] Automatic fax formating sending via Freeswitch In-Reply-To: References: <40650EFDEA0042788AF84CBE8236745A@e1705> Message-ID: <20111108063515.2db0cee4@anubis.defcon1> On Tue, 8 Nov 2011 07:12:45 +0200 Yehavi Bourvine wrote: > I tried HTML but did not find a decent Linux command line tool to convert > it to PDF/TIFF... I use wkhtmltopdf and it rarely miss, even w/ complicated page. AND its syntax is easy (regular use): wkhtmltopdf myscrappyfiletomethatilike.html bioutyfulpresentation.pdf -- You canna change the laws of physics, Captain; I've got to have thirty minutes! From yehavi.bourvine at gmail.com Tue Nov 8 08:36:47 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Nov 2011 07:36:47 +0200 Subject: [Freeswitch-users] Automatic fax formating sending viaFreeswitch In-Reply-To: <2DC019D73EE840CA975E354C32470438@e1705> References: <40650EFDEA0042788AF84CBE8236745A@e1705> <2DC019D73EE840CA975E354C32470438@e1705> Message-ID: I was not aware that gs can read HTML files... Will give it a try. Thanks, __Yehavi: 2011/11/8 Madovsky > ** > do you know linux ghost ? gs as command line > > ----- Original Message ----- > *From:* Yehavi Bourvine > *To:* FreeSWITCH Users Help > *Sent:* Tuesday, November 08, 2011 12:12 AM > *Subject:* Re: [Freeswitch-users] Automatic fax formating sending > viaFreeswitch > > I tried HTML but did not find a decent Linux command line tool to > convert it to PDF/TIFF... > > Thanks! __Yehavi: > > 2011/11/8 Yitzchok > >> TeX is very powerful but isn't so easy to get started why don't you just >> use HTML. >> >> Yitzchok >> >> >> >> On Mon, Nov 7, 2011 at 12:05 PM, Michael Collins wrote: >> >>> There are WYSIWYG editors for Tex, and there are also command line tools >>> to convert Tex to PDF. Just understand that Tex is a bit unforgiving so >>> plan on spending some time getting to know it. However, once you are >>> familiar with it you'll probably find that it is quite powerful and handy. >>> >>> -MC >>> >>> >>> On Sat, Nov 5, 2011 at 11:29 PM, Yehavi Bourvine < >>> yehavi.bourvine at gmail.com> wrote: >>> >>>> Thanks, but that's the final stage. I am looking for the step befoer >>>> convert: A "language" that is textual and has decent WISIWIG editing tools; >>>> I want to take this "language" and feed it to ImageMagic to get the final >>>> image. >>>> >>>> Thanks, __Yehavi: >>>> >>>> 2011/11/3 Madovsky >>>> >>>>> ** >>>>> convert from Imagemagick >>>>> >>>>> ----- Original Message ----- >>>>> *From:* Yehavi Bourvine >>>>> *To:* FreeSWITCH Users Help >>>>> *Sent:* Thursday, November 03, 2011 10:30 AM >>>>> *Subject:* [Freeswitch-users] Automatic fax formating sending via >>>>> Freeswitch >>>>> >>>>> Hello, >>>>> >>>>> This is not really a FreeSwitch question, but I guess people here >>>>> might have answer to this... >>>>> >>>>> I have to generate fax from two files: template and list of fields to >>>>> replace in the template. I would like the template to be a tectual thing so >>>>> I can use sed to replace the fields in it, and then convert the result to >>>>> PFD/TIFF and send it via FS. >>>>> >>>>> Is there some tectual format (like Tex, SVG) that can be eaily created >>>>> by some WISIWIG software? >>>>> >>>>> 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 >>>> >>>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/ada1e121/attachment.html From infos at madovsky.org Tue Nov 8 09:11:03 2011 From: infos at madovsky.org (Madovsky) Date: Tue, 8 Nov 2011 01:11:03 -0500 Subject: [Freeswitch-users] Automatic fax formating sending viaFreeswitch References: <40650EFDEA0042788AF84CBE8236745A@e1705><2DC019D73EE840CA975E354C32470438@e1705> Message-ID: <522B566D569D4A5890E663136D8BEAA0@e1705> Yehavi, another source http://www.go2linux.org/linux/2010/10/convert-html-pdf-linux-785 From: Yehavi Bourvine To: FreeSWITCH Users Help Sent: Tuesday, November 08, 2011 12:36 AM Subject: Re: [Freeswitch-users] Automatic fax formating sending viaFreeswitch I was not aware that gs can read HTML files... Will give it a try. Thanks, __Yehavi: 2011/11/8 Madovsky do you know linux ghost ? gs as command line ----- Original Message ----- From: Yehavi Bourvine To: FreeSWITCH Users Help Sent: Tuesday, November 08, 2011 12:12 AM Subject: Re: [Freeswitch-users] Automatic fax formating sending viaFreeswitch I tried HTML but did not find a decent Linux command line tool to convert it to PDF/TIFF... Thanks! __Yehavi: 2011/11/8 Yitzchok TeX is very powerful but isn't so easy to get started why don't you just use HTML. Yitzchok On Mon, Nov 7, 2011 at 12:05 PM, Michael Collins wrote: There are WYSIWYG editors for Tex, and there are also command line tools to convert Tex to PDF. Just understand that Tex is a bit unforgiving so plan on spending some time getting to know it. However, once you are familiar with it you'll probably find that it is quite powerful and handy. -MC On Sat, Nov 5, 2011 at 11:29 PM, Yehavi Bourvine wrote: Thanks, but that's the final stage. I am looking for the step befoer convert: A "language" that is textual and has decent WISIWIG editing tools; I want to take this "language" and feed it to ImageMagic to get the final image. Thanks, __Yehavi: 2011/11/3 Madovsky convert from Imagemagick ----- Original Message ----- From: Yehavi Bourvine To: FreeSWITCH Users Help Sent: Thursday, November 03, 2011 10:30 AM Subject: [Freeswitch-users] Automatic fax formating sending via Freeswitch Hello, This is not really a FreeSwitch question, but I guess people here might have answer to this... I have to generate fax from two files: template and list of fields to replace in the template. I would like the template to be a tectual thing so I can use sed to replace the fields in it, and then convert the result to PFD/TIFF and send it via FS. Is there some tectual format (like Tex, SVG) that can be eaily created by some WISIWIG software? 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------- FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/009ab624/attachment-0001.html From Stefan.Weigel at allianz-warranty.com Tue Nov 8 10:18:31 2011 From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan) Date: Tue, 8 Nov 2011 08:18:31 +0100 Subject: [Freeswitch-users] Polycom - trigger reload of config files from tftp Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36F97@AZWSMS03.azwarranty.int> Hi list, I'm currently playing around with call forwarding and stuff. I'm asking myself if there's a way to trigger a Polycom phone (Soundpoint IP 560) to reload the config files, especially the overrides/-phone.cfg. Background: I would like to have the Polycoms do call forwarding, so if I change the files in the background I need to trigger a reload of this file. I know about the 'SIP notify' message but this causes the phone to restart. Is there a possibility to set the call forwarding remotely (changing config file, accessing phone through an interface, etc.) ? Thanks and best regards, Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/7880d525/attachment.html From fieldpeak at gmail.com Tue Nov 8 10:30:35 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Tue, 8 Nov 2011 15:30:35 +0800 Subject: [Freeswitch-users] paramter "NDLB-force-rport" did not effect Message-ID: Dear Friends, i met a problem that call a registered user behind NAT, i need FS send INVITE message to the port of the NAT device from which the registered user sent out REGISTRATION but not according to the CONTACT header. see the wiki, i enable below paramter, >From the wireshark trace, it shows FS received the REGISTERATION message from remote port 20438, however, the INVITE message still sent to the port(20347) inside "Contact", it looks the paramter "NDLB-force-rport" did not effect, Could anyone help for this issue, thanks a lot! Below is the registeration info from FS output (press F9), Call-ID: 1721091027 at 10.33.48.203 User: 13580358068 at 124.193.106.104 Contact: "user" Agent: Status: Registered(UDP)(unknown) EXP(2011-11-08 14:38:02) EXPSECS(152) Host: freeswitch IP: 61.144.248.77 Port: 20438 Auth-User: 13580358068 Auth-Realm: 124.193.106.104 MWI-Account: 13580358068 at 124.193.106.104 Cheers! BR, Charles From miha at softnet.si Tue Nov 8 10:47:24 2011 From: miha at softnet.si (Miha Zoubek) Date: Tue, 08 Nov 2011 08:47:24 +0100 Subject: [Freeswitch-users] SIP trunk without password and username In-Reply-To: <4EB7F672.6020902@utoronto.ca> References: <4EB79B10.60608@softnet.si> <4EB7BFB9.5090300@softnet.si> <4EB7F343.7090803@softnet.si> <4EB7F672.6020902@utoronto.ca> Message-ID: <4EB8DE8C.1090508@softnet.si> @Antonio and @Victor thank you very much! Now it work. Just no let you know. I have also put this subnet to ACL domains part () as otherwise it was automatically denied. BR, Miha On 11/7/2011 4:17 PM, Victor Chukalovskiy wrote: > Hello Miha, > > Usually I do this to accept /incoming/ calls from a specifc IP. No > need to create a gateway: > > Create ACL: > > > > > In your SIP profile make use of this ACL: > > > Then incoming calls will hit your public dial-plan. > > For /outbound/, you'll need to create a gateway with register=false. > > Regards, > Victor > > > On 11/07/2011 10:03 AM, Miha Zoubek wrote: >> I deleted gw trunk profil sip_profiles. >> >> I have add in acl.conf.xml my domain: >> >> >> >> >> >> >> >> >> >> For xxx.xxx.xxx.xxx I tried to put ip and network but the error was >> always the same: >> >> 2011-11-07 15:18:57.966016 [CONSOLE] switch_core.c:1164 Created ip >> list domains default (deny) >> 2011-11-07 15:18:57.966047 [WARNING] switch_core.c:1193 Cannot locate >> domain xxx.xxx.xxx.xxx/29 >> >> P.s.: also add in public folder for dialplan. >> >> Please help me out! >> >> BR, >> Miha >> >> >> >> On 11/7/2011 12:23 PM, Miha Zoubek wrote: >>> Hi @Avi, >>> >>> now I am looking about this gw thing. This is trunk and we have only >>> option without authentication (just ip). >>> Do I need to set up in sip_profiles gw? Or must I just allow it in >>> ACL list and that with dialplan redirect to right phone number? >>> >>> I have put this in acl.conf.xml >>> >>> >>> >>> >>> >>> and I get this after starting freeswitch, so I guess this is ok. >>> >>> 2011-11-07 11:39:30.062345 [NOTICE] switch_utils.c:248 Adding >>> 172.31.1.90/29(allow) [] to list strict >>> 2011-11-07 11:39:30.062351 [NOTICE] switch_core.c:1233 Adding >>> 172.31.1.90/29 (allow) to list strict >>> >>> Freeswitch log (why this ip is rejacted if i put it in ACL?): >>> >>> 2011-11-07 11:32:17.662016 [NOTICE] switch_loadable_module.c:254 >>> Adding Application 'auth_function' >>> 2011-11-07 11:32:30.738908 [DEBUG] sofia.c:7269 IP 172.31.1.90 >>> Rejected by acl "domains". Falling back to Digest auth. >>> >>> P.s.: I tried with some usr/pass but also get error with connection >>> to gw. >>> >>> Thank you for your help! >>> >>> BR, >>> Miha >>> >>> >>> On 11/7/2011 11:42 AM, Avi Marcus wrote: >>>> You are trying to register to somewhere? Then just set whatever for >>>> the username/pass, if the other side doesn't ask for authentication >>>> then you'll never provide the username/pass. >>>> -Avi >>>> >>>> >>>> On Mon, Nov 7, 2011 at 10:47 AM, Miha Zoubek >>> > wrote: >>>> >>>> Hi, >>>> >>>> we are having all trunks on SBC without usr/pass. >>>> If i left param name and password empty I get ERROR: username >>>> param is REQUIRED! and also same if I delete this param usr/pass. >>>> >>>> How can I configure GW sip trunk without password and username. >>>> I would like to have only IP. >>>> >>>> Thank you! >>>> BR, >>>> Miha >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/0132f822/attachment-0001.html From gabe at gundy.org Tue Nov 8 10:58:11 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 8 Nov 2011 00:58:11 -0700 Subject: [Freeswitch-users] Access to sip_history_info from python In-Reply-To: References: Message-ID: On Tue, Nov 1, 2011 at 4:28 AM, John O'Brien wrote: > I have written a python script that processes calls that are diverted to it. > Its sort of a messaging system that triggers some other python processes that run outside of mod_python. John, Welcome to FreeSWITCH! If this is an app that runs outside of FS and you just need to trigger that code, I think Event Socket would be a much better fit. http://wiki.freeswitch.org/wiki/Event_Socket You'll be able to listen for nearly any type of even that you can think of: http://wiki.freeswitch.org/wiki/Event_List If you're comfortable with Twisted, there are several modules out there to get you going; I like this one [well, we did write it]. https://parseltone.org/browser/trunk/parseltone/eventsocket The Parseltone code base is still taking shape, but it's used to process 1000's of calls everyday. Checking the wiki and Google will yield other options too. > I want to from within the mod_python executed python code have access to the SIP header field History-Info. > I can use this to help in determining how the call ended up been processed by my python code. While you'll be able to get access to this header in the events, I'd consider making use of channel variables to track the path calls have taken. Good luck and happy hacking. Gabe From gabe at gundy.org Tue Nov 8 11:01:02 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 8 Nov 2011 01:01:02 -0700 Subject: [Freeswitch-users] nibblebill SQL update In-Reply-To: References: Message-ID: On Sun, Oct 30, 2011 at 10:55 PM, Madovsky wrote: > little suggestion about cash update?with nibblebill: > make a check of?if nibble_rate?is set to?0, avoid any update > and reduce write SQL request... Can you create a patch and test it? Gabe From infos at madovsky.org Tue Nov 8 11:14:31 2011 From: infos at madovsky.org (Madovsky) Date: Tue, 8 Nov 2011 03:14:31 -0500 Subject: [Freeswitch-users] nibblebill SQL update References: Message-ID: <25E5B851A64446CD9D4EE64105420693@e1705> sorry as I'm not a C programer I can't touch the source code thanks ----- Original Message ----- From: "Gabriel Gunderson" To: "FreeSWITCH Users Help" Sent: Tuesday, November 08, 2011 3:01 AM Subject: Re: [Freeswitch-users] nibblebill SQL update On Sun, Oct 30, 2011 at 10:55 PM, Madovsky wrote: > little suggestion about cash update with nibblebill: > make a check of if nibble_rate is set to 0, avoid any update > and reduce write SQL request... Can you create a patch and test it? Gabe FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nirc at fring.com Tue Nov 8 11:21:24 2011 From: nirc at fring.com (Nir Cohen) Date: Tue, 8 Nov 2011 10:21:24 +0200 Subject: [Freeswitch-users] Phantom Calls In-Reply-To: References: <754CEFADFC65654BAC484896D31C2C030566AC0D76@exchangesrv2007.ONEFONE.MSDOMAIN> Message-ID: <754CEFADFC65654BAC484896D31C2C030566AC15EB@exchangesrv2007.ONEFONE.MSDOMAIN> I actually did. I realized that all calls which were "stuck" in my freeswitch had 3 common features. 1. They were using a GSM codec. 2. They were behind a Juniper SRX 3. They were going to a specific provider. I'm going to try and eliminate GSM calls and use only other codecs and see if this solves the problem. Furthermore, I'm going to investigate this problem along with my SysAdmin to see the difference between packets using GSM which go through the SRX and other SIP packets.. to see where they get stuck. I'll update on further development. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vellayappan.N Sent: Monday, October 31, 2011 8:46 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Phantom Calls Hi I am also having the same issue. In my case all the calls have the same issue. When I calling from my mobile to SIP phone in freeswtich, it is ringing to the registered sip number. But it continues to ring the sip number even when i disconnect call in my mobile number. Also the registered SIP is automatically start gets ringing with in regular interval until the freeswitch is restarted. Please let me know if you have addressed this issue. Thanks in advance . On Wed, Oct 26, 2011 at 8:00 PM, Nir Cohen > wrote: Hello, I'm new to freeswitch. Even though I've been using it for quite a while, I hadn't had the chance to really dig into the config/monitoring/debugging. I have a freeswitch which runs flawlessly with a large number of calls but a small percentage of them do not disconnect at all until the freeswitch service is restarted. I was wondering if there's some timeout which I need to define or if there's a way to see a direct reason for this to happen. - I know they are stuck because I did "show calls" and saw calls from different times during the time it was up (see below) - I removed the freeswitch from the DNS to make sure it's not receiving anymore calls. - I opened wireshark and saw that the same amount of calls are referred to as "session in progress" - that is, my fs sends a "Status: 183 session progress..." and the app server returns: (ICMP) "destination unreachable". Thank you. call_created,call_created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid 2011-10-09 15:10:08,1318165808,switch_ivr_signal_bridge,1001,1001,r0018095215036,sofia/external/1001 at 172.16.100.155,edac000a-bc5a-8642-aba8-ed3be16cd762,1001,fring,18095215036,sofia/external/18095215036,487fbf53-4d6c-134e-93bc-31cc3ee1baf8 2011-10-09 23:34:57,1318196097,switch_ivr_signal_bridge,1001,1001,r006498287601,sofia/external/1001 at 172.16.100.155,5d4b4b67-def4-2c44-addc-73f3366d6f83,1001,fring,6498287601,sofia/external/6498287601,6087cfe3-e9b6-de44-b38e-bc86d83ff95a 2011-10-09 23:38:30,1318196310,switch_ivr_signal_bridge,1001,1001,r0020472321728,sofia/external/1001 at 172.16.100.155,9913ff13-33ac-9d4b-b601-d761e49b81b6,1001,fring,20472321728,sofia/external/20472321728,954fff0a-4b7c-8444-8821-306dc6729e42 2011-10-09 23:41:09,1318196469,switch_ivr_signal_bridge,1001,1001,r0020108866522,sofia/external/1001 at 172.16.100.155,f6590d4f-bda0-2146-a309-f2e485ff231c,1001,fring,20108866522,sofia/external/20108866522,ba1f72d8-49fc-5145-8376-87e9f4a3b3b4 2011-10-10 02:16:36,1318205796,switch_ivr_signal_bridge,1001,1001,r0050687102330,sofia/external/1001 at 172.16.100.155,c48e48c1-590d-f84a-a34a-e2da40574d59,1001,fring,50687102330,sofia/external/50687102330,95c4c053-c925-5c42-9ac8-87cace55694b 2011-10-10 06:09:18,1318219758,switch_ivr_signal_bridge,1001,1001,r00919744973603,sofia/external/1001 at 172.16.100.155,04127975-c3f9-b949-907e-f915197bf4fc,1001,fring,919744973603,sofia/external/919744973603,f86302d2-e7ab-0940-b927-b673544c7241 2011-10-10 06:21:24,1318220484,switch_ivr_signal_bridge,1001,1001,r00917204674056,sofia/external/1001 at 172.16.100.155,c835ad85-9e04-b84d-993e-426261686cc7,1001,fring,917204674056,sofia/external/917204674056,1304eae6-4def-c743-9214-89991e6f3350 2011-10-10 06:53:04,1318222384,switch_ivr_signal_bridge,1001,1001,r00919715155054,sofia/external/1001 at 172.16.100.155,c4082f14-7cae-e846-9b05-65a519320619,1001,fring,919715155054,sofia/external/919715155054,16fa4e97-eb15-d24b-907d-f39a7741b493 2011-10-10 07:26:37,1318224397,switch_ivr_signal_bridge,1001,1001,r00919215900646,sofia/external/1001 at 172.16.100.155,40a79a56-cf31-824f-bd47-0006f4abb59b,1001,fring,919215900646,sofia/external/919215900646,f2a02bce-d25a-f448-af39-fa60f3509f51 2011-10-10 09:38:45,1318232325,switch_ivr_signal_bridge,1001,1001,r00919779210235,sofia/external/1001 at 172.16.100.155,3249aa91-d99f-8947-bcd4-749e199ecfbb,1001,fring,919779210235,sofia/external/919779210235,6b55ec96-6fe2-d047-a349-efd44df00c67 2011-10-10 13:22:42,1318245762,switch_ivr_signal_bridge,1001,1001,r00919312238801,sofia/external/1001 at 172.16.100.155,1226ffc9-f338-8b42-be8a-93868dc1ed0a,1001,fring,919312238801,sofia/external/919312238801,f63b997f-2371-2441-b0f3-cfcf0330c87a 2011-10-10 13:37:31,1318246651,switch_ivr_signal_bridge,1001,1001,r0091978999666,sofia/external/1001 at 172.16.100.155,3bf75841-ef52-8244-9c18-5ff783ad5bd4,1001,fring,91978999666,sofia/external/91978999666,8de43ba6-ff36-bf40-941f-a71d3574a95d 2011-10-10 14:15:44,1318248944,switch_ivr_signal_bridge,1001,1001,r00919855283870,sofia/external/1001 at 172.16.100.155,a1844ff2-c9f7-eb4b-a9ac-4e31d5c5a591,1001,fring,919855283870,sofia/external/919855283870,5c73749d-3f35-aa46-a13e-9c1b5d78c592 2011-10-10 14:56:30,1318251390,switch_ivr_signal_bridge,1001,1001,r00919891882161,sofia/external/1001 at 172.16.100.155,9b087d32-2a87-db4f-ac66-3bb91d7afdeb,1001,fring,919891882161,sofia/external/919891882161,0972c900-805d-f949-b47a-fb1398a5739d 2011-10-10 16:31:56,1318257116,switch_ivr_signal_bridge,1001,1001,r00919810399899,sofia/external/1001 at 172.16.100.155,c8b382a4-81ce-6740-bb5d-ebc9ab37d388,1001,fring,919810399899,sofia/external/919810399899,84313caf-fe39-5f45-bf4a-d1cc5e575eff 2011-10-10 16:47:27,1318258047,switch_ivr_signal_bridge,1001,1001,r0096597284481,sofia/external/1001 at 172.16.100.155,b8d93c17-97ee-074e-84c9-103e5259ddb1,1001,fring,96597284481,sofia/external/96597284481,1dee8a27-b3f1-e246-99f6-baca4cea0dd6 2011-10-10 20:11:30,1318270290,switch_ivr_signal_bridge,1001,1001,r00967771771821,sofia/external/1001 at 172.16.100.155,16462169-cb2f-6b42-bf01-64e43a8e4311,1001,fring,967771771821,sofia/external/967771771821,0a98e975-6bd1-4948-8a99-2b682935e893 2011-10-11 01:37:56,1318289876,switch_ivr_signal_bridge,1001,1001,r00584120208712,sofia/external/1001 at 172.16.100.155,7566612f-e7f5-a74a-848b-fa52db667bc1,1001,fring,584120208712,sofia/external/584120208712,bec24956-f8cc-9e46-a1b1-e5f75f827306 2011-10-11 02:23:53,1318292633,switch_ivr_signal_bridge,1001,1001,r00919815288834,sofia/external/1001 at 172.16.100.155,0c7c07a8-ed17-3e43-9d00-5928b998ac48,1001,fring,919815288834,sofia/external/919815288834,11c3f91e-4e84-7d4c-93d9-fbadbc0e0d69 2011-10-11 04:30:11,1318300211,switch_ivr_signal_bridge,1001,1001,r00919924392230,sofia/external/1001 at 172.16.100.155,63514e60-511f-5748-b1ed-6ebcc41d5878,1001,fring,919924392230,sofia/external/919924392230,b465f0f0-f0f5-fb4c-bb0f-93b23b59a4cf 2011-10-11 05:05:39,1318302339,switch_ivr_signal_bridge,1001,1001,r00919703570642,sofia/external/1001 at 172.16.100.155,d6181187-6d71-844a-a4cd-b8add2393c0b,1001,fring,919703570642,sofia/external/919703570642,ed78294e-33f5-7d4d-af4b-40e0efbc9d17 2011-10-11 07:24:57,1318310697,switch_ivr_signal_bridge,1001,1001,r00919845045625,sofia/external/1001 at 172.16.100.155,781823e0-1496-2f46-b6bc-a98cee121887,1001,fring,919845045625,sofia/external/919845045625,bd19f919-4b3c-3342-a23f-53088baedc8f 2011-10-11 07:55:46,1318312546,switch_ivr_signal_bridge,1001,1001,r0084090,sofia/external/1001 at 172.16.100.155,8edd5195-a573-4042-adf7-2c7d093b8178,1001,fring,84090,sofia/external/84090,7e83eb35-d155-384a-ab22-4545e0bb62bb 2011-10-11 10:11:50,1318320710,switch_ivr_signal_bridge,1001,1001,r009197845827258,sofia/external/1001 at 172.16.100.155,0dbff49d-e8cd-f44d-ab68-f0d5babd61a8,1001,fring,9197845827258,sofia/external/9197845827258,d11b3020-6def-2849-aac0-7454c25d5077 2011-10-11 11:28:26,1318325306,switch_ivr_signal_bridge,1001,1001,r00996550675305,sofia/external/1001 at 172.16.100.155,9354bc8e-100d-3e4e-972a-5814fde9ce4c,1001,fring,996550675305,sofia/external/996550675305,cf2662b9-3be1-a440-91f5-7d3088d574bb 2011-10-11 16:43:55,1318344235,switch_ivr_signal_bridge,1001,1001,r00919895419251,sofia/external/1001 at 172.16.100.155,69dab1ec-9d20-4c4d-a8b8-036cc2be4382,1001,fring,919895419251,sofia/external/919895419251,db22ba2d-dff2-1544-a67e-578886921592 2011-10-11 17:48:39,1318348119,switch_ivr_signal_bridge,1001,1001,r00919773212368,sofia/external/1001 at 172.16.100.155,7a35e17e-9604-4743-b131-d339e08245d0,1001,fring,919773212368,sofia/external/919773212368,0cce04b4-3776-4c4f-adf8-e42a1555d33a 2011-10-12 01:57:34,1318377454,switch_ivr_signal_bridge,1001,1001,r0023794311238,sofia/external/1001 at 172.16.100.155,4a09bce4-5342-3943-9b36-a9bd7c51d8bc,1001,fring,23794311238,sofia/external/23794311238,ede10bc2-45b2-1240-a20c-9b3e65868823 2011-10-12 05:39:54,1318390794,switch_ivr_signal_bridge,1001,1001,r00918121275024,sofia/external/1001 at 172.16.100.155,4c991cad-f71d-e644-9946-8fff485c9a40,1001,fring,918121275024,sofia/external/918121275024,4b3e9c9f-bb26-814e-89e5-0685c05692df 2011-10-12 06:39:05,1318394345,switch_ivr_signal_bridge,1001,1001,r00919947389305,sofia/external/1001 at 172.16.100.155,d2a0d36e-182e-bc47-8fe2-83032467eff3,1001,fring,919947389305,sofia/external/919947389305,e5d52ab2-2c36-1843-bea8-0cfdce391e15 2011-10-12 06:59:34,1318395574,switch_ivr_signal_bridge,1001,1001,r00865395760139,sofia/external/1001 at 172.16.100.155,03a603a0-6418-7a46-a1d4-5d4d2357bf23,1001,fring,865395760139,sofia/external/865395760139,ef325309-716c-1246-8f01-513c7e9a93bb 2011-10-12 08:49:22,1318402162,switch_ivr_signal_bridge,1001,1001,r00919944731940,sofia/external/1001 at 172.16.100.155,024d66fc-6582-b04b-a962-56c4db8cb013,1001,fring,919944731940,sofia/external/919944731940,1de9cb6b-4eed-3d4d-89bc-6b533b07849a 2011-10-12 09:37:03,1318405023,switch_ivr_signal_bridge,1001,1001,r00919916780545,sofia/external/1001 at 172.16.100.155,379b8c6e-a998-2649-a577-5f8f1b6b3c4e,1001,fring,919916780545,sofia/external/919916780545,097c1a46-1e65-6749-aa14-592b1cae6fd5 2011-10-12 10:17:08,1318407428,switch_ivr_signal_bridge,1001,1001,r00914562224442,sofia/external/1001 at 172.16.100.155,f91ec6f2-e8a1-b549-86b5-cbb1d17039b3,1001,fring,914562224442,sofia/external/914562224442,3c2ced4c-7b98-4948-8520-8bc357e4a437 2011-10-12 13:54:07,1318420447,switch_ivr_signal_bridge,1001,1001,r00919746980086,sofia/external/1001 at 172.16.100.155,6f1a25be-891e-ee4f-8c70-4ae79694a651,1001,fring,919746980086,sofia/external/919746980086,d0fdad10-f3a2-5347-ab1f-95d4383089b7 2011-10-12 21:04:21,1318446261,switch_ivr_signal_bridge,1001,1001,r0096597280025,sofia/external/1001 at 172.16.100.155,3349eb67-9f4b-f64f-98af-854dd9a07c23,1001,fring,96597280025,sofia/external/96597280025,14d7da2f-d2e6-084c-b5d3-375377400832 2011-10-13 08:38:46,1318487926,switch_ivr_signal_bridge,1001,1001,r00639214286751,sofia/external/1001 at 172.16.100.155,9b909f83-2d31-a145-82bb-1c0dd1f46695,1001,fring,639214286751,sofia/external/639214286751,6bd8bfbf-4391-1e48-ae3d-16c7f81c74a9 2011-10-13 09:28:42,1318490922,switch_ivr_signal_bridge,1001,1001,r00919855874737,sofia/external/1001 at 172.16.100.155,ff74debc-509c-2040-bf50-42313aa6629d,1001,fring,919855874737,sofia/external/919855874737,579ecfbf-d00c-7947-9c2f-6029dbf9b507 2011-10-13 10:44:23,1318495463,switch_ivr_signal_bridge,1001,1001,r00919966842195,sofia/external/1001 at 172.16.100.155,16e5bb3f-5ff1-0743-b787-78fbfb032583,1001,fring,919966842195,sofia/external/919966842195,77d980c9-4ccb-1c40-a901-3e6fd6960a2d 2011-10-13 11:54:10,1318499650,switch_ivr_signal_bridge,1001,1001,r00919751864788,sofia/external/1001 at 172.16.100.155,84725b33-97ab-d64f-8a36-b7f10a2b8723,1001,fring,919751864788,sofia/external/919751864788,f7e86569-01a2-5540-9c9b-3e300f1a0797 2011-10-13 14:26:04,1318508764,switch_ivr_signal_bridge,1001,1001,r00919922440249,sofia/external/1001 at 172.16.100.155,ad0c00ab-17c2-4949-880b-9ff164941f1b,1001,fring,919922440249,sofia/external/919922440249,1d139c8d-82eb-9248-8e81-e9d6315ba50d 2011-10-13 16:10:48,1318515048,switch_ivr_signal_bridge,1001,1001,r0020197596666,sofia/external/1001 at 172.16.100.155,ad64dcd2-232c-d543-8d01-2a513c3dc315,1001,fring,20197596666,sofia/external/20197596666,7f9b2f6c-41db-3840-ab78-e0052c56a885 2011-10-13 20:09:02,1318529342,switch_ivr_signal_bridge,1001,1001,r00551158958193,sofia/external/1001 at 172.16.100.155,0730806a-137f-924e-ace6-9852c72bb185,1001,fring,551158958193,sofia/external/551158958193,de18fc0d-d46f-014c-92f1-ff5ac3d69e47 2011-10-13 21:11:25,1318533085,switch_ivr_signal_bridge,1001,1001,r0020123946125,sofia/external/1001 at 172.16.100.155,74c80972-2dd4-c443-b831-d76003497612,1001,fring,20123946125,sofia/external/20123946125,be927ca2-99cc-4840-aa86-8c9a85e32f80 2011-10-13 22:43:24,1318538604,switch_ivr_signal_bridge,1001,1001,r00919769404307,sofia/external/1001 at 172.16.100.155,16b1cdf4-c7de-2642-a3d1-a901c7168829,1001,fring,919769404307,sofia/external/919769404307,74860ee3-d592-8441-a738-3b82103033a1 2011-10-14 03:50:54,1318557054,switch_ivr_signal_bridge,1001,1001,r00919988274760,sofia/external/1001 at 172.16.100.155,378a4ecc-7646-6147-9086-381a2ffc5570,1001,fring,919988274760,sofia/external/919988274760,49650a00-03dc-054a-ae56-2debe46897a4 2011-10-14 04:13:23,1318558403,switch_ivr_signal_bridge,1001,1001,r00919944010360,sofia/external/1001 at 172.16.100.155,5aec4e1e-ed85-ef41-b9fe-7f7e9666b0da,1001,fring,919944010360,sofia/external/919944010360,39aa0999-6817-1148-98a2-dc8b1cf768c8 2011-10-14 05:28:12,1318562892,switch_ivr_signal_bridge,1001,1001,r00919944191129,sofia/external/1001 at 172.16.100.155,f4e6f3dc-543a-0a4e-b81d-58c8e5434b0a,1001,fring,919944191129,sofia/external/919944191129,b702994a-2237-a442-89c3-1026966a1feb 2011-10-14 06:14:54,1318565694,switch_ivr_signal_bridge,1001,1001,r0060148144168,sofia/external/1001 at 172.16.100.155,95f21383-70db-2848-a1a4-bf1e279970cb,1001,fring,60148144168,sofia/external/60148144168,f9c3747a-4c39-aa4d-a655-7cea1222ada9 2011-10-14 06:40:42,1318567242,switch_ivr_signal_bridge,1001,1001,r008615952583367,sofia/external/1001 at 172.16.100.155,d4976a2b-d218-a14a-ab73-ed5252d23223,1001,fring,8615952583367,sofia/external/8615952583367,d514508a-e4c8-f14f-82fc-43b71560f565 2011-10-14 13:30:22,1318591822,switch_ivr_signal_bridge,1001,1001,r00919902730542,sofia/external/1001 at 172.16.100.155,535a6926-431e-154b-bed1-bea1ea8240a0,1001,fring,919902730542,sofia/external/919902730542,45a49ed1-b426-5f4d-8da6-5d7ae5934dc9 2011-10-14 13:52:34,1318593154,switch_ivr_signal_bridge,1001,1001,r008613986690504,sofia/external/1001 at 172.16.100.155,39dd44b8-1854-be40-8cd4-81c34ebbcc11,1001,fring,8613986690504,sofia/external/8613986690504,7d641511-0cd1-c54c-8058-568f7f0f67f0 2011-10-14 14:44:34,1318596274,switch_ivr_signal_bridge,1001,1001,r0020101,sofia/external/1001 at 172.16.100.155,08408567-1407-ea4a-be73-ce0219738962,1001,fring,20101,sofia/external/20101,aa1cbd89-f2f1-114a-a0a4-2d18fb647a1d 2011-10-14 19:30:55,1318613455,switch_ivr_signal_bridge,1001,1001,r00919953239015,sofia/external/1001 at 172.16.100.155,931bd3ec-a4a9-1b47-b6a3-7226cf83dc3b,1001,fring,919953239015,sofia/external/919953239015,37b98044-fbe2-e940-a368-fe9556fb810f 2011-10-15 01:42:58,1318635778,switch_ivr_signal_bridge,1001,1001,r00919032587213,sofia/external/1001 at 172.16.100.155,11537992-801d-eb42-ba4e-763fbff0f381,1001,fring,919032587213,sofia/external/919032587213,4d2e98a0-d4b4-d046-aa6c-11c7d7137c63 2011-10-15 05:32:32,1318649552,switch_ivr_signal_bridge,1001,1001,r00919885688639,sofia/external/1001 at 172.16.100.155,b2b70ad1-0754-8c49-82eb-c0ae5d051618,1001,fring,919885688639,sofia/external/919885688639,a35b33e7-08c0-f34f-b664-2c1b68be3e95 2011-10-15 06:40:56,1318653656,switch_ivr_signal_bridge,1001,1001,r00919886690120,sofia/external/1001 at 172.16.100.155,13a08888-c378-d949-8625-95e2239bac45,1001,fring,919886690120,sofia/external/919886690120,ff62df61-8354-0641-a1ce-cba74fe53448 2011-10-15 09:51:20,1318665080,switch_ivr_signal_bridge,1001,1001,r00265888891340,sofia/external/1001 at 172.16.100.155,41c4263a-18f2-d147-8b7a-3a14739eeb4f,1001,fring,265888891340,sofia/external/265888891340,1c19dbaa-d35c-3e4d-a913-355d2f9b6c0d 2011-10-15 10:06:41,1318666001,switch_ivr_signal_bridge,1001,1001,r00919820158704,sofia/external/1001 at 172.16.100.155,c1363ab8-2905-2549-876c-dd3cbc05563f,1001,fring,919820158704,sofia/external/919820158704,660b2675-974e-a249-8a24-349020a7449a 2011-10-15 11:19:21,1318670361,switch_ivr_signal_bridge,1001,1001,r0020197105643,sofia/external/1001 at 172.16.100.155,cc2f1ede-7d0c-8845-b455-4eb97ed2b12f,1001,fring,20197105643,sofia/external/20197105643,81cb54ff-b63d-6c4e-afa9-bfea68a553a9 2011-10-15 12:14:13,1318673653,switch_ivr_signal_bridge,1001,1001,r00918870183328,sofia/external/1001 at 172.16.100.155,81ba52e8-d157-4441-bfe1-c314e4f1b4c7,1001,fring,918870183328,sofia/external/918870183328,72306599-6017-c54a-a291-9b3eb1650d83 2011-10-15 14:15:42,1318680942,switch_ivr_signal_bridge,1001,1001,r00919539499709,sofia/external/1001 at 172.16.100.155,b24a486b-b798-c048-b0c5-9c624b0ebade,1001,fring,919539499709,sofia/external/919539499709,456681fa-c2bf-004f-875a-4d4b18ea1e99 2011-10-15 23:23:12,1318713792,switch_ivr_signal_bridge,1001,1001,r0020140088523,sofia/external/1001 at 172.16.100.155,c2f98b57-0c7b-1b41-a77c-05a34ab99b79,1001,fring,20140088523,sofia/external/20140088523,f681a50f-d263-a942-a48f-ab7eb2adff38 2011-10-16 02:16:44,1318724204,switch_ivr_signal_bridge,1001,1001,r00919886859264,sofia/external/1001 at 172.16.100.155,764923b2-29b4-fd4d-9de8-cd89f2654a9f,1001,fring,919886859264,sofia/external/919886859264,ecbe5234-bb44-1a40-a93c-4fd36307f3e9 2011-10-16 06:50:47,1318740647,switch_ivr_signal_bridge,1001,1001,r00919915919879,sofia/external/1001 at 172.16.100.155,27d55f39-86e3-f146-ad9b-2f77938ad868,1001,fring,919915919879,sofia/external/919915919879,2df4fa27-2896-6f4d-81ce-97532b2dee0c 2011-10-16 10:02:03,1318752123,switch_ivr_signal_bridge,1001,1001,r00965555751124,sofia/external/1001 at 172.16.100.155,7d69745c-89e1-ad49-8f57-ca146ecc769a,1001,fring,965555751124,sofia/external/965555751124,973c1c3b-0f50-684a-873b-b61bdd21a245 2011-10-16 11:37:30,1318757850,switch_ivr_signal_bridge,1001,1001,r00201001301440,sofia/external/1001 at 172.16.100.155,22fb4d57-3207-4d4c-8be1-145655dad6a5,1001,fring,201001301440,sofia/external/201001301440,397ee409-504c-244c-b11b-b693070ff7dd 2011-10-16 14:55:05,1318769705,switch_ivr_signal_bridge,1001,1001,r00919742055333,sofia/external/1001 at 172.16.100.155,1817b98a-e036-1746-bd7d-437152ed6b0a,1001,fring,919742055333,sofia/external/919742055333,ab2b04ce-a454-a041-a40e-7ee5ddbb29a4 2011-10-16 16:20:52,1318774852,switch_ivr_signal_bridge,1001,1001,r00998662245737,sofia/external/1001 at 172.16.100.155,986c8dc4-831b-0041-9b3c-1bee42818ab5,1001,fring,998662245737,sofia/external/998662245737,a1eaeb3c-6b1a-ea4c-a3b1-f3144572bf72 2011-10-16 16:51:24,1318776684,switch_ivr_signal_bridge,1001,1001,r008801712054706,sofia/external/1001 at 172.16.100.155,84a34d92-8be5-1644-a9cd-487023eee234,1001,fring,8801712054706,sofia/external/8801712054706,1930550c-d3e3-f645-bda9-e4a0d39a6397 2011-10-16 18:14:22,1318781662,switch_ivr_signal_bridge,1001,1001,r00919390177288,sofia/external/1001 at 172.16.100.155,2878943a-56ee-be47-a5fc-632a92c3734a,1001,fring,919390177288,sofia/external/919390177288,4a8de922-86ab-ba41-8e9c-05a79087a79a 2011-10-16 20:06:31,1318788391,switch_ivr_signal_bridge,1001,1001,r00201224955898,sofia/external/1001 at 172.16.100.155,fc68f803-781d-c347-a6dc-f50a8651f61f,1001,fring,201224955898,sofia/external/201224955898,1e873400-c1ca-da42-be32-837931d2c845 2011-10-16 22:52:28,1318798348,switch_ivr_signal_bridge,1001,1001,r00914567242736,sofia/external/1001 at 172.16.100.155,d8b4385a-bcf9-ad4a-ae00-0539fe7c56f8,1001,fring,914567242736,sofia/external/914567242736,5eb0ca19-155a-3047-a480-8f951f26993f 2011-10-16 23:06:09,1318799169,switch_ivr_signal_bridge,1001,1001,r00966552321650,sofia/external/1001 at 172.16.100.155,f9998988-f7f1-9543-afd5-1cd76e035594,1001,fring,966552321650,sofia/external/966552321650,c01c89f6-212b-5a48-90ce-c79587ecdbac 2011-10-17 00:38:08,1318804688,switch_ivr_signal_bridge,1001,1001,r00911821277451,sofia/external/1001 at 172.16.100.155,54365caa-8f25-8545-9010-db155e4bdde8,1001,fring,911821277451,sofia/external/911821277451,8df3edfb-3b35-a047-8c8d-51d57eeb56b3 2011-10-17 06:02:00,1318824120,switch_ivr_signal_bridge,1001,1001,r00919746008642,sofia/external/1001 at 172.16.100.155,ba807588-4536-d94f-81a7-e1d0e8434e05,1001,fring,919746008642,sofia/external/919746008642,6252a3b1-a0a6-ff42-8188-69e771094104 2011-10-17 06:30:17,1318825817,switch_ivr_signal_bridge,1001,1001,r00919824645363,sofia/external/1001 at 172.16.100.155,2e099d56-5e90-2f48-816f-d66e410fc8b9,1001,fring,919824645363,sofia/external/919824645363,7d18d6e9-dfb9-c949-b00b-5a9e62ecc21c 2011-10-17 07:42:55,1318830175,switch_ivr_signal_bridge,1001,1001,r00919824645363,sofia/external/1001 at 172.16.100.155,54960c11-5352-e34b-ae35-672923413dae,1001,fring,919824645363,sofia/external/919824645363,bbc5cdc4-7a02-2945-b685-28baae5fe280 2011-10-17 10:11:26,1318839086,switch_ivr_signal_bridge,1001,1001,r00919711534925,sofia/external/1001 at 172.16.100.155,583ffded-6229-6345-bd46-4ef09652a8aa,1001,fring,919711534925,sofia/external/919711534925,9d101a1a-0e4e-7144-bff1-a132378a3d9f 2011-10-17 17:50:55,1318866655,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/1001 at 172.16.100.155,b6cd78b7-7da4-b44e-a4f4-60e404fafca3,1001,fring,919988175489,sofia/external/919988175489,602bd41d-3f60-d144-b2bb-36d4da232ddc 2011-10-17 22:52:13,1318884733,switch_ivr_signal_bridge,1001,1001,r00919999233886,sofia/external/1001 at 172.16.100.155,eefd784e-9c04-084f-888b-b0a033a5ee5b,1001,fring,919999233886,sofia/external/919999233886,389a824e-e4c5-b74e-ac8a-9315d9821117 2011-10-17 23:24:02,1318886642,switch_ivr_signal_bridge,1001,1001,r00919313467728,sofia/external/1001 at 172.16.100.155,fcf32cd6-9148-cc40-bc90-534145199ba9,1001,fring,919313467728,sofia/external/919313467728,7d13a986-8454-be4d-a056-c47276ba9f5b 2011-10-18 00:41:19,1318891279,switch_ivr_signal_bridge,1001,1001,r00919747072485,sofia/external/1001 at 172.16.100.155,25f36536-4606-1942-825f-5bfaf28a66ca,1001,fring,919747072485,sofia/external/919747072485,ee6298c7-a6d4-3547-9578-7c95f96ede0f 2011-10-18 01:05:08,1318892708,switch_ivr_signal_bridge,1001,1001,r00919313467728,sofia/external/1001 at 172.16.100.155,8c89657f-9ce7-404a-9d5c-a3559f66b141,1001,fring,919313467728,sofia/external/919313467728,b6a3eb92-6419-4643-930f-e7bebeabb33a 2011-10-18 03:25:04,1318901104,switch_ivr_signal_bridge,1001,1001,r006596112485,sofia/external/1001 at 172.16.100.155,b5189c01-0df2-4e47-af11-8a3e96e247a8,1001,fring,6596112485,sofia/external/6596112485,a834eb93-3744-be49-8165-3b39966c9502 2011-10-18 06:25:34,1318911934,switch_ivr_signal_bridge,1001,1001,r00919876855863,sofia/external/1001 at 172.16.100.155,d6589f30-ba06-5c45-9d20-1e99664a10d8,1001,fring,919876855863,sofia/external/919876855863,4999eac4-e560-fe40-b795-fc5d5cf8ac44 2011-10-18 09:41:35,1318923695,switch_ivr_signal_bridge,1001,1001,r00919922965741,sofia/external/1001 at 172.16.100.155,57335341-bda9-3943-8f0d-59c7a7903c15,1001,fring,919922965741,sofia/external/919922965741,d201c220-2c1f-bb4e-b725-bada9895035b 2011-10-18 12:52:37,1318935157,switch_ivr_signal_bridge,1001,1001,r00919967359493,sofia/external/1001 at 172.16.100.155,66fadc05-a756-a848-bb84-5d39c9998fc5,1001,fring,919967359493,sofia/external/919967359493,71bcd983-d4a4-de4c-b615-f0dbf1095739 2011-10-18 16:41:02,1318948862,switch_ivr_signal_bridge,1001,1001,r00919831094384,sofia/external/1001 at 172.16.100.155,2a96ebc1-a6be-8e49-b508-8b3993b62f9a,1001,fring,919831094384,sofia/external/919831094384,86e74690-fca2-6247-9d6a-c7732445a1be 2011-10-18 21:01:18,1318964478,switch_ivr_signal_bridge,1001,1001,r00919729947506,sofia/external/1001 at 172.16.100.155,96995af7-e82c-3e40-addb-1b62c1f05747,1001,fring,919729947506,sofia/external/919729947506,6a82a0ec-e4a2-be4f-85ee-87b34b4ea3e2 2011-10-19 00:20:24,1318976424,switch_ivr_signal_bridge,1001,1001,r00919605053363,sofia/external/1001 at 172.16.100.155,8f9d6670-2aff-c242-b01e-3d9f0e493d12,1001,fring,919605053363,sofia/external/919605053363,7fd8cee5-cd6b-7b44-848f-8d756f26fd4a 2011-10-19 04:30:30,1318991430,switch_ivr_signal_bridge,1001,1001,r00919814950511,sofia/external/1001 at 172.16.100.155,1a6895e1-57a5-564c-bf08-5780df1d79c7,1001,fring,919814950511,sofia/external/919814950511,595fe521-1673-ea44-8bc3-c9d69c2f2de6 2011-10-19 04:40:27,1318992027,switch_ivr_signal_bridge,1001,1001,r00919814950511,sofia/external/1001 at 172.16.100.155,905be5af-844a-d44f-bbfd-a160ff1f1759,1001,fring,919814950511,sofia/external/919814950511,8a41744e-b19e-954a-a68f-7d16e29976e5 2011-10-19 06:23:43,1318998223,switch_ivr_signal_bridge,1001,1001,r00919990249401,sofia/external/1001 at 172.16.100.155,d009a643-b1b3-3f46-b194-c971e2a17b64,1001,fring,919990249401,sofia/external/919990249401,f26ae13f-de35-2f40-92cb-232449391e3e 2011-10-19 09:38:43,1319009923,switch_ivr_signal_bridge,1001,1001,r00919761333537,sofia/external/1001 at 172.16.100.155,353ede21-8d68-b74e-8711-0ddce2a82928,1001,fring,919761333537,sofia/external/919761333537,d0567650-c2f4-9542-9a20-d8bc1718cc80 2011-10-19 10:59:56,1319014796,switch_ivr_signal_bridge,1001,1001,r00918144890121,sofia/external/1001 at 172.16.100.155,8593dc8b-8d4f-bc40-8e3a-daf85d7c7702,1001,fring,918144890121,sofia/external/918144890121,ec3fda7c-cc8e-e041-b4a0-171928ade343 2011-10-19 13:00:41,1319022041,switch_ivr_signal_bridge,1001,1001,r00201060198003,sofia/external/1001 at 172.16.100.155,bba15b1f-785a-674f-a023-bb8f64f57c53,1001,fring,201060198003,sofia/external/201060198003,2c90f16e-a4ec-6d42-9bd1-4f06bf7c2da9 2011-10-19 13:45:11,1319024711,switch_ivr_signal_bridge,1001,1001,r00914312341722,sofia/external/1001 at 172.16.100.155,845903b7-2770-6847-a116-3d0a8cb7b096,1001,fring,914312341722,sofia/external/914312341722,aa20b33f-eb4d-714a-b236-59afbfe158e7 2011-10-19 19:11:01,1319044261,switch_ivr_signal_bridge,1001,1001,r009845967072,sofia/external/1001 at 172.16.100.155,349ef121-64b9-dc44-a22d-3d12acbb6ece,1001,fring,9845967072,sofia/external/9845967072,d89904a7-daf7-ea45-bae3-57c914e88b48 2011-10-20 02:36:13,1319070973,switch_ivr_signal_bridge,1001,1001,r0060166298037,sofia/external/1001 at 172.16.100.155,f6ff096b-612e-0544-a733-9a566b08fc6d,1001,fring,60166298037,sofia/external/60166298037,1c35f10e-3695-2845-9f4c-dd5a517dd6f7 2011-10-20 03:15:43,1319073343,switch_ivr_signal_bridge,1001,1001,r00919988498802,sofia/external/1001 at 172.16.100.155,1a9da86a-bc16-3e49-9b08-94facc657843,1001,fring,919988498802,sofia/external/919988498802,9430553a-ee00-ed4f-b109-41eabdd4dc7e 2011-10-20 04:35:24,1319078124,switch_ivr_signal_bridge,1001,1001,r00919976131008,sofia/external/1001 at 172.16.100.155,12d0429e-a539-6944-831e-a862183d905b,1001,fring,919976131008,sofia/external/919976131008,f672283c-a37e-e947-b45a-b7e66ef7881a 2011-10-20 05:22:23,1319080943,switch_ivr_signal_bridge,1001,1001,r00919973840848,sofia/external/1001 at 172.16.100.155,42e29646-c0c4-d64a-9cf5-7c37aeabc1a6,1001,fring,919973840848,sofia/external/919973840848,1d2ef437-17aa-db4b-86d9-6d3d5310e280 2011-10-20 05:37:34,1319081854,switch_ivr_signal_bridge,1001,1001,r00919840288799,sofia/external/1001 at 172.16.100.155,f48fed29-8041-3545-bffc-5050bc19cbc6,1001,fring,919840288799,sofia/external/919840288799,df29a0df-3ef8-4b43-807e-73e0c0e25f3b 2011-10-20 10:00:26,1319097626,switch_ivr_signal_bridge,1001,1001,r0091955445666,sofia/external/1001 at 172.16.100.155,931102bc-ac35-3140-acab-988224d502b0,1001,fring,91955445666,sofia/external/91955445666,9283eb63-bd0a-6d43-9d8a-7985ec48e23c 2011-10-20 12:16:27,1319105787,switch_ivr_signal_bridge,1001,1001,r00919855120931,sofia/external/1001 at 172.16.100.155,db0f66aa-69f1-fe47-85af-de610a77d517,1001,fring,919855120931,sofia/external/919855120931,ceab5ae5-9b8f-3d4c-b149-c248567ab5d4 2011-10-20 12:42:02,1319107322,switch_ivr_signal_bridge,1001,1001,r00918872855601,sofia/external/1001 at 172.16.100.155,95249509-e793-0d42-9948-5ef53898b942,1001,fring,918872855601,sofia/external/918872855601,8c9fc203-8d54-aa4f-983f-889fbf0ca883 2011-10-20 13:45:10,1319111110,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/1001 at 172.16.100.155,7b645b14-5f89-a64c-b059-2574b2e4e69d,1001,fring,919988175489,sofia/external/919988175489,55f50da5-5679-fe4f-9a7f-b96111f3cd67 2011-10-20 14:49:50,1319114990,switch_ivr_signal_bridge,1001,1001,r00919663195202,sofia/external/1001 at 172.16.100.155,3601f4b8-c0ff-1a4c-9f76-47e4fa95b549,1001,fring,919663195202,sofia/external/919663195202,a22e2d11-0463-c845-a953-a12b49cfe863 2011-10-20 15:10:47,1319116247,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/1001 at 172.16.100.155,e09d3de0-c759-fc4b-96f1-f01353a4297d,1001,fring,919988175489,sofia/external/919988175489,23d4b490-cbb7-7a45-86af-705c45d07fc7 2011-10-20 20:52:17,1319136737,switch_ivr_signal_bridge,1001,1001,r009888560750,sofia/external/1001 at 172.16.100.155,ff8ab1f6-f1c6-224f-928b-1c042b0ed908,1001,fring,9888560750,sofia/external/9888560750,c37b8ebe-2264-934f-a187-6b14c7dcb308 2011-10-20 22:32:38,1319142758,switch_ivr_signal_bridge,1001,1001,r008801816630096,sofia/external/1001 at 172.16.100.155,a38182cb-b661-854d-b243-a830ab0bf992,1001,fring,8801816630096,sofia/external/8801816630096,f88dc530-303e-d041-83cb-eca118780ca2 2011-10-20 22:34:32,1319142872,switch_ivr_signal_bridge,1001,1001,r00919161209997,sofia/external/1001 at 172.16.100.155,0add3c1c-7fb5-2447-a9d6-d0622886fcf9,1001,fring,919161209997,sofia/external/919161209997,2b91f5de-4668-0244-ac76-d031fe827a25 2011-10-20 23:56:16,1319147776,switch_ivr_signal_bridge,1001,1001,r00985468962,sofia/external/1001 at 172.16.100.155,f7137d11-0a87-6a4b-b5b3-b43eeeb2b462,1001,fring,985468962,sofia/external/985468962,025f197b-c5aa-5b4b-8606-20fb6f17e888 2011-10-21 04:36:00,1319164560,switch_ivr_signal_bridge,1001,1001,r00917597234252,sofia/external/1001 at 172.16.100.155,edd2f16b-9a6a-f749-bd62-4d303bc09442,1001,fring,917597234252,sofia/external/917597234252,1c55b21b-9c27-b647-8ba5-1ffce398c747 2011-10-21 04:47:03,1319165223,switch_ivr_signal_bridge,1001,1001,r00919541622040,sofia/external/1001 at 172.16.100.155,336c146b-c96d-494e-b378-c024bcfc0c94,1001,fring,919541622040,sofia/external/919541622040,746877e1-c509-8a45-934e-f3f4b7f82db2 2011-10-21 07:59:54,1319176794,switch_ivr_signal_bridge,1001,1001,r009613235498,sofia/external/1001 at 172.16.100.155,16575979-c92a-5842-879f-e9f6448c7d54,1001,fring,9613235498,sofia/external/9613235498,bb05fd5a-4a6f-3248-81a9-a81074a9eacd 2011-10-21 08:02:05,1319176925,switch_ivr_signal_bridge,1001,1001,r00912026336917,sofia/external/1001 at 172.16.100.155,ebb234aa-8edf-2e4b-b641-602478b25782,1001,fring,912026336917,sofia/external/912026336917,f6a0e4b1-69f7-dc49-a5bf-41e6d8c7c96d 2011-10-21 08:51:57,1319179917,switch_ivr_signal_bridge,1001,1001,r00919372875359,sofia/external/1001 at 172.16.100.155,b090dc91-932e-1246-abb8-8fc949131ae6,1001,fring,919372875359,sofia/external/919372875359,3a4702a6-e7cc-cb41-ad99-defa80b14680 2011-10-21 09:15:47,1319181347,switch_ivr_signal_bridge,1001,1001,r00919915709069,sofia/external/1001 at 172.16.100.155,ae308194-ce89-4945-b8de-6923ce6d0dc0,1001,fring,919915709069,sofia/external/919915709069,502ffe82-0fb9-b14a-9421-4acd25a0b4df 2011-10-21 11:35:16,1319189716,switch_ivr_signal_bridge,1001,1001,r00919847023725,sofia/external/1001 at 172.16.100.155,c7cb7640-7313-d742-b756-a7713273ccea,1001,fring,919847023725,sofia/external/919847023725,4ab50403-04ac-bc40-a759-40f2a62f1699 2011-10-21 14:48:46,1319201326,switch_ivr_signal_bridge,1001,1001,r00919878738103,sofia/external/1001 at 172.16.100.155,91626ac0-9e14-a34d-94f5-b1fbfe7e7629,1001,fring,919878738103,sofia/external/919878738103,39c58424-b087-6749-9ba9-c9a821f46895 2011-10-21 17:41:28,1319211688,switch_ivr_signal_bridge,1001,1001,r009843648082,sofia/external/1001 at 172.16.100.155,3929dc6b-b7de-c648-8eae-3d749f0ac0fa,1001,fring,9843648082,sofia/external/9843648082,6653a9ed-d23b-564e-98e5-9a56a40e499c 2011-10-22 05:07:21,1319252841,switch_ivr_signal_bridge,1001,1001,r00919036392951,sofia/external/1001 at 172.16.100.155,7cd56510-2983-554a-9fa2-a27c3fb1f9e6,1001,fring,919036392951,sofia/external/919036392951,e63642ec-d43a-c243-9291-0ea6725d04bb 2011-10-22 06:42:56,1319258576,switch_ivr_signal_bridge,1001,1001,r00918870521722,sofia/external/1001 at 172.16.100.155,7dd85bc6-e00e-c44e-bd42-89c6d438485e,1001,fring,918870521722,sofia/external/918870521722,8d7e1c17-2ad9-2447-b91b-d2f2b2e27e4c 2011-10-22 08:36:26,1319265386,switch_ivr_signal_bridge,1001,1001,r00911874289543,sofia/external/1001 at 172.16.100.155,1214a817-f235-1a4e-a21c-d2363e9e5760,1001,fring,911874289543,sofia/external/911874289543,13f85a5d-eb37-0c4a-a15e-2c86dbc34817 2011-10-22 12:14:32,1319278472,switch_ivr_signal_bridge,1001,1001,r00918988262666,sofia/external/1001 at 172.16.100.155,1b5cb871-e043-f244-9942-251a7a492136,1001,fring,918988262666,sofia/external/918988262666,21f328fb-0dfe-2643-957b-302f6a80f17f 2011-10-22 16:28:35,1319293715,switch_ivr_signal_bridge,1001,1001,r00918722222400,sofia/external/1001 at 172.16.100.155,912e1b07-a99b-bf4d-9e2f-0b5990a68821,1001,fring,918722222400,sofia/external/918722222400,8b417629-3161-f141-93d7-bb0ddd7982d5 2011-10-22 17:32:10,1319297530,switch_ivr_signal_bridge,1001,1001,r00919839884297,sofia/external/1001 at 172.16.100.155,ec7decdd-c378-6c45-a888-9b962e4ecece,1001,fring,919839884297,sofia/external/919839884297,f12896b7-9605-9b40-bdc9-face2181146b 2011-10-23 06:26:42,1319344002,switch_ivr_signal_bridge,1001,1001,r00919567382898,sofia/external/1001 at 172.16.100.155,db2e199b-7b9f-fe4d-9742-7415dded69c7,1001,fring,919567382898,sofia/external/919567382898,dd8bb627-c804-2f49-8b5b-5acf5b3423d4 2011-10-23 07:21:29,1319347289,switch_ivr_signal_bridge,1001,1001,r00966537646047,sofia/external/1001 at 172.16.100.155,e97ba53e-1ef6-2944-b53b-b3abe5dfef36,1001,fring,966537646047,sofia/external/966537646047,9eea7d3f-5092-de4b-9291-e7f0a37db2a1 2011-10-23 08:12:33,1319350353,switch_ivr_signal_bridge,1001,1001,r00914942588715,sofia/external/1001 at 172.16.100.155,30ca9d2d-5cae-6645-a16c-6ae8735b6316,1001,fring,914942588715,sofia/external/914942588715,6774d23c-f131-7d41-854d-2fcff77c2cf6 2011-10-23 13:04:33,1319367873,switch_ivr_signal_bridge,1001,1001,r00919625833433,sofia/external/1001 at 172.16.100.155,284691f9-dd3b-cc4a-97c7-9465bd3a6b6c,1001,fring,919625833433,sofia/external/919625833433,4efb7446-11f2-c84c-88c4-10ab97eddedf 2011-10-23 14:35:20,1319373320,switch_ivr_signal_bridge,1001,1001,r00919888880401,sofia/external/1001 at 172.16.100.155,475abe11-2137-c94a-ada7-825ad8524bd8,1001,fring,919888880401,sofia/external/919888880401,5e06efb5-2c72-ba4f-9443-c15568ad2608 2011-10-23 15:59:20,1319378360,switch_ivr_signal_bridge,1001,1001,r009878539161,sofia/external/1001 at 172.16.100.155,aa5c4f8f-5a69-e947-a239-75f4a584b9f0,1001,fring,9878539161,sofia/external/9878539161,366b56d1-3ce4-fa43-9d3d-872c04f5229f 2011-10-23 20:52:23,1319395943,switch_ivr_signal_bridge,1001,1001,r0017787079272,sofia/external/1001 at 172.16.100.155,3f9c8b11-2e7a-e641-8cfc-3b7dc65e7118,1001,fring,17787079272,sofia/external/17787079272,d5b4a2e0-ee48-7048-b54c-f6ea968428af 2011-10-24 10:23:29,1319444609,switch_ivr_signal_bridge,1001,1001,r00919881375673,sofia/external/1001 at 172.16.100.155,527d1ca5-3f05-a54b-8b0c-192b7515eebb,1001,fring,919881375673,sofia/external/919881375673,4d1da2aa-87c1-7d41-b57d-05034e503ede 2011-10-24 10:29:44,1319444984,switch_ivr_signal_bridge,1001,1001,r0061430594924,sofia/external/1001 at 172.16.100.155,6ce52143-219d-ed40-a27f-fc519eef5092,1001,fring,61430594924,sofia/external/61430594924,11ef495a-d651-564e-9479-94fa579a5286 2011-10-24 10:56:32,1319446592,switch_ivr_signal_bridge,1001,1001,r00919003574979,sofia/external/1001 at 172.16.100.155,8d83282b-f96b-0a43-ac58-1f3d3e4e26f0,1001,fring,919003574979,sofia/external/919003574979,2bd6e807-c5ef-8c47-807f-1e5171bcbc3d 2011-10-24 13:58:59,1319457539,switch_ivr_signal_bridge,1001,1001,r00919876395511,sofia/external/1001 at 172.16.100.155,68fdbb19-f76f-0a40-b46b-e0296eed8690,1001,fring,919876395511,sofia/external/919876395511,6cf77cf5-8cd9-f340-98c0-bb896304dbf9 2011-10-24 14:16:45,1319458605,switch_ivr_signal_bridge,1001,1001,r00919988170711,sofia/external/1001 at 172.16.100.155,57e0187b-9d25-044c-bffb-9d11431c251a,1001,fring,919988170711,sofia/external/919988170711,b71d6822-289f-614b-8e5b-2bc20225d9ef 2011-10-25 00:16:09,1319494569,switch_ivr_signal_bridge,1001,1001,r005016630551,sofia/external/1001 at 172.16.100.155,f5e81b48-8f86-4644-b206-cb066e15418d,1001,fring,5016630551,sofia/external/5016630551,b1467ed2-56e3-bb42-ba6c-cdc272b2533b 2011-10-25 02:30:30,1319502630,switch_ivr_signal_bridge,1001,1001,r00919729325377,sofia/external/1001 at 172.16.100.155,f6eb7c48-cc1d-9b42-bbac-46ac759de0e0,1001,fring,919729325377,sofia/external/919729325377,d8eb68b9-ab39-814b-a70d-915347e2afe7 2011-10-25 09:35:05,1319528105,switch_ivr_signal_bridge,1001,1001,r00966554656169,sofia/external/1001 at 172.16.100.155,03c9a95e-4d36-8847-a2a2-3545c38e233e,1001,fring,966554656169,sofia/external/966554656169,1d2927f5-7202-914c-a463-6dcd00588cfc 2011-10-25 12:53:58,1319540038,switch_ivr_signal_bridge,1001,1001,r00201094811534,sofia/external/1001 at 172.16.100.155,3bfe5634-dcf9-c84d-ba72-26f8fee4a8eb,1001,fring,201094811534,sofia/external/201094811534,d0c58bb9-53a3-0549-9b78-4ca5f14fcb63 2011-10-26 04:22:12,1319595732,switch_ivr_signal_bridge,1001,1001,r00913871398117,sofia/external/1001 at 172.16.100.155,1479c10c-9922-854c-8030-6f2a6ce4f605,1001,fring,913871398117,sofia/external/913871398117,fbdc8cf2-e898-0340-bc48-b4ad7c54ac0e 2011-10-26 10:01:27,1319616087,switch_ivr_signal_bridge,1001,1001,r00201113961426,sofia/external/1001 at 172.16.100.155,db18777e-aefa-c248-84ea-06150fe6989f,1001,fring,201113961426,sofia/external/201113961426,0a5572c6-54b7-1d40-bf4e-918e6880b3ce 147 total. FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Regards Vellayappan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/2131821d/attachment-0001.html From benkokakao at gmail.com Tue Nov 8 13:19:53 2011 From: benkokakao at gmail.com (Christian Benke) Date: Tue, 8 Nov 2011 11:19:53 +0100 Subject: [Freeswitch-users] Polycom - trigger reload of config files from tftp In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36F97@AZWSMS03.azwarranty.int> References: <5003D7D3E06F514E8C682F18D223265C04D3B36F97@AZWSMS03.azwarranty.int> Message-ID: > I?m currently playing around with call forwarding and stuff. I?m asking > myself if there?s a way to trigger a Polycom phone (Soundpoint IP 560) to > reload the config files, especially the overrides/-phone.cfg. > > Background: I would like to have the Polycoms do call forwarding, so if I > change the files in the background I need to trigger a reload of this file. > I know about the ?SIP notify? message but this causes the phone to restart. Hi Stefan! I wanted to tell you about the flush_inbound_reg command(http://wiki.freeswitch.org/wiki/Sofia#Flushing_and_rebooting_registered_endpoints), but obviously you already know this option("SIP notify"). I don't think there's another way to remotely reload the config-files without restarting the phone - we would love such an option though to reload the internal phonebook... > Is there a possibility to set the call forwarding remotely (changing config > file, accessing phone through an interface, etc.) ? Regards, Christian From fdelawarde at wirelessmundi.com Tue Nov 8 13:39:21 2011 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 08 Nov 2011 11:39:21 +0100 Subject: [Freeswitch-users] BLF to show something In-Reply-To: References: <4EB75FDA.5070104@freemail.hu> <1320657465.4842.31.camel@luna.madrid.commsmundi.com> Message-ID: <1320748761.5817.32.camel@luna.madrid.commsmundi.com> Of course! The same thing could be done for chat, but the chatplan is still a really great thing to have. I hope one day soon I'll be able to remove my crappy XMPP server and do everything with SIP (voice, video, chat, and presence) using FS and FSComm. Thanks BTW! Fran?ois. On Mon, 2011-11-07 at 13:19 -0600, Anthony Minessale wrote: > you can sit in ESL and listen for PRESENCE_PROBE events and respond > with PRESENCE_IN all you want FYI > > On Mon, Nov 7, 2011 at 3:17 AM, Fran?ois Delawarde > wrote: > On Mon, 2011-11-07 at 05:34 +0100, T?th R?bert wrote: > > I got two idea: > > - sub to dp+ext where ext is the extent you want to monitor > > - use the new mapping feature to map it to a proto. > > > A really cool feature would be to have some type of > "presenceplan" (like > dialplan or chatplan) called upon SUBSCRIBE. > > It would allow to setup mappings to local > extensions/queues/..., given > any type of condition on the origin and destination, refuse > the > subscription given ACLs, send a status fetched from ODBC, or > why not > even call an external program to fetch a status every time we > send a > NOTIFY. > > More or less, "The Ultimate Presence Server"... > > I would note that the new mapping feature also seems quite > powerful and > fun! > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Stefan.Weigel at allianz-warranty.com Tue Nov 8 13:41:02 2011 From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan) Date: Tue, 8 Nov 2011 11:41:02 +0100 Subject: [Freeswitch-users] Polycom - trigger reload of config files from tftp In-Reply-To: References: <5003D7D3E06F514E8C682F18D223265C04D3B36F97@AZWSMS03.azwarranty.int> Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36F9E@AZWSMS03.azwarranty.int> Hi Christian, I know about the command, but it's "only" sending the 'SIP notify'.. Best regards, Stefan -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Christian Benke Gesendet: Dienstag, 8. November 2011 11:20 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Polycom - trigger reload of config files from tftp > I?m currently playing around with call forwarding and stuff. I?m asking > myself if there?s a way to trigger a Polycom phone (Soundpoint IP 560) to > reload the config files, especially the overrides/-phone.cfg. > > Background: I would like to have the Polycoms do call forwarding, so if I > change the files in the background I need to trigger a reload of this file. > I know about the ?SIP notify? message but this causes the phone to restart. Hi Stefan! I wanted to tell you about the flush_inbound_reg command(http://wiki.freeswitch.org/wiki/Sofia#Flushing_and_rebooting_registered_endpoints), but obviously you already know this option("SIP notify"). I don't think there's another way to remotely reload the config-files without restarting the phone - we would love such an option though to reload the internal phonebook... > Is there a possibility to set the call forwarding remotely (changing config > file, accessing phone through an interface, etc.) ? Regards, Christian FreeSWITCH-users mailing list FreeSWITCH-users at lists.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 Tue Nov 8 14:05:55 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Nov 2011 13:05:55 +0200 Subject: [Freeswitch-users] Knowing whether a fax has been sent or not Message-ID: Hello, I am building a system to send faxes with txfax() application. How can I get the result and know whether the fax has been sent or not? ESL, or is there some callback which I enable? Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/4388c87f/attachment.html From Stefan.Weigel at allianz-warranty.com Tue Nov 8 14:08:57 2011 From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan) Date: Tue, 8 Nov 2011 12:08:57 +0100 Subject: [Freeswitch-users] Knowing whether a fax has been sent or not In-Reply-To: References: Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36FA0@AZWSMS03.azwarranty.int> Hi Yehavi, an event will be fired, when a fax is transmitted/received: http://wiki.freeswitch.org/wiki/Mod_spandsp#Events Best regards, Stefan Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Yehavi Bourvine Gesendet: Dienstag, 8. November 2011 12:06 An: FreeSWITCH Users Help Betreff: [Freeswitch-users] Knowing whether a fax has been sent or not Hello, I am building a system to send faxes with txfax() application. How can I get the result and know whether the fax has been sent or not? ESL, or is there some callback which I enable? Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/a1a187f2/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 5972 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/a1a187f2/attachment-0001.bin From yehavi.bourvine at gmail.com Tue Nov 8 14:19:26 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Nov 2011 13:19:26 +0200 Subject: [Freeswitch-users] Knowing whether a fax has been sent or not In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36FA0@AZWSMS03.azwarranty.int> References: <5003D7D3E06F514E8C682F18D223265C04D3B36FA0@AZWSMS03.azwarranty.int> Message-ID: Thanks! __Yehavi: 2011/11/8 Weigel, Stefan > Hi Yehavi,**** > > ** ** > > an event will be fired, when a fax is transmitted/received:**** > > ** ** > > http://wiki.freeswitch.org/wiki/Mod_spandsp#Events**** > > ** ** > > ** ** > > ** ** > > Best regards,**** > > ** ** > > Stefan**** > > ** ** > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Yehavi > Bourvine > *Gesendet:* Dienstag, 8. November 2011 12:06 > *An:* FreeSWITCH Users Help > *Betreff:* [Freeswitch-users] Knowing whether a fax has been sent or not** > ** > > ** ** > > Hello,**** > > **** > > I am building a system to send faxes with txfax() application. How can I > get the result and know whether the fax has been sent or not? ESL, or is > there some callback which I enable?**** > > **** > > 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/20111108/1b57e116/attachment.html From bnaylor at sirran.com Tue Nov 8 14:30:10 2011 From: bnaylor at sirran.com (Ben Naylor) Date: Tue, 8 Nov 2011 11:30:10 -0000 Subject: [Freeswitch-users] zRTP proxy through Freeswitch without de-cryption Message-ID: <01a201cc9e09$c7199aa0$554ccfe0$@sirran.com> Hello Freeswitch community I was wondering if anyone has been able to get Freeswitch to proxy a zRTP/SRTP stream through it, without de-crypting and re-encrypting it? By design, when Freeswitch is in proxy media mode, I believe it only avoids transcoding the media, but still de-crypts the packets. This means that the server would have to be set up as a trusted MITM, which I am trying to avoid. Is there any way you are able to configure the server so that it acts more like a TURN server, and passes the media through while keeping the media encryption intact? Unfortunately Bypass media is not an option at the moment. Any help is greatly appreciated. Cheers Ben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/525a4171/attachment.html From dipkumar.mehta at gmail.com Tue Nov 8 16:18:27 2011 From: dipkumar.mehta at gmail.com (Dip Mehta) Date: Tue, 8 Nov 2011 18:48:27 +0530 Subject: [Freeswitch-users] Issues using loopback Message-ID: Hi, I am trying to integrate Newfies with Freeswitch. I tried using loopback to route it to 9171 to default context. However, it doesnot works. When the customer picks up the call, there is a dead air. Pleae advice. Logs for your refernce. 13526785574 = external number 2011-11-08 08:11:21.192674 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute set(hangup_after_bridge=false) EXECUTE sofia/external/13526785574 set(hangup_after_bridge=false) 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1177 sofia/external/13526785574 SET [hangup_after_bridge]=[false] 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute unset(call_timeout) EXECUTE sofia/external/13526785574 unset(call_timeout) 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1294 UNSET [call_timeout] 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute unset(effective_caller_id_number) EXECUTE sofia/external/13526785574 unset(effective_caller_id_number) 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1294 UNSET [effective_caller_id_number] 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute unset(effective_caller_id_name) EXECUTE sofia/external/13526785574 unset(effective_caller_id_name) 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1294 UNSET [effective_caller_id_name] 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute set(continue_on_fail=true) EXECUTE sofia/external/13526785574 set(continue_on_fail=true) 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1177 sofia/external/13526785574 SET [continue_on_fail]=[true] 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute set(hangup_after_bridge=false) EXECUTE sofia/external/13526785574 set(hangup_after_bridge=false) 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1177 sofia/external/13526785574 SET [hangup_after_bridge]=[false] 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute set(plivo_dial_rang=false) EXECUTE sofia/external/13526785574 set(plivo_dial_rang=false) 2011-11-08 08:11:21.232676 [DEBUG] mod_dptools.c:1177 sofia/external/13526785574 SET [plivo_dial_rang]=[false] 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute unset(bridge_terminate_key) EXECUTE sofia/external/13526785574 unset(bridge_terminate_key) 2011-11-08 08:11:21.232676 [DEBUG] mod_dptools.c:1294 UNSET [bridge_terminate_key] 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute set(bridge_early_media=false) EXECUTE sofia/external/13526785574 set(bridge_early_media=false) 2011-11-08 08:11:21.232676 [DEBUG] mod_dptools.c:1177 sofia/external/13526785574 SET [bridge_early_media]=[false] 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute set(instant_ringback=true) EXECUTE sofia/external/13526785574 set(instant_ringback=true) 2011-11-08 08:11:21.232676 [DEBUG] mod_dptools.c:1177 sofia/external/13526785574 SET [instant_ringback]=[true] 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute set(ringback=$${us-ring}}) EXECUTE sofia/external/13526785574 set(ringback=$%(2000,4000,440,480)}) 2011-11-08 08:11:21.232676 [DEBUG] mod_dptools.c:1177 sofia/external/13526785574 SET [ringback]=[$%(2000,4000,440,480)}] 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute ring_ready() EXECUTE sofia/external/13526785574 ring_ready() 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute bridge([leg_timeout=10]loopback/9171/default/xml:_:) EXECUTE sofia/external/13526785574 bridge([leg_timeout=10]loopback/9171/default/xml:_:) 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:1413 Parsing ultra-global variables 2011-11-08 08:11:21.232676 [DEBUG] switch_event.c:1521 Parsing variable [api_on_ring]=[uuid_setvar 225b5e18-0a0b-11e1-ba82-590058a85379 plivo_dial_rang true] 2011-11-08 08:11:21.232676 [DEBUG] switch_event.c:1521 Parsing variable [api_on_pre_answer]=[uuid_setvar 225b5e18-0a0b-11e1-ba82-590058a85379 plivo_dial_rang true] 2011-11-08 08:11:21.232676 [DEBUG] switch_event.c:1521 Parsing variable [api_on_answer_1]=[sched_api +300 26f2e3f6-0a0b-11e1-8b92-bc305bec2901 uuid_transfer 225b5e18-0a0b-11e1-ba82-590058a85379 -bleg hangup:ALLOTTED_TIMEOUT inline] 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:2299 Parsing session specific variables 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-11-08 08:11:21.232676 [DEBUG] switch_event.c:1521 Parsing variable [leg_timeout]=[10] 2011-11-08 08:11:21.232676 [WARNING] switch_ivr_originate.c:1903 No origination URL specified! 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:3348 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2011-11-08 08:11:21.232676 [NOTICE] switch_channel.c:915 New Channel loopback/9171/default/xml-a [26f5283c-0a0b-11e1-ba88-590058a85379] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:143 loopback/9171/default/xml-a setup codec L16/8000/20 2011-11-08 08:11:21.232676 [NOTICE] switch_channel.c:913 Rename Channel loopback/9171/default/xml-a->loopback/9171-a [26f5283c-0a0b-11e1-ba88-590058a85379] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:943 (loopback/9171-a) State Change CS_NEW -> CS_INIT 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-a [BREAK] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-a CHANNEL KILL 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:2554 loopback/9171-a Setting leg timeout to 10 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 (loopback/9171-a) Running State Change CS_INIT 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:401 (loopback/9171-a) State INIT 2011-11-08 08:11:21.232676 [NOTICE] switch_channel.c:915 New Channel loopback/9171-b [26f53af2-0a0b-11e1-ba8a-590058a85379] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:143 loopback/9171-b setup codec L16/8000/20 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:256 (loopback/9171-b) State Change CS_NEW -> CS_INIT 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-b [BREAK] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:302 (loopback/9171-a) State Change CS_INIT -> CS_ROUTING 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-a [BREAK] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-a CHANNEL KILL 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:401 (loopback/9171-a) State INIT going to sleep 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 (loopback/9171-b) Running State Change CS_INIT 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:401 (loopback/9171-b) State INIT 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:302 (loopback/9171-b) State Change CS_INIT -> CS_ROUTING 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-b [BREAK] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:401 (loopback/9171-b) State INIT going to sleep 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 (loopback/9171-a) Running State Change CS_ROUTING 2011-11-08 08:11:21.232676 [DEBUG] switch_channel.c:1844 (loopback/9171-a) Callstate Change DOWN -> RINGING 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 (loopback/9171-b) Running State Change CS_ROUTING 2011-11-08 08:11:21.232676 [DEBUG] switch_channel.c:1844 (loopback/9171-b) Callstate Change DOWN -> RINGING 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:410 (loopback/9171-a) State ROUTING 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:334 loopback/9171-a CHANNEL ROUTING 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:66 (loopback/9171-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-a [BREAK] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-a CHANNEL KILL 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:410 (loopback/9171-a) State ROUTING going to sleep 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 (loopback/9171-a) Running State Change CS_CONSUME_MEDIA 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:410 (loopback/9171-b) State ROUTING 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:429 (loopback/9171-a) State CONSUME_MEDIA 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:334 loopback/9171-b CHANNEL ROUTING 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:533 CHANNEL CONSUME_MEDIA 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:104 loopback/9171-b Standard ROUTING 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:429 (loopback/9171-a) State CONSUME_MEDIA going to sleep 2011-11-08 08:11:21.232676 [INFO] mod_dialplan_xml.c:336 Processing Outbound Call <13526785574>->9171 in context default Dialplan: loopback/9171-b parsing [default->plivo] continue=false Dialplan: loopback/9171-b Regex (PASS) [plivo] destination_number(9171) =~ /^(\d+)$/ break=on-false Dialplan: loopback/9171-b Action socket(127.0.0.1:8084 async full) 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:154 (loopback/9171-b) State Change CS_ROUTING -> CS_EXECUTE 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-b [BREAK] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:410 (loopback/9171-b) State ROUTING going to sleep 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 (loopback/9171-b) Running State Change CS_EXECUTE 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:417 (loopback/9171-b) State EXECUTE 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:373 loopback/9171-b CHANNEL EXECUTE 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:192 loopback/9171-b Standard EXECUTE EXECUTE loopback/9171-b socket(127.0.0.1:8084 async full) 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1009 Send signal loopback/9171-b [BREAK] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1009 Send signal loopback/9171-b [BREAK] 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1009 Send signal loopback/9171-b [BREAK] 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] mod_event_socket.c:2617 (loopback/9171-b) State Change CS_EXECUTE -> CS_RESET 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-b [BREAK] 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] switch_ivr.c:577 loopback/9171-b Command Execute set(plivo_app=true) EXECUTE loopback/9171-b set(plivo_app=true) 2011-11-08 08:11:21.252674 [DEBUG] mod_dptools.c:1177 loopback/9171-b SET [plivo_app]=[true] 2011-11-08 08:11:21.252674 [DEBUG] switch_ivr.c:577 loopback/9171-b Command Execute set(hangup_after_bridge=false) EXECUTE loopback/9171-b set(hangup_after_bridge=false) 2011-11-08 08:11:21.252674 [DEBUG] mod_dptools.c:1177 loopback/9171-b SET [hangup_after_bridge]=[false] 2011-11-08 08:11:21.252674 [DEBUG] switch_ivr.c:577 loopback/9171-b Command Execute hangup() EXECUTE loopback/9171-b hangup() 2011-11-08 08:11:21.252674 [DEBUG] switch_channel.c:2804 (loopback/9171-b) Callstate Change RINGING -> HANGUP 2011-11-08 08:11:21.252674 [NOTICE] mod_dptools.c:1030 Hangup loopback/9171-b [CS_RESET] [NORMAL_CLEARING] 2011-11-08 08:11:21.252674 [DEBUG] switch_channel.c:2820 Send signal loopback/9171-b [KILL] 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-b [BREAK] 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:2262 loopback/9171-b skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:2262 loopback/9171-b skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:417 (loopback/9171-b) State EXECUTE going to sleep 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:362 (loopback/9171-b) Running State Change CS_HANGUP 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:602 (loopback/9171-b) State HANGUP 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:425 loopback/9171-b CHANNEL HANGUP 2011-11-08 08:11:21.252674 [DEBUG] switch_channel.c:2804 (loopback/9171-a) Callstate Change RINGING -> HANGUP 2011-11-08 08:11:21.252674 [NOTICE] mod_loopback.c:436 Hangup loopback/9171-a [CS_CONSUME_MEDIA] [NORMAL_CLEARING] 2011-11-08 08:11:21.252674 [DEBUG] switch_channel.c:2820 Send signal loopback/9171-a [KILL] 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-a CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-a [BREAK] 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-a CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:47 loopback/9171-b Standard HANGUP, cause: NORMAL_CLEARING 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:602 (loopback/9171-b) State HANGUP going to sleep 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:362 (loopback/9171-a) Running State Change CS_HANGUP 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:393 (loopback/9171-b) State Change CS_HANGUP -> CS_REPORTING 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-b [BREAK] 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:362 (loopback/9171-b) Running State Change CS_REPORTING 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:662 (loopback/9171-b) State REPORTING 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:602 (loopback/9171-a) State HANGUP 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:425 loopback/9171-a CHANNEL HANGUP 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:47 loopback/9171-a Standard HANGUP, cause: NORMAL_CLEARING 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:602 (loopback/9171-a) State HANGUP going to sleep 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:393 (loopback/9171-a) State Change CS_HANGUP -> CS_REPORTING 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-a [BREAK] 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-a CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:362 (loopback/9171-a) Running State Change CS_REPORTING 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:662 (loopback/9171-a) State REPORTING 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:366 Got error [0] posting to web server [NEWFIES_API_STORE_CDR] 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:373 Retry will be with url [NEWFIES_API_STORE_CDR] 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:384 Unable to post to web server, writing to file 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:79 loopback/9171-b Standard REPORTING, cause: NORMAL_CLEARING 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:662 (loopback/9171-b) State REPORTING going to sleep 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:387 (loopback/9171-b) State Change CS_REPORTING -> CS_DESTROY 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-b [BREAK] 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1377 Session 148 (loopback/9171-b) Locked, Waiting on external entities 2011-11-08 08:11:21.252674 [NOTICE] switch_core_session.c:1395 Session 148 (loopback/9171-b) Ended 2011-11-08 08:11:21.252674 [NOTICE] switch_core_session.c:1397 Close Channel loopback/9171-b [CS_DESTROY] 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:491 (loopback/9171-b) Callstate Change HANGUP -> DOWN 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:494 (loopback/9171-b) Running State Change CS_DESTROY 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:504 (loopback/9171-b) State DESTROY 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:86 loopback/9171-b Standard DESTROY 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:504 (loopback/9171-b) State DESTROY going to sleep 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:366 Got error [0] posting to web server [NEWFIES_API_STORE_CDR] 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:373 Retry will be with url [NEWFIES_API_STORE_CDR] 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:384 Unable to post to web server, writing to file 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:79 loopback/9171-a Standard REPORTING, cause: NORMAL_CLEARING 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:662 (loopback/9171-a) State REPORTING going to sleep 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:387 (loopback/9171-a) State Change CS_REPORTING -> CS_DESTROY 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal loopback/9171-a [BREAK] 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-a CHANNEL KILL 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1377 Session 147 (loopback/9171-a) Locked, Waiting on external entities 2011-11-08 08:11:21.252674 [DEBUG] switch_ivr_originate.c:3348 Originate Resulted in Error Cause: 16 [NORMAL_CLEARING] 2011-11-08 08:11:21.252674 [NOTICE] switch_core_session.c:1395 Session 147 (loopback/9171-a) Ended 2011-11-08 08:11:21.252674 [NOTICE] switch_core_session.c:1397 Close Channel loopback/9171-a [CS_DESTROY] 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:491 (loopback/9171-a) Callstate Change HANGUP -> DOWN 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:494 (loopback/9171-a) Running State Change CS_DESTROY 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:504 (loopback/9171-a) State DESTROY 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:86 loopback/9171-a Standard DESTROY 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:504 (loopback/9171-a) State DESTROY going to sleep 2011-11-08 08:11:21.272673 [DEBUG] switch_rtp.c:3181 Correct ip/port confirmed. 2011-11-08 08:11:21.312678 [INFO] mod_dptools.c:2811 Originate Failed. Cause: NORMAL_CLEARING 2011-11-08 08:11:21.312678 [DEBUG] switch_core_session.c:1009 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.312678 [DEBUG] mod_event_socket.c:2617 (sofia/external/13526785574) State Change CS_EXECUTE -> CS_RESET 2011-11-08 08:11:21.312678 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.332673 [DEBUG] switch_ivr.c:577 sofia/external/13526785574 Command Execute hangup() EXECUTE sofia/external/13526785574 hangup() 2011-11-08 08:11:21.332673 [DEBUG] switch_channel.c:2804 (sofia/external/13526785574) Callstate Change ACTIVE -> HANGUP 2011-11-08 08:11:21.332673 [NOTICE] mod_dptools.c:1030 Hangup sofia/external/13526785574 [CS_RESET] [NORMAL_CLEARING] 2011-11-08 08:11:21.332673 [DEBUG] switch_channel.c:2820 Send signal sofia/external/13526785574 [KILL] 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:2262 sofia/external/13526785574 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:2262 sofia/external/13526785574 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:417 (sofia/external/13526785574) State EXECUTE going to sleep 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:362 (sofia/external/13526785574) Running State Change CS_HANGUP 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:602 (sofia/external/13526785574) State HANGUP 2011-11-08 08:11:21.332673 [DEBUG] mod_sofia.c:465 Channel sofia/external/13526785574 hanging up, cause: NORMAL_CLEARING 2011-11-08 08:11:21.332673 [DEBUG] mod_sofia.c:509 Sending BYE to sofia/external/13526785574 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:47 sofia/external/13526785574 Standard HANGUP, cause: NORMAL_CLEARING 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:602 (sofia/external/13526785574) State HANGUP going to sleep 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:393 (sofia/external/13526785574) State Change CS_HANGUP -> CS_REPORTING 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:362 (sofia/external/13526785574) Running State Change CS_REPORTING 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:662 (sofia/external/13526785574) State REPORTING 2011-11-08 08:11:21.332673 [ERR] mod_xml_cdr.c:366 Got error [0] posting to web server [NEWFIES_API_STORE_CDR] 2011-11-08 08:11:21.332673 [ERR] mod_xml_cdr.c:373 Retry will be with url [NEWFIES_API_STORE_CDR] 2011-11-08 08:11:21.332673 [ERR] mod_xml_cdr.c:384 Unable to post to web server, writing to file 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:79 sofia/external/13526785574 Standard REPORTING, cause: NORMAL_CLEARING 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:662 (sofia/external/13526785574) State REPORTING going to sleep 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:387 (sofia/external/13526785574) State Change CS_REPORTING -> CS_DESTROY 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/13526785574 [BREAK] 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:1377 Session 146 (sofia/external/13526785574) Locked, Waiting on external entities 2011-11-08 08:11:21.332673 [NOTICE] switch_core_session.c:1395 Session 146 (sofia/external/13526785574) Ended 2011-11-08 08:11:21.332673 [NOTICE] switch_core_session.c:1397 Close Channel sofia/external/13526785574 [CS_DESTROY] 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:491 (sofia/external/13526785574) Callstate Change HANGUP -> DOWN 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:494 (sofia/external/13526785574) Running State Change CS_DESTROY 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:504 (sofia/external/13526785574) State DESTROY 2011-11-08 08:11:21.332673 [DEBUG] mod_sofia.c:370 sofia/external/13526785574 SOFIA DESTROY 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:86 sofia/external/13526785574 Standard DESTROY 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:504 (sofia/external/13526785574) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/952a0481/attachment-0001.html From miha at softnet.si Tue Nov 8 17:08:33 2011 From: miha at softnet.si (Miha Zoubek) Date: Tue, 08 Nov 2011 15:08:33 +0100 Subject: [Freeswitch-users] One information about freeswitch:) Message-ID: <4EB937E1.1030203@softnet.si> Hi, I am just looking over Internet about freeswitch. Ok, I have read that is capable of 1000 simultaneous calls. What about references. Is it used by a lot of VOIP ISP companies? I did not find any reference about who is using it. Regards, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/74bd98ab/attachment.html From 12ukwn at gmail.com Tue Nov 8 17:29:26 2011 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Tue, 8 Nov 2011 15:29:26 +0100 Subject: [Freeswitch-users] One information about freeswitch:) In-Reply-To: <4EB937E1.1030203@softnet.si> References: <4EB937E1.1030203@softnet.si> Message-ID: <20111108152926.327f7dde@anubis.defcon1> On Tue, 08 Nov 2011 15:08:33 +0100 Miha Zoubek wrote: > I am just looking over Internet about freeswitch. Ok, I have read that > is capable of 1000 simultaneous calls. > What about references. Is it used by a lot of VOIP ISP companies? I did > not find any reference about who is using it. It depend on what you call 'calls' (switching calls only (huge), calls to voicemail (less huge, of course), transcoded calls, etc). About references, grep the archive (http://lists.freeswitch.org/pipermail/freeswitch-users/) and you'll see that many professionals are subscribed. I guess FS success comes from different things (among other): * It can do the whole shebang, * Trunk is always usable, * The ML is very active & reactive, * The dev team is also very reactive, * It can be programmed w/ various languages. My 2 ? PS: I'm not tied in any way to FS. -- Love is the only game that is not called on account of darkness. -- M. Hirschfield From andy at fabulous4.co.uk Tue Nov 8 17:42:22 2011 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 8 Nov 2011 14:42:22 -0000 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' Message-ID: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> Hi, I would like to use freeswitch to host a one to many broadcast style conference call. The challenge I have is that there won't always be any listeners so the conference application isn't quite working for me. Essentially, the speaker dials in with something important to say, by default the system will simply record what they say as an archived message and then they hang up. This works fine but I would like to extend the functionality so that other people dialling in have the option to connect into this call and listen to the message live. Can you advise me on the best way of achieving this: conference, bridge, eavesdrop or something else? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/1c1b60db/attachment.html From curriegrad2004 at gmail.com Tue Nov 8 18:40:51 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 8 Nov 2011 07:40:51 -0800 Subject: [Freeswitch-users] Phantom Calls In-Reply-To: <754CEFADFC65654BAC484896D31C2C030566AC15EB@exchangesrv2007.ONEFONE.MSDOMAIN> References: <754CEFADFC65654BAC484896D31C2C030566AC0D76@exchangesrv2007.ONEFONE.MSDOMAIN> <754CEFADFC65654BAC484896D31C2C030566AC15EB@exchangesrv2007.ONEFONE.MSDOMAIN> Message-ID: I have calls like this being sent to my FS box. Do you have the raw xml cdr file where these guys are calling from? Because if you're getting those calls, it's probably some guy who's brute forcing any sip boxes out there to see if somebody didn't configure their IP-PBXes right in the first place. I used to keep an audio recording of what was going on between the "nasty extension" and the attacker, but I've seem to have deleted them a while back because they usually are quiet on the attacker's end. On Tue, Nov 8, 2011 at 12:21 AM, Nir Cohen wrote: > I actually did. I realized that all calls which were "stuck" in my > freeswitch had 3 common features. > > 1.?????? They were using a GSM codec. > > 2.?????? They were behind a Juniper SRX > > 3.?????? They were going to a specific provider. > > > > I'm going to try and eliminate GSM calls and use only other codecs and see > if this solves the problem. > > Furthermore, I'm going to investigate this problem along with my SysAdmin to > see the difference between packets using GSM which go through the SRX and > other SIP packets.. to see where they get stuck. > > > > I'll update on further development. > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Vellayappan.N > Sent: Monday, October 31, 2011 8:46 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Phantom Calls > > > > Hi > > I am also having the same issue. In my case all the calls have the same > issue. When I calling from my mobile to SIP phone in freeswtich,? it is > ringing to the registered sip number.? But? it continues to ring the sip > number even when i disconnect call in my mobile number.? Also the registered > SIP is automatically start gets ringing with in regular interval? until the > freeswitch is restarted. > > Please let me know if you have addressed this issue.? Thanks in advance > > . > > > On Wed, Oct 26, 2011 at 8:00 PM, Nir Cohen wrote: > > Hello, > > > > I'm new to freeswitch. Even though I've been using it for quite a while, I > hadn't had the chance to really dig into the config/monitoring/debugging. > > > > I have a freeswitch which runs flawlessly with a large number of calls but a > small percentage of them do not disconnect at all until the freeswitch > service is restarted. > > I was wondering if there's some timeout which I need to define or if there's > a way to see a direct reason for this to happen. > > > > -????????? I know they are stuck because I did "show calls" and saw calls > from different times during the time it was up (see below) > > -????????? I removed the freeswitch from the DNS to make sure it's not > receiving anymore calls. > > -????????? I opened wireshark and saw that the same amount of calls are > referred to as "session in progress" ? that is, my fs sends a "Status: 183 > session progress?" and the app server returns: (ICMP) "destination > unreachable". > > > > Thank you. > > > > > > > > > > > > > > > > > > > > > > call_created,call_created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid > > 2011-10-09 > 15:10:08,1318165808,switch_ivr_signal_bridge,1001,1001,r0018095215036,sofia/external/1001 at 172.16.100.155,edac000a-bc5a-8642-aba8-ed3be16cd762,1001,fring,18095215036,sofia/external/18095215036,487fbf53-4d6c-134e-93bc-31cc3ee1baf8 > > 2011-10-09 > 23:34:57,1318196097,switch_ivr_signal_bridge,1001,1001,r006498287601,sofia/external/1001 at 172.16.100.155,5d4b4b67-def4-2c44-addc-73f3366d6f83,1001,fring,6498287601,sofia/external/6498287601,6087cfe3-e9b6-de44-b38e-bc86d83ff95a > > 2011-10-09 > 23:38:30,1318196310,switch_ivr_signal_bridge,1001,1001,r0020472321728,sofia/external/1001 at 172.16.100.155,9913ff13-33ac-9d4b-b601-d761e49b81b6,1001,fring,20472321728,sofia/external/20472321728,954fff0a-4b7c-8444-8821-306dc6729e42 > > 2011-10-09 > 23:41:09,1318196469,switch_ivr_signal_bridge,1001,1001,r0020108866522,sofia/external/1001 at 172.16.100.155,f6590d4f-bda0-2146-a309-f2e485ff231c,1001,fring,20108866522,sofia/external/20108866522,ba1f72d8-49fc-5145-8376-87e9f4a3b3b4 > > 2011-10-10 > 02:16:36,1318205796,switch_ivr_signal_bridge,1001,1001,r0050687102330,sofia/external/1001 at 172.16.100.155,c48e48c1-590d-f84a-a34a-e2da40574d59,1001,fring,50687102330,sofia/external/50687102330,95c4c053-c925-5c42-9ac8-87cace55694b > > 2011-10-10 > 06:09:18,1318219758,switch_ivr_signal_bridge,1001,1001,r00919744973603,sofia/external/1001 at 172.16.100.155,04127975-c3f9-b949-907e-f915197bf4fc,1001,fring,919744973603,sofia/external/919744973603,f86302d2-e7ab-0940-b927-b673544c7241 > > 2011-10-10 > 06:21:24,1318220484,switch_ivr_signal_bridge,1001,1001,r00917204674056,sofia/external/1001 at 172.16.100.155,c835ad85-9e04-b84d-993e-426261686cc7,1001,fring,917204674056,sofia/external/917204674056,1304eae6-4def-c743-9214-89991e6f3350 > > 2011-10-10 > 06:53:04,1318222384,switch_ivr_signal_bridge,1001,1001,r00919715155054,sofia/external/1001 at 172.16.100.155,c4082f14-7cae-e846-9b05-65a519320619,1001,fring,919715155054,sofia/external/919715155054,16fa4e97-eb15-d24b-907d-f39a7741b493 > > 2011-10-10 > 07:26:37,1318224397,switch_ivr_signal_bridge,1001,1001,r00919215900646,sofia/external/1001 at 172.16.100.155,40a79a56-cf31-824f-bd47-0006f4abb59b,1001,fring,919215900646,sofia/external/919215900646,f2a02bce-d25a-f448-af39-fa60f3509f51 > > 2011-10-10 > 09:38:45,1318232325,switch_ivr_signal_bridge,1001,1001,r00919779210235,sofia/external/1001 at 172.16.100.155,3249aa91-d99f-8947-bcd4-749e199ecfbb,1001,fring,919779210235,sofia/external/919779210235,6b55ec96-6fe2-d047-a349-efd44df00c67 > > 2011-10-10 > 13:22:42,1318245762,switch_ivr_signal_bridge,1001,1001,r00919312238801,sofia/external/1001 at 172.16.100.155,1226ffc9-f338-8b42-be8a-93868dc1ed0a,1001,fring,919312238801,sofia/external/919312238801,f63b997f-2371-2441-b0f3-cfcf0330c87a > > 2011-10-10 > 13:37:31,1318246651,switch_ivr_signal_bridge,1001,1001,r0091978999666,sofia/external/1001 at 172.16.100.155,3bf75841-ef52-8244-9c18-5ff783ad5bd4,1001,fring,91978999666,sofia/external/91978999666,8de43ba6-ff36-bf40-941f-a71d3574a95d > > 2011-10-10 > 14:15:44,1318248944,switch_ivr_signal_bridge,1001,1001,r00919855283870,sofia/external/1001 at 172.16.100.155,a1844ff2-c9f7-eb4b-a9ac-4e31d5c5a591,1001,fring,919855283870,sofia/external/919855283870,5c73749d-3f35-aa46-a13e-9c1b5d78c592 > > 2011-10-10 > 14:56:30,1318251390,switch_ivr_signal_bridge,1001,1001,r00919891882161,sofia/external/1001 at 172.16.100.155,9b087d32-2a87-db4f-ac66-3bb91d7afdeb,1001,fring,919891882161,sofia/external/919891882161,0972c900-805d-f949-b47a-fb1398a5739d > > 2011-10-10 > 16:31:56,1318257116,switch_ivr_signal_bridge,1001,1001,r00919810399899,sofia/external/1001 at 172.16.100.155,c8b382a4-81ce-6740-bb5d-ebc9ab37d388,1001,fring,919810399899,sofia/external/919810399899,84313caf-fe39-5f45-bf4a-d1cc5e575eff > > 2011-10-10 > 16:47:27,1318258047,switch_ivr_signal_bridge,1001,1001,r0096597284481,sofia/external/1001 at 172.16.100.155,b8d93c17-97ee-074e-84c9-103e5259ddb1,1001,fring,96597284481,sofia/external/96597284481,1dee8a27-b3f1-e246-99f6-baca4cea0dd6 > > 2011-10-10 > 20:11:30,1318270290,switch_ivr_signal_bridge,1001,1001,r00967771771821,sofia/external/1001 at 172.16.100.155,16462169-cb2f-6b42-bf01-64e43a8e4311,1001,fring,967771771821,sofia/external/967771771821,0a98e975-6bd1-4948-8a99-2b682935e893 > > 2011-10-11 > 01:37:56,1318289876,switch_ivr_signal_bridge,1001,1001,r00584120208712,sofia/external/1001 at 172.16.100.155,7566612f-e7f5-a74a-848b-fa52db667bc1,1001,fring,584120208712,sofia/external/584120208712,bec24956-f8cc-9e46-a1b1-e5f75f827306 > > 2011-10-11 > 02:23:53,1318292633,switch_ivr_signal_bridge,1001,1001,r00919815288834,sofia/external/1001 at 172.16.100.155,0c7c07a8-ed17-3e43-9d00-5928b998ac48,1001,fring,919815288834,sofia/external/919815288834,11c3f91e-4e84-7d4c-93d9-fbadbc0e0d69 > > 2011-10-11 > 04:30:11,1318300211,switch_ivr_signal_bridge,1001,1001,r00919924392230,sofia/external/1001 at 172.16.100.155,63514e60-511f-5748-b1ed-6ebcc41d5878,1001,fring,919924392230,sofia/external/919924392230,b465f0f0-f0f5-fb4c-bb0f-93b23b59a4cf > > 2011-10-11 > 05:05:39,1318302339,switch_ivr_signal_bridge,1001,1001,r00919703570642,sofia/external/1001 at 172.16.100.155,d6181187-6d71-844a-a4cd-b8add2393c0b,1001,fring,919703570642,sofia/external/919703570642,ed78294e-33f5-7d4d-af4b-40e0efbc9d17 > > 2011-10-11 > 07:24:57,1318310697,switch_ivr_signal_bridge,1001,1001,r00919845045625,sofia/external/1001 at 172.16.100.155,781823e0-1496-2f46-b6bc-a98cee121887,1001,fring,919845045625,sofia/external/919845045625,bd19f919-4b3c-3342-a23f-53088baedc8f > > 2011-10-11 > 07:55:46,1318312546,switch_ivr_signal_bridge,1001,1001,r0084090,sofia/external/1001 at 172.16.100.155,8edd5195-a573-4042-adf7-2c7d093b8178,1001,fring,84090,sofia/external/84090,7e83eb35-d155-384a-ab22-4545e0bb62bb > > 2011-10-11 > 10:11:50,1318320710,switch_ivr_signal_bridge,1001,1001,r009197845827258,sofia/external/1001 at 172.16.100.155,0dbff49d-e8cd-f44d-ab68-f0d5babd61a8,1001,fring,9197845827258,sofia/external/9197845827258,d11b3020-6def-2849-aac0-7454c25d5077 > > 2011-10-11 > 11:28:26,1318325306,switch_ivr_signal_bridge,1001,1001,r00996550675305,sofia/external/1001 at 172.16.100.155,9354bc8e-100d-3e4e-972a-5814fde9ce4c,1001,fring,996550675305,sofia/external/996550675305,cf2662b9-3be1-a440-91f5-7d3088d574bb > > 2011-10-11 > 16:43:55,1318344235,switch_ivr_signal_bridge,1001,1001,r00919895419251,sofia/external/1001 at 172.16.100.155,69dab1ec-9d20-4c4d-a8b8-036cc2be4382,1001,fring,919895419251,sofia/external/919895419251,db22ba2d-dff2-1544-a67e-578886921592 > > 2011-10-11 > 17:48:39,1318348119,switch_ivr_signal_bridge,1001,1001,r00919773212368,sofia/external/1001 at 172.16.100.155,7a35e17e-9604-4743-b131-d339e08245d0,1001,fring,919773212368,sofia/external/919773212368,0cce04b4-3776-4c4f-adf8-e42a1555d33a > > 2011-10-12 > 01:57:34,1318377454,switch_ivr_signal_bridge,1001,1001,r0023794311238,sofia/external/1001 at 172.16.100.155,4a09bce4-5342-3943-9b36-a9bd7c51d8bc,1001,fring,23794311238,sofia/external/23794311238,ede10bc2-45b2-1240-a20c-9b3e65868823 > > 2011-10-12 > 05:39:54,1318390794,switch_ivr_signal_bridge,1001,1001,r00918121275024,sofia/external/1001 at 172.16.100.155,4c991cad-f71d-e644-9946-8fff485c9a40,1001,fring,918121275024,sofia/external/918121275024,4b3e9c9f-bb26-814e-89e5-0685c05692df > > 2011-10-12 > 06:39:05,1318394345,switch_ivr_signal_bridge,1001,1001,r00919947389305,sofia/external/1001 at 172.16.100.155,d2a0d36e-182e-bc47-8fe2-83032467eff3,1001,fring,919947389305,sofia/external/919947389305,e5d52ab2-2c36-1843-bea8-0cfdce391e15 > > 2011-10-12 > 06:59:34,1318395574,switch_ivr_signal_bridge,1001,1001,r00865395760139,sofia/external/1001 at 172.16.100.155,03a603a0-6418-7a46-a1d4-5d4d2357bf23,1001,fring,865395760139,sofia/external/865395760139,ef325309-716c-1246-8f01-513c7e9a93bb > > 2011-10-12 > 08:49:22,1318402162,switch_ivr_signal_bridge,1001,1001,r00919944731940,sofia/external/1001 at 172.16.100.155,024d66fc-6582-b04b-a962-56c4db8cb013,1001,fring,919944731940,sofia/external/919944731940,1de9cb6b-4eed-3d4d-89bc-6b533b07849a > > 2011-10-12 > 09:37:03,1318405023,switch_ivr_signal_bridge,1001,1001,r00919916780545,sofia/external/1001 at 172.16.100.155,379b8c6e-a998-2649-a577-5f8f1b6b3c4e,1001,fring,919916780545,sofia/external/919916780545,097c1a46-1e65-6749-aa14-592b1cae6fd5 > > 2011-10-12 > 10:17:08,1318407428,switch_ivr_signal_bridge,1001,1001,r00914562224442,sofia/external/1001 at 172.16.100.155,f91ec6f2-e8a1-b549-86b5-cbb1d17039b3,1001,fring,914562224442,sofia/external/914562224442,3c2ced4c-7b98-4948-8520-8bc357e4a437 > > 2011-10-12 > 13:54:07,1318420447,switch_ivr_signal_bridge,1001,1001,r00919746980086,sofia/external/1001 at 172.16.100.155,6f1a25be-891e-ee4f-8c70-4ae79694a651,1001,fring,919746980086,sofia/external/919746980086,d0fdad10-f3a2-5347-ab1f-95d4383089b7 > > 2011-10-12 > 21:04:21,1318446261,switch_ivr_signal_bridge,1001,1001,r0096597280025,sofia/external/1001 at 172.16.100.155,3349eb67-9f4b-f64f-98af-854dd9a07c23,1001,fring,96597280025,sofia/external/96597280025,14d7da2f-d2e6-084c-b5d3-375377400832 > > 2011-10-13 > 08:38:46,1318487926,switch_ivr_signal_bridge,1001,1001,r00639214286751,sofia/external/1001 at 172.16.100.155,9b909f83-2d31-a145-82bb-1c0dd1f46695,1001,fring,639214286751,sofia/external/639214286751,6bd8bfbf-4391-1e48-ae3d-16c7f81c74a9 > > 2011-10-13 > 09:28:42,1318490922,switch_ivr_signal_bridge,1001,1001,r00919855874737,sofia/external/1001 at 172.16.100.155,ff74debc-509c-2040-bf50-42313aa6629d,1001,fring,919855874737,sofia/external/919855874737,579ecfbf-d00c-7947-9c2f-6029dbf9b507 > > 2011-10-13 > 10:44:23,1318495463,switch_ivr_signal_bridge,1001,1001,r00919966842195,sofia/external/1001 at 172.16.100.155,16e5bb3f-5ff1-0743-b787-78fbfb032583,1001,fring,919966842195,sofia/external/919966842195,77d980c9-4ccb-1c40-a901-3e6fd6960a2d > > 2011-10-13 > 11:54:10,1318499650,switch_ivr_signal_bridge,1001,1001,r00919751864788,sofia/external/1001 at 172.16.100.155,84725b33-97ab-d64f-8a36-b7f10a2b8723,1001,fring,919751864788,sofia/external/919751864788,f7e86569-01a2-5540-9c9b-3e300f1a0797 > > 2011-10-13 > 14:26:04,1318508764,switch_ivr_signal_bridge,1001,1001,r00919922440249,sofia/external/1001 at 172.16.100.155,ad0c00ab-17c2-4949-880b-9ff164941f1b,1001,fring,919922440249,sofia/external/919922440249,1d139c8d-82eb-9248-8e81-e9d6315ba50d > > 2011-10-13 > 16:10:48,1318515048,switch_ivr_signal_bridge,1001,1001,r0020197596666,sofia/external/1001 at 172.16.100.155,ad64dcd2-232c-d543-8d01-2a513c3dc315,1001,fring,20197596666,sofia/external/20197596666,7f9b2f6c-41db-3840-ab78-e0052c56a885 > > 2011-10-13 > 20:09:02,1318529342,switch_ivr_signal_bridge,1001,1001,r00551158958193,sofia/external/1001 at 172.16.100.155,0730806a-137f-924e-ace6-9852c72bb185,1001,fring,551158958193,sofia/external/551158958193,de18fc0d-d46f-014c-92f1-ff5ac3d69e47 > > 2011-10-13 > 21:11:25,1318533085,switch_ivr_signal_bridge,1001,1001,r0020123946125,sofia/external/1001 at 172.16.100.155,74c80972-2dd4-c443-b831-d76003497612,1001,fring,20123946125,sofia/external/20123946125,be927ca2-99cc-4840-aa86-8c9a85e32f80 > > 2011-10-13 > 22:43:24,1318538604,switch_ivr_signal_bridge,1001,1001,r00919769404307,sofia/external/1001 at 172.16.100.155,16b1cdf4-c7de-2642-a3d1-a901c7168829,1001,fring,919769404307,sofia/external/919769404307,74860ee3-d592-8441-a738-3b82103033a1 > > 2011-10-14 > 03:50:54,1318557054,switch_ivr_signal_bridge,1001,1001,r00919988274760,sofia/external/1001 at 172.16.100.155,378a4ecc-7646-6147-9086-381a2ffc5570,1001,fring,919988274760,sofia/external/919988274760,49650a00-03dc-054a-ae56-2debe46897a4 > > 2011-10-14 > 04:13:23,1318558403,switch_ivr_signal_bridge,1001,1001,r00919944010360,sofia/external/1001 at 172.16.100.155,5aec4e1e-ed85-ef41-b9fe-7f7e9666b0da,1001,fring,919944010360,sofia/external/919944010360,39aa0999-6817-1148-98a2-dc8b1cf768c8 > > 2011-10-14 > 05:28:12,1318562892,switch_ivr_signal_bridge,1001,1001,r00919944191129,sofia/external/1001 at 172.16.100.155,f4e6f3dc-543a-0a4e-b81d-58c8e5434b0a,1001,fring,919944191129,sofia/external/919944191129,b702994a-2237-a442-89c3-1026966a1feb > > 2011-10-14 > 06:14:54,1318565694,switch_ivr_signal_bridge,1001,1001,r0060148144168,sofia/external/1001 at 172.16.100.155,95f21383-70db-2848-a1a4-bf1e279970cb,1001,fring,60148144168,sofia/external/60148144168,f9c3747a-4c39-aa4d-a655-7cea1222ada9 > > 2011-10-14 > 06:40:42,1318567242,switch_ivr_signal_bridge,1001,1001,r008615952583367,sofia/external/1001 at 172.16.100.155,d4976a2b-d218-a14a-ab73-ed5252d23223,1001,fring,8615952583367,sofia/external/8615952583367,d514508a-e4c8-f14f-82fc-43b71560f565 > > 2011-10-14 > 13:30:22,1318591822,switch_ivr_signal_bridge,1001,1001,r00919902730542,sofia/external/1001 at 172.16.100.155,535a6926-431e-154b-bed1-bea1ea8240a0,1001,fring,919902730542,sofia/external/919902730542,45a49ed1-b426-5f4d-8da6-5d7ae5934dc9 > > 2011-10-14 > 13:52:34,1318593154,switch_ivr_signal_bridge,1001,1001,r008613986690504,sofia/external/1001 at 172.16.100.155,39dd44b8-1854-be40-8cd4-81c34ebbcc11,1001,fring,8613986690504,sofia/external/8613986690504,7d641511-0cd1-c54c-8058-568f7f0f67f0 > > 2011-10-14 > 14:44:34,1318596274,switch_ivr_signal_bridge,1001,1001,r0020101,sofia/external/1001 at 172.16.100.155,08408567-1407-ea4a-be73-ce0219738962,1001,fring,20101,sofia/external/20101,aa1cbd89-f2f1-114a-a0a4-2d18fb647a1d > > 2011-10-14 > 19:30:55,1318613455,switch_ivr_signal_bridge,1001,1001,r00919953239015,sofia/external/1001 at 172.16.100.155,931bd3ec-a4a9-1b47-b6a3-7226cf83dc3b,1001,fring,919953239015,sofia/external/919953239015,37b98044-fbe2-e940-a368-fe9556fb810f > > 2011-10-15 > 01:42:58,1318635778,switch_ivr_signal_bridge,1001,1001,r00919032587213,sofia/external/1001 at 172.16.100.155,11537992-801d-eb42-ba4e-763fbff0f381,1001,fring,919032587213,sofia/external/919032587213,4d2e98a0-d4b4-d046-aa6c-11c7d7137c63 > > 2011-10-15 > 05:32:32,1318649552,switch_ivr_signal_bridge,1001,1001,r00919885688639,sofia/external/1001 at 172.16.100.155,b2b70ad1-0754-8c49-82eb-c0ae5d051618,1001,fring,919885688639,sofia/external/919885688639,a35b33e7-08c0-f34f-b664-2c1b68be3e95 > > 2011-10-15 > 06:40:56,1318653656,switch_ivr_signal_bridge,1001,1001,r00919886690120,sofia/external/1001 at 172.16.100.155,13a08888-c378-d949-8625-95e2239bac45,1001,fring,919886690120,sofia/external/919886690120,ff62df61-8354-0641-a1ce-cba74fe53448 > > 2011-10-15 > 09:51:20,1318665080,switch_ivr_signal_bridge,1001,1001,r00265888891340,sofia/external/1001 at 172.16.100.155,41c4263a-18f2-d147-8b7a-3a14739eeb4f,1001,fring,265888891340,sofia/external/265888891340,1c19dbaa-d35c-3e4d-a913-355d2f9b6c0d > > 2011-10-15 > 10:06:41,1318666001,switch_ivr_signal_bridge,1001,1001,r00919820158704,sofia/external/1001 at 172.16.100.155,c1363ab8-2905-2549-876c-dd3cbc05563f,1001,fring,919820158704,sofia/external/919820158704,660b2675-974e-a249-8a24-349020a7449a > > 2011-10-15 > 11:19:21,1318670361,switch_ivr_signal_bridge,1001,1001,r0020197105643,sofia/external/1001 at 172.16.100.155,cc2f1ede-7d0c-8845-b455-4eb97ed2b12f,1001,fring,20197105643,sofia/external/20197105643,81cb54ff-b63d-6c4e-afa9-bfea68a553a9 > > 2011-10-15 > 12:14:13,1318673653,switch_ivr_signal_bridge,1001,1001,r00918870183328,sofia/external/1001 at 172.16.100.155,81ba52e8-d157-4441-bfe1-c314e4f1b4c7,1001,fring,918870183328,sofia/external/918870183328,72306599-6017-c54a-a291-9b3eb1650d83 > > 2011-10-15 > 14:15:42,1318680942,switch_ivr_signal_bridge,1001,1001,r00919539499709,sofia/external/1001 at 172.16.100.155,b24a486b-b798-c048-b0c5-9c624b0ebade,1001,fring,919539499709,sofia/external/919539499709,456681fa-c2bf-004f-875a-4d4b18ea1e99 > > 2011-10-15 > 23:23:12,1318713792,switch_ivr_signal_bridge,1001,1001,r0020140088523,sofia/external/1001 at 172.16.100.155,c2f98b57-0c7b-1b41-a77c-05a34ab99b79,1001,fring,20140088523,sofia/external/20140088523,f681a50f-d263-a942-a48f-ab7eb2adff38 > > 2011-10-16 > 02:16:44,1318724204,switch_ivr_signal_bridge,1001,1001,r00919886859264,sofia/external/1001 at 172.16.100.155,764923b2-29b4-fd4d-9de8-cd89f2654a9f,1001,fring,919886859264,sofia/external/919886859264,ecbe5234-bb44-1a40-a93c-4fd36307f3e9 > > 2011-10-16 > 06:50:47,1318740647,switch_ivr_signal_bridge,1001,1001,r00919915919879,sofia/external/1001 at 172.16.100.155,27d55f39-86e3-f146-ad9b-2f77938ad868,1001,fring,919915919879,sofia/external/919915919879,2df4fa27-2896-6f4d-81ce-97532b2dee0c > > 2011-10-16 > 10:02:03,1318752123,switch_ivr_signal_bridge,1001,1001,r00965555751124,sofia/external/1001 at 172.16.100.155,7d69745c-89e1-ad49-8f57-ca146ecc769a,1001,fring,965555751124,sofia/external/965555751124,973c1c3b-0f50-684a-873b-b61bdd21a245 > > 2011-10-16 > 11:37:30,1318757850,switch_ivr_signal_bridge,1001,1001,r00201001301440,sofia/external/1001 at 172.16.100.155,22fb4d57-3207-4d4c-8be1-145655dad6a5,1001,fring,201001301440,sofia/external/201001301440,397ee409-504c-244c-b11b-b693070ff7dd > > 2011-10-16 > 14:55:05,1318769705,switch_ivr_signal_bridge,1001,1001,r00919742055333,sofia/external/1001 at 172.16.100.155,1817b98a-e036-1746-bd7d-437152ed6b0a,1001,fring,919742055333,sofia/external/919742055333,ab2b04ce-a454-a041-a40e-7ee5ddbb29a4 > > 2011-10-16 > 16:20:52,1318774852,switch_ivr_signal_bridge,1001,1001,r00998662245737,sofia/external/1001 at 172.16.100.155,986c8dc4-831b-0041-9b3c-1bee42818ab5,1001,fring,998662245737,sofia/external/998662245737,a1eaeb3c-6b1a-ea4c-a3b1-f3144572bf72 > > 2011-10-16 > 16:51:24,1318776684,switch_ivr_signal_bridge,1001,1001,r008801712054706,sofia/external/1001 at 172.16.100.155,84a34d92-8be5-1644-a9cd-487023eee234,1001,fring,8801712054706,sofia/external/8801712054706,1930550c-d3e3-f645-bda9-e4a0d39a6397 > > 2011-10-16 > 18:14:22,1318781662,switch_ivr_signal_bridge,1001,1001,r00919390177288,sofia/external/1001 at 172.16.100.155,2878943a-56ee-be47-a5fc-632a92c3734a,1001,fring,919390177288,sofia/external/919390177288,4a8de922-86ab-ba41-8e9c-05a79087a79a > > 2011-10-16 > 20:06:31,1318788391,switch_ivr_signal_bridge,1001,1001,r00201224955898,sofia/external/1001 at 172.16.100.155,fc68f803-781d-c347-a6dc-f50a8651f61f,1001,fring,201224955898,sofia/external/201224955898,1e873400-c1ca-da42-be32-837931d2c845 > > 2011-10-16 > 22:52:28,1318798348,switch_ivr_signal_bridge,1001,1001,r00914567242736,sofia/external/1001 at 172.16.100.155,d8b4385a-bcf9-ad4a-ae00-0539fe7c56f8,1001,fring,914567242736,sofia/external/914567242736,5eb0ca19-155a-3047-a480-8f951f26993f > > 2011-10-16 > 23:06:09,1318799169,switch_ivr_signal_bridge,1001,1001,r00966552321650,sofia/external/1001 at 172.16.100.155,f9998988-f7f1-9543-afd5-1cd76e035594,1001,fring,966552321650,sofia/external/966552321650,c01c89f6-212b-5a48-90ce-c79587ecdbac > > 2011-10-17 > 00:38:08,1318804688,switch_ivr_signal_bridge,1001,1001,r00911821277451,sofia/external/1001 at 172.16.100.155,54365caa-8f25-8545-9010-db155e4bdde8,1001,fring,911821277451,sofia/external/911821277451,8df3edfb-3b35-a047-8c8d-51d57eeb56b3 > > 2011-10-17 > 06:02:00,1318824120,switch_ivr_signal_bridge,1001,1001,r00919746008642,sofia/external/1001 at 172.16.100.155,ba807588-4536-d94f-81a7-e1d0e8434e05,1001,fring,919746008642,sofia/external/919746008642,6252a3b1-a0a6-ff42-8188-69e771094104 > > 2011-10-17 > 06:30:17,1318825817,switch_ivr_signal_bridge,1001,1001,r00919824645363,sofia/external/1001 at 172.16.100.155,2e099d56-5e90-2f48-816f-d66e410fc8b9,1001,fring,919824645363,sofia/external/919824645363,7d18d6e9-dfb9-c949-b00b-5a9e62ecc21c > > 2011-10-17 > 07:42:55,1318830175,switch_ivr_signal_bridge,1001,1001,r00919824645363,sofia/external/1001 at 172.16.100.155,54960c11-5352-e34b-ae35-672923413dae,1001,fring,919824645363,sofia/external/919824645363,bbc5cdc4-7a02-2945-b685-28baae5fe280 > > 2011-10-17 > 10:11:26,1318839086,switch_ivr_signal_bridge,1001,1001,r00919711534925,sofia/external/1001 at 172.16.100.155,583ffded-6229-6345-bd46-4ef09652a8aa,1001,fring,919711534925,sofia/external/919711534925,9d101a1a-0e4e-7144-bff1-a132378a3d9f > > 2011-10-17 > 17:50:55,1318866655,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/1001 at 172.16.100.155,b6cd78b7-7da4-b44e-a4f4-60e404fafca3,1001,fring,919988175489,sofia/external/919988175489,602bd41d-3f60-d144-b2bb-36d4da232ddc > > 2011-10-17 > 22:52:13,1318884733,switch_ivr_signal_bridge,1001,1001,r00919999233886,sofia/external/1001 at 172.16.100.155,eefd784e-9c04-084f-888b-b0a033a5ee5b,1001,fring,919999233886,sofia/external/919999233886,389a824e-e4c5-b74e-ac8a-9315d9821117 > > 2011-10-17 > 23:24:02,1318886642,switch_ivr_signal_bridge,1001,1001,r00919313467728,sofia/external/1001 at 172.16.100.155,fcf32cd6-9148-cc40-bc90-534145199ba9,1001,fring,919313467728,sofia/external/919313467728,7d13a986-8454-be4d-a056-c47276ba9f5b > > 2011-10-18 > 00:41:19,1318891279,switch_ivr_signal_bridge,1001,1001,r00919747072485,sofia/external/1001 at 172.16.100.155,25f36536-4606-1942-825f-5bfaf28a66ca,1001,fring,919747072485,sofia/external/919747072485,ee6298c7-a6d4-3547-9578-7c95f96ede0f > > 2011-10-18 > 01:05:08,1318892708,switch_ivr_signal_bridge,1001,1001,r00919313467728,sofia/external/1001 at 172.16.100.155,8c89657f-9ce7-404a-9d5c-a3559f66b141,1001,fring,919313467728,sofia/external/919313467728,b6a3eb92-6419-4643-930f-e7bebeabb33a > > 2011-10-18 > 03:25:04,1318901104,switch_ivr_signal_bridge,1001,1001,r006596112485,sofia/external/1001 at 172.16.100.155,b5189c01-0df2-4e47-af11-8a3e96e247a8,1001,fring,6596112485,sofia/external/6596112485,a834eb93-3744-be49-8165-3b39966c9502 > > 2011-10-18 > 06:25:34,1318911934,switch_ivr_signal_bridge,1001,1001,r00919876855863,sofia/external/1001 at 172.16.100.155,d6589f30-ba06-5c45-9d20-1e99664a10d8,1001,fring,919876855863,sofia/external/919876855863,4999eac4-e560-fe40-b795-fc5d5cf8ac44 > > 2011-10-18 > 09:41:35,1318923695,switch_ivr_signal_bridge,1001,1001,r00919922965741,sofia/external/1001 at 172.16.100.155,57335341-bda9-3943-8f0d-59c7a7903c15,1001,fring,919922965741,sofia/external/919922965741,d201c220-2c1f-bb4e-b725-bada9895035b > > 2011-10-18 > 12:52:37,1318935157,switch_ivr_signal_bridge,1001,1001,r00919967359493,sofia/external/1001 at 172.16.100.155,66fadc05-a756-a848-bb84-5d39c9998fc5,1001,fring,919967359493,sofia/external/919967359493,71bcd983-d4a4-de4c-b615-f0dbf1095739 > > 2011-10-18 > 16:41:02,1318948862,switch_ivr_signal_bridge,1001,1001,r00919831094384,sofia/external/1001 at 172.16.100.155,2a96ebc1-a6be-8e49-b508-8b3993b62f9a,1001,fring,919831094384,sofia/external/919831094384,86e74690-fca2-6247-9d6a-c7732445a1be > > 2011-10-18 > 21:01:18,1318964478,switch_ivr_signal_bridge,1001,1001,r00919729947506,sofia/external/1001 at 172.16.100.155,96995af7-e82c-3e40-addb-1b62c1f05747,1001,fring,919729947506,sofia/external/919729947506,6a82a0ec-e4a2-be4f-85ee-87b34b4ea3e2 > > 2011-10-19 > 00:20:24,1318976424,switch_ivr_signal_bridge,1001,1001,r00919605053363,sofia/external/1001 at 172.16.100.155,8f9d6670-2aff-c242-b01e-3d9f0e493d12,1001,fring,919605053363,sofia/external/919605053363,7fd8cee5-cd6b-7b44-848f-8d756f26fd4a > > 2011-10-19 > 04:30:30,1318991430,switch_ivr_signal_bridge,1001,1001,r00919814950511,sofia/external/1001 at 172.16.100.155,1a6895e1-57a5-564c-bf08-5780df1d79c7,1001,fring,919814950511,sofia/external/919814950511,595fe521-1673-ea44-8bc3-c9d69c2f2de6 > > 2011-10-19 > 04:40:27,1318992027,switch_ivr_signal_bridge,1001,1001,r00919814950511,sofia/external/1001 at 172.16.100.155,905be5af-844a-d44f-bbfd-a160ff1f1759,1001,fring,919814950511,sofia/external/919814950511,8a41744e-b19e-954a-a68f-7d16e29976e5 > > 2011-10-19 > 06:23:43,1318998223,switch_ivr_signal_bridge,1001,1001,r00919990249401,sofia/external/1001 at 172.16.100.155,d009a643-b1b3-3f46-b194-c971e2a17b64,1001,fring,919990249401,sofia/external/919990249401,f26ae13f-de35-2f40-92cb-232449391e3e > > 2011-10-19 > 09:38:43,1319009923,switch_ivr_signal_bridge,1001,1001,r00919761333537,sofia/external/1001 at 172.16.100.155,353ede21-8d68-b74e-8711-0ddce2a82928,1001,fring,919761333537,sofia/external/919761333537,d0567650-c2f4-9542-9a20-d8bc1718cc80 > > 2011-10-19 > 10:59:56,1319014796,switch_ivr_signal_bridge,1001,1001,r00918144890121,sofia/external/1001 at 172.16.100.155,8593dc8b-8d4f-bc40-8e3a-daf85d7c7702,1001,fring,918144890121,sofia/external/918144890121,ec3fda7c-cc8e-e041-b4a0-171928ade343 > > 2011-10-19 > 13:00:41,1319022041,switch_ivr_signal_bridge,1001,1001,r00201060198003,sofia/external/1001 at 172.16.100.155,bba15b1f-785a-674f-a023-bb8f64f57c53,1001,fring,201060198003,sofia/external/201060198003,2c90f16e-a4ec-6d42-9bd1-4f06bf7c2da9 > > 2011-10-19 > 13:45:11,1319024711,switch_ivr_signal_bridge,1001,1001,r00914312341722,sofia/external/1001 at 172.16.100.155,845903b7-2770-6847-a116-3d0a8cb7b096,1001,fring,914312341722,sofia/external/914312341722,aa20b33f-eb4d-714a-b236-59afbfe158e7 > > 2011-10-19 > 19:11:01,1319044261,switch_ivr_signal_bridge,1001,1001,r009845967072,sofia/external/1001 at 172.16.100.155,349ef121-64b9-dc44-a22d-3d12acbb6ece,1001,fring,9845967072,sofia/external/9845967072,d89904a7-daf7-ea45-bae3-57c914e88b48 > > 2011-10-20 > 02:36:13,1319070973,switch_ivr_signal_bridge,1001,1001,r0060166298037,sofia/external/1001 at 172.16.100.155,f6ff096b-612e-0544-a733-9a566b08fc6d,1001,fring,60166298037,sofia/external/60166298037,1c35f10e-3695-2845-9f4c-dd5a517dd6f7 > > 2011-10-20 > 03:15:43,1319073343,switch_ivr_signal_bridge,1001,1001,r00919988498802,sofia/external/1001 at 172.16.100.155,1a9da86a-bc16-3e49-9b08-94facc657843,1001,fring,919988498802,sofia/external/919988498802,9430553a-ee00-ed4f-b109-41eabdd4dc7e > > 2011-10-20 > 04:35:24,1319078124,switch_ivr_signal_bridge,1001,1001,r00919976131008,sofia/external/1001 at 172.16.100.155,12d0429e-a539-6944-831e-a862183d905b,1001,fring,919976131008,sofia/external/919976131008,f672283c-a37e-e947-b45a-b7e66ef7881a > > 2011-10-20 > 05:22:23,1319080943,switch_ivr_signal_bridge,1001,1001,r00919973840848,sofia/external/1001 at 172.16.100.155,42e29646-c0c4-d64a-9cf5-7c37aeabc1a6,1001,fring,919973840848,sofia/external/919973840848,1d2ef437-17aa-db4b-86d9-6d3d5310e280 > > 2011-10-20 > 05:37:34,1319081854,switch_ivr_signal_bridge,1001,1001,r00919840288799,sofia/external/1001 at 172.16.100.155,f48fed29-8041-3545-bffc-5050bc19cbc6,1001,fring,919840288799,sofia/external/919840288799,df29a0df-3ef8-4b43-807e-73e0c0e25f3b > > 2011-10-20 > 10:00:26,1319097626,switch_ivr_signal_bridge,1001,1001,r0091955445666,sofia/external/1001 at 172.16.100.155,931102bc-ac35-3140-acab-988224d502b0,1001,fring,91955445666,sofia/external/91955445666,9283eb63-bd0a-6d43-9d8a-7985ec48e23c > > 2011-10-20 > 12:16:27,1319105787,switch_ivr_signal_bridge,1001,1001,r00919855120931,sofia/external/1001 at 172.16.100.155,db0f66aa-69f1-fe47-85af-de610a77d517,1001,fring,919855120931,sofia/external/919855120931,ceab5ae5-9b8f-3d4c-b149-c248567ab5d4 > > 2011-10-20 > 12:42:02,1319107322,switch_ivr_signal_bridge,1001,1001,r00918872855601,sofia/external/1001 at 172.16.100.155,95249509-e793-0d42-9948-5ef53898b942,1001,fring,918872855601,sofia/external/918872855601,8c9fc203-8d54-aa4f-983f-889fbf0ca883 > > 2011-10-20 > 13:45:10,1319111110,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/1001 at 172.16.100.155,7b645b14-5f89-a64c-b059-2574b2e4e69d,1001,fring,919988175489,sofia/external/919988175489,55f50da5-5679-fe4f-9a7f-b96111f3cd67 > > 2011-10-20 > 14:49:50,1319114990,switch_ivr_signal_bridge,1001,1001,r00919663195202,sofia/external/1001 at 172.16.100.155,3601f4b8-c0ff-1a4c-9f76-47e4fa95b549,1001,fring,919663195202,sofia/external/919663195202,a22e2d11-0463-c845-a953-a12b49cfe863 > > 2011-10-20 > 15:10:47,1319116247,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/1001 at 172.16.100.155,e09d3de0-c759-fc4b-96f1-f01353a4297d,1001,fring,919988175489,sofia/external/919988175489,23d4b490-cbb7-7a45-86af-705c45d07fc7 > > 2011-10-20 > 20:52:17,1319136737,switch_ivr_signal_bridge,1001,1001,r009888560750,sofia/external/1001 at 172.16.100.155,ff8ab1f6-f1c6-224f-928b-1c042b0ed908,1001,fring,9888560750,sofia/external/9888560750,c37b8ebe-2264-934f-a187-6b14c7dcb308 > > 2011-10-20 > 22:32:38,1319142758,switch_ivr_signal_bridge,1001,1001,r008801816630096,sofia/external/1001 at 172.16.100.155,a38182cb-b661-854d-b243-a830ab0bf992,1001,fring,8801816630096,sofia/external/8801816630096,f88dc530-303e-d041-83cb-eca118780ca2 > > 2011-10-20 > 22:34:32,1319142872,switch_ivr_signal_bridge,1001,1001,r00919161209997,sofia/external/1001 at 172.16.100.155,0add3c1c-7fb5-2447-a9d6-d0622886fcf9,1001,fring,919161209997,sofia/external/919161209997,2b91f5de-4668-0244-ac76-d031fe827a25 > > 2011-10-20 > 23:56:16,1319147776,switch_ivr_signal_bridge,1001,1001,r00985468962,sofia/external/1001 at 172.16.100.155,f7137d11-0a87-6a4b-b5b3-b43eeeb2b462,1001,fring,985468962,sofia/external/985468962,025f197b-c5aa-5b4b-8606-20fb6f17e888 > > 2011-10-21 > 04:36:00,1319164560,switch_ivr_signal_bridge,1001,1001,r00917597234252,sofia/external/1001 at 172.16.100.155,edd2f16b-9a6a-f749-bd62-4d303bc09442,1001,fring,917597234252,sofia/external/917597234252,1c55b21b-9c27-b647-8ba5-1ffce398c747 > > 2011-10-21 > 04:47:03,1319165223,switch_ivr_signal_bridge,1001,1001,r00919541622040,sofia/external/1001 at 172.16.100.155,336c146b-c96d-494e-b378-c024bcfc0c94,1001,fring,919541622040,sofia/external/919541622040,746877e1-c509-8a45-934e-f3f4b7f82db2 > > 2011-10-21 > 07:59:54,1319176794,switch_ivr_signal_bridge,1001,1001,r009613235498,sofia/external/1001 at 172.16.100.155,16575979-c92a-5842-879f-e9f6448c7d54,1001,fring,9613235498,sofia/external/9613235498,bb05fd5a-4a6f-3248-81a9-a81074a9eacd > > 2011-10-21 > 08:02:05,1319176925,switch_ivr_signal_bridge,1001,1001,r00912026336917,sofia/external/1001 at 172.16.100.155,ebb234aa-8edf-2e4b-b641-602478b25782,1001,fring,912026336917,sofia/external/912026336917,f6a0e4b1-69f7-dc49-a5bf-41e6d8c7c96d > > 2011-10-21 > 08:51:57,1319179917,switch_ivr_signal_bridge,1001,1001,r00919372875359,sofia/external/1001 at 172.16.100.155,b090dc91-932e-1246-abb8-8fc949131ae6,1001,fring,919372875359,sofia/external/919372875359,3a4702a6-e7cc-cb41-ad99-defa80b14680 > > 2011-10-21 > 09:15:47,1319181347,switch_ivr_signal_bridge,1001,1001,r00919915709069,sofia/external/1001 at 172.16.100.155,ae308194-ce89-4945-b8de-6923ce6d0dc0,1001,fring,919915709069,sofia/external/919915709069,502ffe82-0fb9-b14a-9421-4acd25a0b4df > > 2011-10-21 > 11:35:16,1319189716,switch_ivr_signal_bridge,1001,1001,r00919847023725,sofia/external/1001 at 172.16.100.155,c7cb7640-7313-d742-b756-a7713273ccea,1001,fring,919847023725,sofia/external/919847023725,4ab50403-04ac-bc40-a759-40f2a62f1699 > > 2011-10-21 > 14:48:46,1319201326,switch_ivr_signal_bridge,1001,1001,r00919878738103,sofia/external/1001 at 172.16.100.155,91626ac0-9e14-a34d-94f5-b1fbfe7e7629,1001,fring,919878738103,sofia/external/919878738103,39c58424-b087-6749-9ba9-c9a821f46895 > > 2011-10-21 > 17:41:28,1319211688,switch_ivr_signal_bridge,1001,1001,r009843648082,sofia/external/1001 at 172.16.100.155,3929dc6b-b7de-c648-8eae-3d749f0ac0fa,1001,fring,9843648082,sofia/external/9843648082,6653a9ed-d23b-564e-98e5-9a56a40e499c > > 2011-10-22 > 05:07:21,1319252841,switch_ivr_signal_bridge,1001,1001,r00919036392951,sofia/external/1001 at 172.16.100.155,7cd56510-2983-554a-9fa2-a27c3fb1f9e6,1001,fring,919036392951,sofia/external/919036392951,e63642ec-d43a-c243-9291-0ea6725d04bb > > 2011-10-22 > 06:42:56,1319258576,switch_ivr_signal_bridge,1001,1001,r00918870521722,sofia/external/1001 at 172.16.100.155,7dd85bc6-e00e-c44e-bd42-89c6d438485e,1001,fring,918870521722,sofia/external/918870521722,8d7e1c17-2ad9-2447-b91b-d2f2b2e27e4c > > 2011-10-22 > 08:36:26,1319265386,switch_ivr_signal_bridge,1001,1001,r00911874289543,sofia/external/1001 at 172.16.100.155,1214a817-f235-1a4e-a21c-d2363e9e5760,1001,fring,911874289543,sofia/external/911874289543,13f85a5d-eb37-0c4a-a15e-2c86dbc34817 > > 2011-10-22 > 12:14:32,1319278472,switch_ivr_signal_bridge,1001,1001,r00918988262666,sofia/external/1001 at 172.16.100.155,1b5cb871-e043-f244-9942-251a7a492136,1001,fring,918988262666,sofia/external/918988262666,21f328fb-0dfe-2643-957b-302f6a80f17f > > 2011-10-22 > 16:28:35,1319293715,switch_ivr_signal_bridge,1001,1001,r00918722222400,sofia/external/1001 at 172.16.100.155,912e1b07-a99b-bf4d-9e2f-0b5990a68821,1001,fring,918722222400,sofia/external/918722222400,8b417629-3161-f141-93d7-bb0ddd7982d5 > > 2011-10-22 > 17:32:10,1319297530,switch_ivr_signal_bridge,1001,1001,r00919839884297,sofia/external/1001 at 172.16.100.155,ec7decdd-c378-6c45-a888-9b962e4ecece,1001,fring,919839884297,sofia/external/919839884297,f12896b7-9605-9b40-bdc9-face2181146b > > 2011-10-23 > 06:26:42,1319344002,switch_ivr_signal_bridge,1001,1001,r00919567382898,sofia/external/1001 at 172.16.100.155,db2e199b-7b9f-fe4d-9742-7415dded69c7,1001,fring,919567382898,sofia/external/919567382898,dd8bb627-c804-2f49-8b5b-5acf5b3423d4 > > 2011-10-23 > 07:21:29,1319347289,switch_ivr_signal_bridge,1001,1001,r00966537646047,sofia/external/1001 at 172.16.100.155,e97ba53e-1ef6-2944-b53b-b3abe5dfef36,1001,fring,966537646047,sofia/external/966537646047,9eea7d3f-5092-de4b-9291-e7f0a37db2a1 > > 2011-10-23 > 08:12:33,1319350353,switch_ivr_signal_bridge,1001,1001,r00914942588715,sofia/external/1001 at 172.16.100.155,30ca9d2d-5cae-6645-a16c-6ae8735b6316,1001,fring,914942588715,sofia/external/914942588715,6774d23c-f131-7d41-854d-2fcff77c2cf6 > > 2011-10-23 > 13:04:33,1319367873,switch_ivr_signal_bridge,1001,1001,r00919625833433,sofia/external/1001 at 172.16.100.155,284691f9-dd3b-cc4a-97c7-9465bd3a6b6c,1001,fring,919625833433,sofia/external/919625833433,4efb7446-11f2-c84c-88c4-10ab97eddedf > > 2011-10-23 > 14:35:20,1319373320,switch_ivr_signal_bridge,1001,1001,r00919888880401,sofia/external/1001 at 172.16.100.155,475abe11-2137-c94a-ada7-825ad8524bd8,1001,fring,919888880401,sofia/external/919888880401,5e06efb5-2c72-ba4f-9443-c15568ad2608 > > 2011-10-23 > 15:59:20,1319378360,switch_ivr_signal_bridge,1001,1001,r009878539161,sofia/external/1001 at 172.16.100.155,aa5c4f8f-5a69-e947-a239-75f4a584b9f0,1001,fring,9878539161,sofia/external/9878539161,366b56d1-3ce4-fa43-9d3d-872c04f5229f > > 2011-10-23 > 20:52:23,1319395943,switch_ivr_signal_bridge,1001,1001,r0017787079272,sofia/external/1001 at 172.16.100.155,3f9c8b11-2e7a-e641-8cfc-3b7dc65e7118,1001,fring,17787079272,sofia/external/17787079272,d5b4a2e0-ee48-7048-b54c-f6ea968428af > > 2011-10-24 > 10:23:29,1319444609,switch_ivr_signal_bridge,1001,1001,r00919881375673,sofia/external/1001 at 172.16.100.155,527d1ca5-3f05-a54b-8b0c-192b7515eebb,1001,fring,919881375673,sofia/external/919881375673,4d1da2aa-87c1-7d41-b57d-05034e503ede > > 2011-10-24 > 10:29:44,1319444984,switch_ivr_signal_bridge,1001,1001,r0061430594924,sofia/external/1001 at 172.16.100.155,6ce52143-219d-ed40-a27f-fc519eef5092,1001,fring,61430594924,sofia/external/61430594924,11ef495a-d651-564e-9479-94fa579a5286 > > 2011-10-24 > 10:56:32,1319446592,switch_ivr_signal_bridge,1001,1001,r00919003574979,sofia/external/1001 at 172.16.100.155,8d83282b-f96b-0a43-ac58-1f3d3e4e26f0,1001,fring,919003574979,sofia/external/919003574979,2bd6e807-c5ef-8c47-807f-1e5171bcbc3d > > 2011-10-24 > 13:58:59,1319457539,switch_ivr_signal_bridge,1001,1001,r00919876395511,sofia/external/1001 at 172.16.100.155,68fdbb19-f76f-0a40-b46b-e0296eed8690,1001,fring,919876395511,sofia/external/919876395511,6cf77cf5-8cd9-f340-98c0-bb896304dbf9 > > 2011-10-24 > 14:16:45,1319458605,switch_ivr_signal_bridge,1001,1001,r00919988170711,sofia/external/1001 at 172.16.100.155,57e0187b-9d25-044c-bffb-9d11431c251a,1001,fring,919988170711,sofia/external/919988170711,b71d6822-289f-614b-8e5b-2bc20225d9ef > > 2011-10-25 > 00:16:09,1319494569,switch_ivr_signal_bridge,1001,1001,r005016630551,sofia/external/1001 at 172.16.100.155,f5e81b48-8f86-4644-b206-cb066e15418d,1001,fring,5016630551,sofia/external/5016630551,b1467ed2-56e3-bb42-ba6c-cdc272b2533b > > 2011-10-25 > 02:30:30,1319502630,switch_ivr_signal_bridge,1001,1001,r00919729325377,sofia/external/1001 at 172.16.100.155,f6eb7c48-cc1d-9b42-bbac-46ac759de0e0,1001,fring,919729325377,sofia/external/919729325377,d8eb68b9-ab39-814b-a70d-915347e2afe7 > > 2011-10-25 > 09:35:05,1319528105,switch_ivr_signal_bridge,1001,1001,r00966554656169,sofia/external/1001 at 172.16.100.155,03c9a95e-4d36-8847-a2a2-3545c38e233e,1001,fring,966554656169,sofia/external/966554656169,1d2927f5-7202-914c-a463-6dcd00588cfc > > 2011-10-25 > 12:53:58,1319540038,switch_ivr_signal_bridge,1001,1001,r00201094811534,sofia/external/1001 at 172.16.100.155,3bfe5634-dcf9-c84d-ba72-26f8fee4a8eb,1001,fring,201094811534,sofia/external/201094811534,d0c58bb9-53a3-0549-9b78-4ca5f14fcb63 > > 2011-10-26 > 04:22:12,1319595732,switch_ivr_signal_bridge,1001,1001,r00913871398117,sofia/external/1001 at 172.16.100.155,1479c10c-9922-854c-8030-6f2a6ce4f605,1001,fring,913871398117,sofia/external/913871398117,fbdc8cf2-e898-0340-bc48-b4ad7c54ac0e > > 2011-10-26 > 10:01:27,1319616087,switch_ivr_signal_bridge,1001,1001,r00201113961426,sofia/external/1001 at 172.16.100.155,db18777e-aefa-c248-84ea-06150fe6989f,1001,fring,201113961426,sofia/external/201113961426,0a5572c6-54b7-1d40-bf4e-918e6880b3ce > > > > 147 total. > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Regards > Vellayappan > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From turqmr2 at gmail.com Tue Nov 8 16:03:11 2011 From: turqmr2 at gmail.com (Jacob Smith) Date: Tue, 08 Nov 2011 08:03:11 -0500 Subject: [Freeswitch-users] Help with home setup Message-ID: <4EB9288F.6050705@gmail.com> I have been trying to get this working for the last week and would really appreciate your help. I have a pfSense 2.0 server with Freeswitch and FusionPBX installed. I followed the instructions at http://wiki.fusionpbx.com/index.php?title=PfSense_Install and can access the web GUI. My goal is to use a Gigaset C610A-IP connected to my Google Voice account. However, my first failure seems to be the most basic: through FusionPBX, I created a user and extension to use with the phone but I can not register it with FreeSwitch. So, how do I get the phone to work so I can test that the basic functions of FreeSwitch are working and move on? Thanks! From msc at freeswitch.org Tue Nov 8 19:12:26 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 08:12:26 -0800 Subject: [Freeswitch-users] Issues using loopback In-Reply-To: References: Message-ID: You should ask on the Newfies mailing list about this one. I don't know if loopback is supported or not, but Areski can probably answer that question for you. -MC On Tue, Nov 8, 2011 at 5:18 AM, Dip Mehta wrote: > Hi, > > I am trying to integrate Newfies with Freeswitch. > > I tried using loopback to route it to 9171 to default context. However, it > doesnot works. When the customer picks up the call, there is a dead air. > > Pleae advice. > > Logs for your refernce. > > 13526785574 = external number > > 2011-11-08 08:11:21.192674 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute set(hangup_after_bridge=false) > EXECUTE sofia/external/13526785574 set(hangup_after_bridge=false) > 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1177 sofia/external/ > 13526785574 SET [hangup_after_bridge]=[false] > 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute unset(call_timeout) > EXECUTE sofia/external/13526785574 unset(call_timeout) > 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1294 UNSET [call_timeout] > 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute unset(effective_caller_id_number) > EXECUTE sofia/external/13526785574 unset(effective_caller_id_number) > 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1294 UNSET > [effective_caller_id_number] > 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute unset(effective_caller_id_name) > EXECUTE sofia/external/13526785574 unset(effective_caller_id_name) > 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1294 UNSET > [effective_caller_id_name] > 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute set(continue_on_fail=true) > EXECUTE sofia/external/13526785574 set(continue_on_fail=true) > 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1177 sofia/external/ > 13526785574 SET [continue_on_fail]=[true] > 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.212673 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute set(hangup_after_bridge=false) > EXECUTE sofia/external/13526785574 set(hangup_after_bridge=false) > 2011-11-08 08:11:21.212673 [DEBUG] mod_dptools.c:1177 sofia/external/ > 13526785574 SET [hangup_after_bridge]=[false] > 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.212673 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute set(plivo_dial_rang=false) > EXECUTE sofia/external/13526785574 set(plivo_dial_rang=false) > 2011-11-08 08:11:21.232676 [DEBUG] mod_dptools.c:1177 sofia/external/ > 13526785574 SET [plivo_dial_rang]=[false] > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute unset(bridge_terminate_key) > EXECUTE sofia/external/13526785574 unset(bridge_terminate_key) > 2011-11-08 08:11:21.232676 [DEBUG] mod_dptools.c:1294 UNSET > [bridge_terminate_key] > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute set(bridge_early_media=false) > EXECUTE sofia/external/13526785574 set(bridge_early_media=false) > 2011-11-08 08:11:21.232676 [DEBUG] mod_dptools.c:1177 sofia/external/ > 13526785574 SET [bridge_early_media]=[false] > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute set(instant_ringback=true) > EXECUTE sofia/external/13526785574 set(instant_ringback=true) > 2011-11-08 08:11:21.232676 [DEBUG] mod_dptools.c:1177 sofia/external/ > 13526785574 SET [instant_ringback]=[true] > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute set(ringback=$${us-ring}}) > EXECUTE sofia/external/13526785574 set(ringback=$%(2000,4000,440,480)}) > 2011-11-08 08:11:21.232676 [DEBUG] mod_dptools.c:1177 sofia/external/ > 13526785574 SET [ringback]=[$%(2000,4000,440,480)}] > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute ring_ready() > EXECUTE sofia/external/13526785574 ring_ready() > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute bridge( 225b5e18-0a0b-11e1-ba82-590058a85379 plivo_dial_rang > true',api_on_pre_answer='uuid_setvar 225b5e18-0a0b-11e1-ba82-590058a85379 > plivo_dial_rang true',api_on_answer_1='sched_api +300 > 26f2e3f6-0a0b-11e1-8b92-bc305bec2901 uuid_transfer > 225b5e18-0a0b-11e1-ba82-590058a85379 -bleg hangup:ALLOTTED_TIMEOUT > inline'>[leg_timeout=10]loopback/9171/default/xml:_:) > EXECUTE sofia/external/13526785574 bridge( 225b5e18-0a0b-11e1-ba82-590058a85379 plivo_dial_rang > true',api_on_pre_answer='uuid_setvar 225b5e18-0a0b-11e1-ba82-590058a85379 > plivo_dial_rang true',api_on_answer_1='sched_api +300 > 26f2e3f6-0a0b-11e1-8b92-bc305bec2901 uuid_transfer > 225b5e18-0a0b-11e1-ba82-590058a85379 -bleg hangup:ALLOTTED_TIMEOUT > inline'>[leg_timeout=10]loopback/9171/default/xml:_:) > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:1413 Parsing > ultra-global variables > 2011-11-08 08:11:21.232676 [DEBUG] switch_event.c:1521 Parsing variable > [api_on_ring]=[uuid_setvar 225b5e18-0a0b-11e1-ba82-590058a85379 > plivo_dial_rang true] > 2011-11-08 08:11:21.232676 [DEBUG] switch_event.c:1521 Parsing variable > [api_on_pre_answer]=[uuid_setvar 225b5e18-0a0b-11e1-ba82-590058a85379 > plivo_dial_rang true] > 2011-11-08 08:11:21.232676 [DEBUG] switch_event.c:1521 Parsing variable > [api_on_answer_1]=[sched_api +300 26f2e3f6-0a0b-11e1-8b92-bc305bec2901 > uuid_transfer 225b5e18-0a0b-11e1-ba82-590058a85379 -bleg > hangup:ALLOTTED_TIMEOUT inline] > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:2299 Parsing > session specific variables > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2011-11-08 08:11:21.232676 [DEBUG] switch_event.c:1521 Parsing variable > [leg_timeout]=[10] > 2011-11-08 08:11:21.232676 [WARNING] switch_ivr_originate.c:1903 No > origination URL specified! > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:3348 Originate > Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] > 2011-11-08 08:11:21.232676 [NOTICE] switch_channel.c:915 New Channel > loopback/9171/default/xml-a [26f5283c-0a0b-11e1-ba88-590058a85379] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:143 > loopback/9171/default/xml-a setup codec L16/8000/20 > 2011-11-08 08:11:21.232676 [NOTICE] switch_channel.c:913 Rename Channel > loopback/9171/default/xml-a->loopback/9171-a > [26f5283c-0a0b-11e1-ba88-590058a85379] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:943 (loopback/9171-a) > State Change CS_NEW -> CS_INIT > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-a [BREAK] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-a > CHANNEL KILL > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:2554 > loopback/9171-a Setting leg timeout to 10 > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 > (loopback/9171-a) Running State Change CS_INIT > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:401 > (loopback/9171-a) State INIT > 2011-11-08 08:11:21.232676 [NOTICE] switch_channel.c:915 New Channel > loopback/9171-b [26f53af2-0a0b-11e1-ba8a-590058a85379] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:143 loopback/9171-b > setup codec L16/8000/20 > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:256 (loopback/9171-b) > State Change CS_NEW -> CS_INIT > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-b [BREAK] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:302 (loopback/9171-a) > State Change CS_INIT -> CS_ROUTING > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-a [BREAK] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-a > CHANNEL KILL > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:401 > (loopback/9171-a) State INIT going to sleep > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 > (loopback/9171-b) Running State Change CS_INIT > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:401 > (loopback/9171-b) State INIT > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:302 (loopback/9171-b) > State Change CS_INIT -> CS_ROUTING > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-b [BREAK] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:401 > (loopback/9171-b) State INIT going to sleep > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 > (loopback/9171-a) Running State Change CS_ROUTING > 2011-11-08 08:11:21.232676 [DEBUG] switch_channel.c:1844 (loopback/9171-a) > Callstate Change DOWN -> RINGING > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 > (loopback/9171-b) Running State Change CS_ROUTING > 2011-11-08 08:11:21.232676 [DEBUG] switch_channel.c:1844 (loopback/9171-b) > Callstate Change DOWN -> RINGING > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:410 > (loopback/9171-a) State ROUTING > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:334 loopback/9171-a > CHANNEL ROUTING > 2011-11-08 08:11:21.232676 [DEBUG] switch_ivr_originate.c:66 > (loopback/9171-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-a [BREAK] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-a > CHANNEL KILL > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:410 > (loopback/9171-a) State ROUTING going to sleep > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 > (loopback/9171-a) Running State Change CS_CONSUME_MEDIA > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:410 > (loopback/9171-b) State ROUTING > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:429 > (loopback/9171-a) State CONSUME_MEDIA > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:334 loopback/9171-b > CHANNEL ROUTING > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:533 CHANNEL CONSUME_MEDIA > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:104 > loopback/9171-b Standard ROUTING > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:429 > (loopback/9171-a) State CONSUME_MEDIA going to sleep > 2011-11-08 08:11:21.232676 [INFO] mod_dialplan_xml.c:336 Processing > Outbound Call <13526785574>->9171 in context default > Dialplan: loopback/9171-b parsing [default->plivo] continue=false > Dialplan: loopback/9171-b Regex (PASS) [plivo] destination_number(9171) =~ > /^(\d+)$/ break=on-false > Dialplan: loopback/9171-b Action socket(127.0.0.1:8084 async full) > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:154 > (loopback/9171-b) State Change CS_ROUTING -> CS_EXECUTE > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-b [BREAK] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:410 > (loopback/9171-b) State ROUTING going to sleep > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:362 > (loopback/9171-b) Running State Change CS_EXECUTE > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:417 > (loopback/9171-b) State EXECUTE > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:373 loopback/9171-b > CHANNEL EXECUTE > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_state_machine.c:192 > loopback/9171-b Standard EXECUTE > EXECUTE loopback/9171-b socket(127.0.0.1:8084 async full) > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1009 Send signal > loopback/9171-b [BREAK] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.232676 [DEBUG] switch_core_session.c:1009 Send signal > loopback/9171-b [BREAK] > 2011-11-08 08:11:21.232676 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1009 Send signal > loopback/9171-b [BREAK] > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] mod_event_socket.c:2617 > (loopback/9171-b) State Change CS_EXECUTE -> CS_RESET > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-b [BREAK] > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] switch_ivr.c:577 loopback/9171-b > Command Execute set(plivo_app=true) > EXECUTE loopback/9171-b set(plivo_app=true) > 2011-11-08 08:11:21.252674 [DEBUG] mod_dptools.c:1177 loopback/9171-b SET > [plivo_app]=[true] > 2011-11-08 08:11:21.252674 [DEBUG] switch_ivr.c:577 loopback/9171-b > Command Execute set(hangup_after_bridge=false) > EXECUTE loopback/9171-b set(hangup_after_bridge=false) > 2011-11-08 08:11:21.252674 [DEBUG] mod_dptools.c:1177 loopback/9171-b SET > [hangup_after_bridge]=[false] > 2011-11-08 08:11:21.252674 [DEBUG] switch_ivr.c:577 loopback/9171-b > Command Execute hangup() > EXECUTE loopback/9171-b hangup() > 2011-11-08 08:11:21.252674 [DEBUG] switch_channel.c:2804 (loopback/9171-b) > Callstate Change RINGING -> HANGUP > 2011-11-08 08:11:21.252674 [NOTICE] mod_dptools.c:1030 Hangup > loopback/9171-b [CS_RESET] [NORMAL_CLEARING] > 2011-11-08 08:11:21.252674 [DEBUG] switch_channel.c:2820 Send signal > loopback/9171-b [KILL] > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-b [BREAK] > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:2262 > loopback/9171-b skip receive message [APPLICATION_EXEC_COMPLETE] (channel > is hungup already) > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:2262 > loopback/9171-b skip receive message [APPLICATION_EXEC_COMPLETE] (channel > is hungup already) > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:417 > (loopback/9171-b) State EXECUTE going to sleep > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:362 > (loopback/9171-b) Running State Change CS_HANGUP > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:602 > (loopback/9171-b) State HANGUP > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:425 loopback/9171-b > CHANNEL HANGUP > 2011-11-08 08:11:21.252674 [DEBUG] switch_channel.c:2804 (loopback/9171-a) > Callstate Change RINGING -> HANGUP > 2011-11-08 08:11:21.252674 [NOTICE] mod_loopback.c:436 Hangup > loopback/9171-a [CS_CONSUME_MEDIA] [NORMAL_CLEARING] > 2011-11-08 08:11:21.252674 [DEBUG] switch_channel.c:2820 Send signal > loopback/9171-a [KILL] > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-a > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-a [BREAK] > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-a > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:47 > loopback/9171-b Standard HANGUP, cause: NORMAL_CLEARING > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:602 > (loopback/9171-b) State HANGUP going to sleep > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:362 > (loopback/9171-a) Running State Change CS_HANGUP > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:393 > (loopback/9171-b) State Change CS_HANGUP -> CS_REPORTING > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-b [BREAK] > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:362 > (loopback/9171-b) Running State Change CS_REPORTING > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:662 > (loopback/9171-b) State REPORTING > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:602 > (loopback/9171-a) State HANGUP > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:425 loopback/9171-a > CHANNEL HANGUP > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:47 > loopback/9171-a Standard HANGUP, cause: NORMAL_CLEARING > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:602 > (loopback/9171-a) State HANGUP going to sleep > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:393 > (loopback/9171-a) State Change CS_HANGUP -> CS_REPORTING > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-a [BREAK] > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-a > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:362 > (loopback/9171-a) Running State Change CS_REPORTING > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:662 > (loopback/9171-a) State REPORTING > 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:366 Got error [0] posting > to web server [NEWFIES_API_STORE_CDR] > 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:373 Retry will be with url > [NEWFIES_API_STORE_CDR] > 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:384 Unable to post to web > server, writing to file > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:79 > loopback/9171-b Standard REPORTING, cause: NORMAL_CLEARING > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:662 > (loopback/9171-b) State REPORTING going to sleep > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:387 > (loopback/9171-b) State Change CS_REPORTING -> CS_DESTROY > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-b [BREAK] > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-b > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1377 Session 148 > (loopback/9171-b) Locked, Waiting on external entities > 2011-11-08 08:11:21.252674 [NOTICE] switch_core_session.c:1395 Session 148 > (loopback/9171-b) Ended > 2011-11-08 08:11:21.252674 [NOTICE] switch_core_session.c:1397 Close > Channel loopback/9171-b [CS_DESTROY] > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:491 > (loopback/9171-b) Callstate Change HANGUP -> DOWN > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:494 > (loopback/9171-b) Running State Change CS_DESTROY > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:504 > (loopback/9171-b) State DESTROY > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:86 > loopback/9171-b Standard DESTROY > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:504 > (loopback/9171-b) State DESTROY going to sleep > 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:366 Got error [0] posting > to web server [NEWFIES_API_STORE_CDR] > 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:373 Retry will be with url > [NEWFIES_API_STORE_CDR] > 2011-11-08 08:11:21.252674 [ERR] mod_xml_cdr.c:384 Unable to post to web > server, writing to file > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:79 > loopback/9171-a Standard REPORTING, cause: NORMAL_CLEARING > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:662 > (loopback/9171-a) State REPORTING going to sleep > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:387 > (loopback/9171-a) State Change CS_REPORTING -> CS_DESTROY > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1177 Send signal > loopback/9171-a [BREAK] > 2011-11-08 08:11:21.252674 [DEBUG] mod_loopback.c:473 loopback/9171-a > CHANNEL KILL > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_session.c:1377 Session 147 > (loopback/9171-a) Locked, Waiting on external entities > 2011-11-08 08:11:21.252674 [DEBUG] switch_ivr_originate.c:3348 Originate > Resulted in Error Cause: 16 [NORMAL_CLEARING] > 2011-11-08 08:11:21.252674 [NOTICE] switch_core_session.c:1395 Session 147 > (loopback/9171-a) Ended > 2011-11-08 08:11:21.252674 [NOTICE] switch_core_session.c:1397 Close > Channel loopback/9171-a [CS_DESTROY] > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:491 > (loopback/9171-a) Callstate Change HANGUP -> DOWN > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:494 > (loopback/9171-a) Running State Change CS_DESTROY > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:504 > (loopback/9171-a) State DESTROY > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:86 > loopback/9171-a Standard DESTROY > 2011-11-08 08:11:21.252674 [DEBUG] switch_core_state_machine.c:504 > (loopback/9171-a) State DESTROY going to sleep > 2011-11-08 08:11:21.272673 [DEBUG] switch_rtp.c:3181 Correct ip/port > confirmed. > 2011-11-08 08:11:21.312678 [INFO] mod_dptools.c:2811 Originate Failed. > Cause: NORMAL_CLEARING > 2011-11-08 08:11:21.312678 [DEBUG] switch_core_session.c:1009 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.312678 [DEBUG] mod_event_socket.c:2617 (sofia/external/ > 13526785574) State Change CS_EXECUTE -> CS_RESET > 2011-11-08 08:11:21.312678 [DEBUG] switch_core_session.c:1177 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.332673 [DEBUG] switch_ivr.c:577 sofia/external/ > 13526785574 Command Execute hangup() > EXECUTE sofia/external/13526785574 hangup() > 2011-11-08 08:11:21.332673 [DEBUG] switch_channel.c:2804 (sofia/external/ > 13526785574) Callstate Change ACTIVE -> HANGUP > 2011-11-08 08:11:21.332673 [NOTICE] mod_dptools.c:1030 Hangup > sofia/external/13526785574 [CS_RESET] [NORMAL_CLEARING] > 2011-11-08 08:11:21.332673 [DEBUG] switch_channel.c:2820 Send signal > sofia/external/13526785574 [KILL] > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:1177 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:2262 > sofia/external/13526785574 skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:2262 > sofia/external/13526785574 skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:417 > (sofia/external/13526785574) State EXECUTE going to sleep > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/13526785574) Running State Change CS_HANGUP > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/13526785574) State HANGUP > 2011-11-08 08:11:21.332673 [DEBUG] mod_sofia.c:465 Channel sofia/external/ > 13526785574 hanging up, cause: NORMAL_CLEARING > 2011-11-08 08:11:21.332673 [DEBUG] mod_sofia.c:509 Sending BYE to > sofia/external/13526785574 > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:47 > sofia/external/13526785574 Standard HANGUP, cause: NORMAL_CLEARING > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/13526785574) State HANGUP going to sleep > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:393 > (sofia/external/13526785574) State Change CS_HANGUP -> CS_REPORTING > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:1177 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/13526785574) Running State Change CS_REPORTING > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/13526785574) State REPORTING > 2011-11-08 08:11:21.332673 [ERR] mod_xml_cdr.c:366 Got error [0] posting > to web server [NEWFIES_API_STORE_CDR] > 2011-11-08 08:11:21.332673 [ERR] mod_xml_cdr.c:373 Retry will be with url > [NEWFIES_API_STORE_CDR] > 2011-11-08 08:11:21.332673 [ERR] mod_xml_cdr.c:384 Unable to post to web > server, writing to file > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:79 > sofia/external/13526785574 Standard REPORTING, cause: NORMAL_CLEARING > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/13526785574) State REPORTING going to sleep > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:387 > (sofia/external/13526785574) State Change CS_REPORTING -> CS_DESTROY > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:1177 Send signal > sofia/external/13526785574 [BREAK] > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_session.c:1377 Session 146 > (sofia/external/13526785574) Locked, Waiting on external entities > 2011-11-08 08:11:21.332673 [NOTICE] switch_core_session.c:1395 Session 146 > (sofia/external/13526785574) Ended > 2011-11-08 08:11:21.332673 [NOTICE] switch_core_session.c:1397 Close > Channel sofia/external/13526785574 [CS_DESTROY] > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:491 > (sofia/external/13526785574) Callstate Change HANGUP -> DOWN > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:494 > (sofia/external/13526785574) Running State Change CS_DESTROY > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/13526785574) State DESTROY > 2011-11-08 08:11:21.332673 [DEBUG] mod_sofia.c:370 sofia/external/ > 13526785574 SOFIA DESTROY > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:86 > sofia/external/13526785574 Standard DESTROY > 2011-11-08 08:11:21.332673 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/13526785574) 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 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/e59056b2/attachment-0001.html From msc at freeswitch.org Tue Nov 8 19:18:20 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 08:18:20 -0800 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> Message-ID: You can use mod_local_stream. Just set up a stream like the hold music on but using the recorded message. Note: a local stream just plays over and over again in a loop, so each call won't necessarily start at the beginning of the message. -MC On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers wrote: > Hi,**** > > ** ** > > I would like to use freeswitch to host a one to many broadcast style > conference call. The challenge I have is that there won?t always be any > listeners so the conference application isn?t quite working for me.**** > > ** ** > > Essentially, the speaker dials in with something important to say, by > default the system will simply record what they say as an archived message > and then they hang up. This works fine but I would like to extend the > functionality so that other people dialling in have the option to connect > into this call and listen to the message live.**** > > ** ** > > Can you advise me on the best way of achieving this: conference, bridge, > eavesdrop or something else?**** > > ** ** > > Many thanks**** > > Andy**** > > ** ** > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/03e32f04/attachment.html From msc at freeswitch.org Tue Nov 8 19:21:46 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 08:21:46 -0800 Subject: [Freeswitch-users] paramter "NDLB-force-rport" did not effect In-Reply-To: References: Message-ID: Check this one out: http://wiki.freeswitch.org/wiki/NAT_Traversal#NDLB-connectile-dysfunction -MC On Mon, Nov 7, 2011 at 11:30 PM, fieldpeak wrote: > Dear Friends, > > i met a problem that call a registered user behind NAT, i need FS send > INVITE message to the port of the NAT device from which the registered > user sent out REGISTRATION but not according to the CONTACT header. > > see the wiki, i enable below paramter, > > > >From the wireshark trace, it shows FS received the REGISTERATION > message from remote port 20438, however, the INVITE message still sent > to the port(20347) inside "Contact", it looks the paramter > "NDLB-force-rport" did not effect, Could anyone help for this issue, > thanks a lot! > > Below is the registeration info from FS output (press F9), > > > Call-ID: 1721091027 at 10.33.48.203 > User: 13580358068 at 124.193.106.104 > Contact: "user" > Agent: > Status: Registered(UDP)(unknown) EXP(2011-11-08 14:38:02) > EXPSECS(152) > Host: freeswitch > IP: 61.144.248.77 > Port: 20438 > Auth-User: 13580358068 > Auth-Realm: 124.193.106.104 > MWI-Account: 13580358068 at 124.193.106.104 > > Cheers! > BR, > Charles > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/2800900f/attachment.html From anthony.minessale at gmail.com Tue Nov 8 19:23:33 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Nov 2011 10:23:33 -0600 Subject: [Freeswitch-users] paramter "NDLB-force-rport" did not effect In-Reply-To: References: Message-ID: I think you need to review your understanding of what rport is and what force-rport does. On Tue, Nov 8, 2011 at 1:30 AM, fieldpeak wrote: > Dear Friends, > > i met a problem that call a registered user behind NAT, i need FS send > INVITE message to the port of the NAT device from which the registered > user sent out REGISTRATION but not according to the CONTACT header. > > see the wiki, i enable below paramter, > > > >From the wireshark trace, it shows FS received the REGISTERATION > message from remote port 20438, however, the INVITE message still sent > to the port(20347) inside "Contact", it looks the paramter > "NDLB-force-rport" did not effect, Could anyone help for this issue, > thanks a lot! > > Below is the registeration info from FS output (press F9), > > > Call-ID: 1721091027 at 10.33.48.203 > User: 13580358068 at 124.193.106.104 > Contact: "user" > Agent: > Status: Registered(UDP)(unknown) EXP(2011-11-08 14:38:02) > EXPSECS(152) > Host: freeswitch > IP: 61.144.248.77 > Port: 20438 > Auth-User: 13580358068 > Auth-Realm: 124.193.106.104 > MWI-Account: 13580358068 at 124.193.106.104 > > Cheers! > BR, > Charles > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/d68de19b/attachment.html From andy at fabulous4.co.uk Tue Nov 8 19:31:49 2011 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 8 Nov 2011 16:31:49 +0000 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> Message-ID: Many thanks for the quick response Michael, I'd rather not use the recorded stream if I can help it as I'm pushing the recorded stream out to a shoutcast server and recording it there. Is there a way to acheive what I need by connecting the calls instead? On 8 November 2011 16:18, Michael Collins wrote: > You can use mod_local_stream. Just set up a stream like the hold music on > but using the recorded message. Note: a local stream just plays over and > over again in a loop, so each call won't necessarily start at the beginning > of the message. > > -MC > > On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers wrote: > >> Hi,**** >> >> ** ** >> >> I would like to use freeswitch to host a one to many broadcast style >> conference call. The challenge I have is that there won?t always be any >> listeners so the conference application isn?t quite working for me.**** >> >> ** ** >> >> Essentially, the speaker dials in with something important to say, by >> default the system will simply record what they say as an archived message >> and then they hang up. This works fine but I would like to extend the >> functionality so that other people dialling in have the option to connect >> into this call and listen to the message live.**** >> >> ** ** >> >> Can you advise me on the best way of achieving this: conference, bridge, >> eavesdrop or something else?**** >> >> ** ** >> >> Many thanks**** >> >> Andy**** >> >> ** ** >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/641ee121/attachment.html From alec.taylor6 at gmail.com Tue Nov 8 19:34:18 2011 From: alec.taylor6 at gmail.com (Alec Taylor) Date: Wed, 9 Nov 2011 03:34:18 +1100 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> Message-ID: Sounds like you want to just use SHOUTcast directly, without using FreeSwitch. Will there be anyone participating in the conference call? On Wed, Nov 9, 2011 at 3:31 AM, Andy Ayers wrote: > Many thanks for the quick response Michael, I'd rather not use the recorded > stream if I can help it as I'm pushing the recorded stream out to a > shoutcast server and recording it there. Is there a way to acheive what I > need by connecting the calls instead? > > On 8 November 2011 16:18, Michael Collins wrote: >> >> You can use mod_local_stream. Just set up a stream like the hold music on >> but using the recorded message. Note: a local stream just plays over and >> over again in a loop, so each call won't necessarily start at the beginning >> of the message. >> -MC >> >> On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers wrote: >>> >>> Hi, >>> >>> >>> >>> I would like to use freeswitch to host a one to many broadcast style >>> conference call. The challenge I have is that there won?t always be any >>> listeners so the conference application isn?t quite working for me. >>> >>> >>> >>> Essentially, the speaker dials in with something important to say, by >>> default the system will simply record what they say as an archived message >>> and then they hang up. This works fine but I would like to extend the >>> functionality so that other people dialling in have the option to connect >>> into this call and listen to the message live. >>> >>> >>> >>> Can you advise me on the best way of achieving this: conference, bridge, >>> eavesdrop or something else? >>> >>> >>> >>> Many thanks >>> >>> Andy >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From andy at fabulous4.co.uk Tue Nov 8 19:37:36 2011 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 8 Nov 2011 16:37:36 +0000 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> Message-ID: Hi Alec, Yes sometimes, hence the challenge. Andy On 8 November 2011 16:34, Alec Taylor wrote: > Sounds like you want to just use SHOUTcast directly, without using > FreeSwitch. > > Will there be anyone participating in the conference call? > > On Wed, Nov 9, 2011 at 3:31 AM, Andy Ayers wrote: > > Many thanks for the quick response Michael, I'd rather not use the > recorded > > stream if I can help it as I'm pushing the recorded stream out to a > > shoutcast server and recording it there. Is there a way to acheive what I > > need by connecting the calls instead? > > > > On 8 November 2011 16:18, Michael Collins wrote: > >> > >> You can use mod_local_stream. Just set up a stream like the hold music > on > >> but using the recorded message. Note: a local stream just plays over and > >> over again in a loop, so each call won't necessarily start at the > beginning > >> of the message. > >> -MC > >> > >> On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers > wrote: > >>> > >>> Hi, > >>> > >>> > >>> > >>> I would like to use freeswitch to host a one to many broadcast style > >>> conference call. The challenge I have is that there won?t always be any > >>> listeners so the conference application isn?t quite working for me. > >>> > >>> > >>> > >>> Essentially, the speaker dials in with something important to say, by > >>> default the system will simply record what they say as an archived > message > >>> and then they hang up. This works fine but I would like to extend the > >>> functionality so that other people dialling in have the option to > connect > >>> into this call and listen to the message live. > >>> > >>> > >>> > >>> Can you advise me on the best way of achieving this: conference, > bridge, > >>> eavesdrop or something else? > >>> > >>> > >>> > >>> Many thanks > >>> > >>> Andy > >>> > >>> > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/f0a20f44/attachment.html From infos at madovsky.org Tue Nov 8 19:40:57 2011 From: infos at madovsky.org (Madovsky) Date: Tue, 8 Nov 2011 11:40:57 -0500 Subject: [Freeswitch-users] transcoding channel var Message-ID: <7CEB663B7C60453FBA8DC5FAF544FDE4@e1705> Hi Folks, is there any channel var like transcoding=true to know if the channel is transcoding or not ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/062a53c8/attachment.html From avi at avimarcus.net Tue Nov 8 19:29:13 2011 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 8 Nov 2011 18:29:13 +0200 Subject: [Freeswitch-users] Help with home setup In-Reply-To: <4EB9288F.6050705@gmail.com> References: <4EB9288F.6050705@gmail.com> Message-ID: Most basic debug: open the fs_cli (/usr/local/freeswitch/bin/fs_cli) and see if you see the register attempts. If not, then there's an IP or firewall issue (perhaps your FS is listening only on a public IP or the like) Or, it will tell you the authentication error. -Avi On Tue, Nov 8, 2011 at 3:03 PM, Jacob Smith wrote: > I have been trying to get this working for the last week and would > really appreciate your help. > > I have a pfSense 2.0 server with Freeswitch and FusionPBX installed. I > followed the instructions at > http://wiki.fusionpbx.com/index.php?title=PfSense_Install and can access > the web GUI. > > My goal is to use a Gigaset C610A-IP connected to my Google Voice > account. However, my first failure seems to be the most basic: through > FusionPBX, I created a user and extension to use with the phone but I > can not register it with FreeSwitch. So, how do I get the phone to work > so I can test that the basic functions of FreeSwitch are working and > move on? > > 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/20111108/75029643/attachment.html From msc at freeswitch.org Tue Nov 8 20:51:58 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 09:51:58 -0800 Subject: [Freeswitch-users] Knowing whether a fax has been sent or not In-Reply-To: References: Message-ID: Also, don't forget about the as-yet undocumented "execute_on" vars: execute_on_fax_detect execute_on_fax_failure execute_on_fax_result execute_on_fax_success Apologies for not getting these on the wiki just yet. If anyone can help w/ documenting these then I would be eternally grateful... -MC On Tue, Nov 8, 2011 at 3:05 AM, Yehavi Bourvine wrote: > Hello, > > I am building a system to send faxes with txfax() application. How can I > get the result and know whether the fax has been sent or not? ESL, or is > there some callback which I enable? > > 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/20111108/11bf7ba8/attachment.html From jmoran at secureachsystems.com Tue Nov 8 20:55:48 2011 From: jmoran at secureachsystems.com (Jason Moran) Date: Tue, 8 Nov 2011 12:55:48 -0500 Subject: [Freeswitch-users] Hangup Causes References: <361E98F99D3CC3439EED59BC1924ED69547635@SERVER2003.SecuReachSystems.local> Message-ID: <361E98F99D3CC3439EED59BC1924ED69589FBF@SERVER2003.SecuReachSystems.local> Are there any linux packages that will look within subdirectories and automatically delete files/directories either after age or file size thresholds are met? (I installed pcapsipdump, but of course I worry about how much hard drive space will be used). -Jason From: Michael Collins [mailto:msc at freeswitch.org] Sent: Friday, October 28, 2011 2:10 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Hangup Causes Although not easy on a production system, your best bet is to catch the behavior in the act. Usually this means that you have to rely upon humans to report the symptoms to you in a timely and accurate manner. One way you can help track these down is to use something like pcapsipdump and do a ring buffer that covers at least several days. Then, when you hear about a symptom you can go back and look at the corresponding pcap file which will have both signaling and audio. From there you can hopefully trace the root of the problem. -MC http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump On Fri, Oct 28, 2011 at 9:19 AM, Jason Moran wrote: I have some questions about the types of statuses we are getting (Yes, I'm already familiar with http://wiki.freeswitch.org/wiki/Hangup_causes ) We just deployed FS to production a couple of days ago and here is a sample of our Hangup Causes: CALL_REJECTED 1 NO_ANSWER 738 NO_USER_RESPONSE 175 NORMAL_CLEARING 1667 (Expected that this is only Early Hangups) NORMAL_TEMPORARY_FAILURE 4 NORMAL_TEMPORARY_FAILURE (Rang) 37 (Rang means this status happened more than 5 seconds after the session was started) Complete 224 (Complete is our own status instead of NORMAL_CLEARING, meaning WE hung up because we finished the entire message). ORIGINATOR_CANCEL (Rang) 3 RECOVERY_ON_TIMER_EXPIRE (Rang) 9 UNALLOCATED_NUMBER 2 USER_BUSY 35 My questions: 1. NORMAL_TEMPORARY_FAILURE happens a lot, and often returns 30 or more seconds after starting. Could this be a "silent call" (etc)? We have gotten 3-4 complaints of silent calls and I'm hoping to find the cause. 2. Is there anything here (or ideas elsewhere?) to explain why a call will be silent once in a while? 3. Is this spread of causes normal or does it seem strange 4. We've had a few people note that the message on their voicemail is over before they can hear it - even though we play the message twice (to make short messages longer). Does anybody know of a solution for this? Thanks! -Jason FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/94d89936/attachment-0001.html From vallimamod.abdullah at imtelecom.fr Tue Nov 8 21:06:48 2011 From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH) Date: Tue, 8 Nov 2011 19:06:48 +0100 Subject: [Freeswitch-users] Hangup Causes In-Reply-To: <361E98F99D3CC3439EED59BC1924ED69589FBF@SERVER2003.SecuReachSystems.local> References: <361E98F99D3CC3439EED59BC1924ED69547635@SERVER2003.SecuReachSystems.local> <361E98F99D3CC3439EED59BC1924ED69589FBF@SERVER2003.SecuReachSystems.local> Message-ID: <85612334-3105-4E4C-96AD-5C2724F1B58C@imtelecom.fr> On Nov 8, 2011, at 6:55 PM, Jason Moran wrote: > Are there any linux packages that will look within subdirectories and automatically delete files/directories either after age or file size thresholds are met? (I installed pcapsipdump, but of course I worry about how much hard drive space will be used). You can use find. Have a look on the TEST sections of the manpage (especially -ctime and -size). Regards, - vma . From msc at freeswitch.org Tue Nov 8 21:09:19 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 10:09:19 -0800 Subject: [Freeswitch-users] Hangup Causes In-Reply-To: <361E98F99D3CC3439EED59BC1924ED69589FBF@SERVER2003.SecuReachSystems.local> References: <361E98F99D3CC3439EED59BC1924ED69547635@SERVER2003.SecuReachSystems.local> <361E98F99D3CC3439EED59BC1924ED69589FBF@SERVER2003.SecuReachSystems.local> Message-ID: You can also use tcpdump to create a ring buffer and then use pcapsipdump to parse the pcaps that are produced by tcpdump. -MC On Tue, Nov 8, 2011 at 9:55 AM, Jason Moran wrote: > Are there any linux packages that will look within subdirectories and > automatically delete files/directories either after age or file size > thresholds are met? (I installed pcapsipdump, but of course I worry about > how much hard drive space will be used).**** > > ** ** > > -Jason**** > > ** ** > > *From:* Michael Collins [mailto:msc at freeswitch.org] > *Sent:* Friday, October 28, 2011 2:10 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Hangup Causes**** > > ** ** > > Although not easy on a production system, your best bet is to catch the > behavior in the act. Usually this means that you have to rely upon humans > to report the symptoms to you in a timely and accurate manner. **** > > ** ** > > One way you can help track these down is to use something like pcapsipdump > and do a ring buffer that covers at least several days. Then, when you hear > about a symptom you can go back and look at the corresponding pcap file > which will have both signaling and audio. From there you can hopefully > trace the root of the problem.**** > > ** ** > > -MC**** > > ** ** > > http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump**** > > ** ** > > On Fri, Oct 28, 2011 at 9:19 AM, Jason Moran > wrote:**** > > I have some questions about the types of statuses we are getting (Yes, I?m > already familiar with http://wiki.freeswitch.org/wiki/Hangup_causes )**** > > **** > > We just deployed FS to production a couple of days ago and here is a > sample of our Hangup Causes:**** > > CALL_REJECTED 1**** > > NO_ANSWER 738**** > > NO_USER_RESPONSE 175**** > > NORMAL_CLEARING 1667 (Expected that this is only Early Hangups)**** > > NORMAL_TEMPORARY_FAILURE 4**** > > NORMAL_TEMPORARY_FAILURE (Rang) 37 (Rang means this status > happened more than 5 seconds after the session was started)**** > > Complete 224 (Complete is our own status instead of > NORMAL_CLEARING, meaning WE hung up because we finished the entire message). > **** > > ORIGINATOR_CANCEL (Rang) 3**** > > RECOVERY_ON_TIMER_EXPIRE (Rang) 9 **** > > UNALLOCATED_NUMBER 2**** > > USER_BUSY 35**** > > **** > > My questions:**** > > 1. NORMAL_TEMPORARY_FAILURE happens a lot, and often returns 30 or > more seconds after starting. Could this be a ?silent call? (etc)? We have > gotten 3-4 complaints of silent calls and I?m hoping to find the cause.*** > * > > 2. Is there anything here (or ideas elsewhere?) to explain why a > call will be silent once in a while?**** > > 3. Is this spread of causes normal or does it seem strange**** > > 4. We?ve had a few people note that the message on their voicemail > is over before they can hear it - even though we play the message twice (to > make short messages longer). Does anybody know of a solution for this?**** > > **** > > Thanks! -Jason**** > > **** > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/4555177f/attachment.html From brian at freeswitch.org Tue Nov 8 21:11:44 2011 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Nov 2011 12:11:44 -0600 Subject: [Freeswitch-users] paramter "NDLB-force-rport" did not effect In-Reply-To: References: Message-ID: What is the user agent? /b On Nov 8, 2011, at 1:30 AM, fieldpeak wrote: > Dear Friends, > > i met a problem that call a registered user behind NAT, i need FS send > INVITE message to the port of the NAT device from which the registered > user sent out REGISTRATION but not according to the CONTACT header. > > see the wiki, i enable below paramter, > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/5cc35f6e/attachment.html From msc at freeswitch.org Tue Nov 8 21:14:59 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 10:14:59 -0800 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> Message-ID: What are the conditions under which some will participate? Will there be a defined subset of callers who can participate and another subset who will be listen only? I don't see how you can avoid using a conference if more than one audio source will be used and multiple other endpoints will be listening. -MC On Tue, Nov 8, 2011 at 8:37 AM, Andy Ayers wrote: > Hi Alec, > > Yes sometimes, hence the challenge. > > Andy > > > On 8 November 2011 16:34, Alec Taylor wrote: > >> Sounds like you want to just use SHOUTcast directly, without using >> FreeSwitch. >> >> Will there be anyone participating in the conference call? >> >> On Wed, Nov 9, 2011 at 3:31 AM, Andy Ayers wrote: >> > Many thanks for the quick response Michael, I'd rather not use the >> recorded >> > stream if I can help it as I'm pushing the recorded stream out to a >> > shoutcast server and recording it there. Is there a way to acheive what >> I >> > need by connecting the calls instead? >> > >> > On 8 November 2011 16:18, Michael Collins wrote: >> >> >> >> You can use mod_local_stream. Just set up a stream like the hold music >> on >> >> but using the recorded message. Note: a local stream just plays over >> and >> >> over again in a loop, so each call won't necessarily start at the >> beginning >> >> of the message. >> >> -MC >> >> >> >> On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers >> wrote: >> >>> >> >>> Hi, >> >>> >> >>> >> >>> >> >>> I would like to use freeswitch to host a one to many broadcast style >> >>> conference call. The challenge I have is that there won?t always be >> any >> >>> listeners so the conference application isn?t quite working for me. >> >>> >> >>> >> >>> >> >>> Essentially, the speaker dials in with something important to say, by >> >>> default the system will simply record what they say as an archived >> message >> >>> and then they hang up. This works fine but I would like to extend the >> >>> functionality so that other people dialling in have the option to >> connect >> >>> into this call and listen to the message live. >> >>> >> >>> >> >>> >> >>> Can you advise me on the best way of achieving this: conference, >> bridge, >> >>> eavesdrop or something else? >> >>> >> >>> >> >>> >> >>> Many thanks >> >>> >> >>> Andy >> >>> >> >>> >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/948470ea/attachment-0001.html From brian at freeswitch.org Tue Nov 8 21:16:45 2011 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Nov 2011 12:16:45 -0600 Subject: [Freeswitch-users] Knowing whether a fax has been sent or not In-Reply-To: References: Message-ID: <29467D10-8491-4CC4-8074-D4C658717613@freeswitch.org> DOH! These are nice... I totally missed tony adding them.. shame on me! /b On Nov 8, 2011, at 11:51 AM, Michael Collins wrote: > Also, don't forget about the as-yet undocumented "execute_on" vars: > > execute_on_fax_detect > execute_on_fax_failure > execute_on_fax_result > execute_on_fax_success > > Apologies for not getting these on the wiki just yet. If anyone can help w/ > documenting these then I would be eternally grateful... > > -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/ecae05a9/attachment.html From andy at fabulous4.co.uk Tue Nov 8 21:39:42 2011 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 8 Nov 2011 18:39:42 -0000 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> Message-ID: <021f01cc9e45$c8a859b0$59f90d10$@fabulous4.co.uk> Hi Michael, Apologies for the confusion there will only ever be a single audio source, is the conference still the best way or can the other callers eavesdrop somehow on the main call? Many thanks for your help Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 08 November 2011 18:15 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best way to create a one to many 'broadcast' What are the conditions under which some will participate? Will there be a defined subset of callers who can participate and another subset who will be listen only? I don't see how you can avoid using a conference if more than one audio source will be used and multiple other endpoints will be listening. -MC On Tue, Nov 8, 2011 at 8:37 AM, Andy Ayers wrote: Hi Alec, Yes sometimes, hence the challenge. Andy On 8 November 2011 16:34, Alec Taylor wrote: Sounds like you want to just use SHOUTcast directly, without using FreeSwitch. Will there be anyone participating in the conference call? On Wed, Nov 9, 2011 at 3:31 AM, Andy Ayers wrote: > Many thanks for the quick response Michael, I'd rather not use the recorded > stream if I can help it as I'm pushing the recorded stream out to a > shoutcast server and recording it there. Is there a way to acheive what I > need by connecting the calls instead? > > On 8 November 2011 16:18, Michael Collins wrote: >> >> You can use mod_local_stream. Just set up a stream like the hold music on >> but using the recorded message. Note: a local stream just plays over and >> over again in a loop, so each call won't necessarily start at the beginning >> of the message. >> -MC >> >> On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers wrote: >>> >>> Hi, >>> >>> >>> >>> I would like to use freeswitch to host a one to many broadcast style >>> conference call. The challenge I have is that there won't always be any >>> listeners so the conference application isn't quite working for me. >>> >>> >>> >>> Essentially, the speaker dials in with something important to say, by >>> default the system will simply record what they say as an archived message >>> and then they hang up. This works fine but I would like to extend the >>> functionality so that other people dialling in have the option to connect >>> into this call and listen to the message live. >>> >>> >>> >>> Can you advise me on the best way of achieving this: conference, bridge, >>> eavesdrop or something else? >>> >>> >>> >>> Many thanks >>> >>> Andy >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/d00fc41a/attachment.html From hynek.cihlar at gmail.com Tue Nov 8 21:42:59 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Tue, 8 Nov 2011 19:42:59 +0100 Subject: [Freeswitch-users] Registration status timeouts Message-ID: <7255624560545829990@unknownmsgid> Is it normal to have sip devices timeout in the sofia status (or show registrations). and still be reachable with originate? It looks like that the devices in question have longer timeouts than the timeout set on the sip profile. Can a sip device be forced a re-register timeout? Should it be? Is it ok to have timeouts on the profile? What's the correct reg timeout value on the sip profile? Sent from my mobile device From sunwood360 at gmail.com Tue Nov 8 21:53:04 2011 From: sunwood360 at gmail.com (envelopes envelopes) Date: Tue, 8 Nov 2011 10:53:04 -0800 Subject: [Freeswitch-users] Fs event socket inbound or outbound? In-Reply-To: References: Message-ID: Both modes can send command and receive events. Which mode is more scalable? And pros vs cons of each mode. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/09d8eb0c/attachment.html From msc at freeswitch.org Tue Nov 8 21:53:17 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 10:53:17 -0800 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: <021f01cc9e45$c8a859b0$59f90d10$@fabulous4.co.uk> References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> <021f01cc9e45$c8a859b0$59f90d10$@fabulous4.co.uk> Message-ID: What is the "main call" in this scenario? Sorry, I'm more confused than ever. Perhaps you could restate the question. I also respond well to pictures. :P -MC On Tue, Nov 8, 2011 at 10:39 AM, Andy Ayers wrote: > Hi Michael,**** > > ** ** > > Apologies for the confusion there will only ever be a single audio source, > is the conference still the best way or can the other callers eavesdrop > somehow on the main call?**** > > ** ** > > Many thanks for your help**** > > Andy**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 08 November 2011 18:15 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best way to create a one to many > 'broadcast'**** > > ** ** > > What are the conditions under which some will participate? Will there be a > defined subset of callers who can participate and another subset who will > be listen only? I don't see how you can avoid using a conference if more > than one audio source will be used and multiple other endpoints will be > listening.**** > > ** ** > > -MC**** > > On Tue, Nov 8, 2011 at 8:37 AM, Andy Ayers wrote:** > ** > > Hi Alec, > > Yes sometimes, hence the challenge. > > Andy**** > > ** ** > > On 8 November 2011 16:34, Alec Taylor wrote:**** > > Sounds like you want to just use SHOUTcast directly, without using > FreeSwitch. > > Will there be anyone participating in the conference call? > > On Wed, Nov 9, 2011 at 3:31 AM, Andy Ayers wrote: > > Many thanks for the quick response Michael, I'd rather not use the > recorded > > stream if I can help it as I'm pushing the recorded stream out to a > > shoutcast server and recording it there. Is there a way to acheive what I > > need by connecting the calls instead? > > > > On 8 November 2011 16:18, Michael Collins wrote: > >> > >> You can use mod_local_stream. Just set up a stream like the hold music > on > >> but using the recorded message. Note: a local stream just plays over and > >> over again in a loop, so each call won't necessarily start at the > beginning > >> of the message. > >> -MC > >> > >> On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers > wrote: > >>> > >>> Hi, > >>> > >>> > >>> > >>> I would like to use freeswitch to host a one to many broadcast style > >>> conference call. The challenge I have is that there won?t always be any > >>> listeners so the conference application isn?t quite working for me. > >>> > >>> > >>> > >>> Essentially, the speaker dials in with something important to say, by > >>> default the system will simply record what they say as an archived > message > >>> and then they hang up. This works fine but I would like to extend the > >>> functionality so that other people dialling in have the option to > connect > >>> into this call and listen to the message live. > >>> > >>> > >>> > >>> Can you advise me on the best way of achieving this: conference, > bridge, > >>> eavesdrop or something else? > >>> > >>> > >>> > >>> Many thanks > >>> > >>> Andy > >>> > >>> > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/56935352/attachment-0001.html From msc at freeswitch.org Tue Nov 8 21:59:04 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 10:59:04 -0800 Subject: [Freeswitch-users] Fs event socket inbound or outbound? In-Reply-To: References: Message-ID: The modes are both "event socket" and therefore quite scalable. The primary difference is in how the connection gets started. With outbound event socket, there is always a call involved. A call comes in to FreeSWITCH and somewhere in the dialplan there is a call to the socket application. At that point FreeSWITCH makes an "outbound" connection to your application which is listening for connections on the IP:Port in question. With inbound event socket, FreeSWITCH is the one sitting there waiting for your application to connect, i.e. an "inbound" socket connection from your program. Once your program is connected it can do pretty much whatever you want: show channels to see what's going on, originate to create new channels, fifo list, conference list, etc. etc. BTW, fs_cli is a great example of an inbound event socket application. fs_cli literally connects to FS via event socket. Hope this helps. -MC On Tue, Nov 8, 2011 at 10:53 AM, envelopes envelopes wrote: > Both modes can send command and receive events. Which mode is more > scalable? > And pros vs cons of each mode. > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/4951aa7a/attachment.html From ljjimenez at gmail.com Tue Nov 8 22:10:34 2011 From: ljjimenez at gmail.com (Luis Jimenez) Date: Tue, 8 Nov 2011 19:10:34 +0000 Subject: [Freeswitch-users] Fs event socket inbound or outbound? In-Reply-To: References: Message-ID: <1460947918-1320779434-cardhu_decombobulator_blackberry.rim.net-291110320-@b13.c27.bise6.blackberry> What is the best way to implement such server, using threads or fork? Thanks in advance Luis Jimenez -----Original Message----- From: Michael Collins Sender: freeswitch-users-bounces at lists.freeswitch.org Date: Tue, 8 Nov 2011 10:59:04 To: FreeSWITCH Users Help Reply-To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fs event socket inbound or outbound? FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From spencer at 5ninesolutions.com Tue Nov 8 22:18:46 2011 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Tue, 8 Nov 2011 11:18:46 -0800 Subject: [Freeswitch-users] Cisco SPA DTMF Settings Message-ID: <847DED73-FB60-42E7-B862-F51ECE19FDD5@5ninesolutions.com> Hello all, I have a question regarding the provisioning of Cisco SPA-500 series phones. This also pertains to Linksys SPA-900s as well. The phones have an option to set the marker bit on all rfc2833 packets, not just the first one. Both options seem to work with FreeSWITCH. Which would be the more correct setting? Thanks, Spencer From sunwood360 at gmail.com Tue Nov 8 22:34:54 2011 From: sunwood360 at gmail.com (envelopes envelopes) Date: Tue, 8 Nov 2011 11:34:54 -0800 Subject: [Freeswitch-users] Fs event socket inbound or outbound? In-Reply-To: References: Message-ID: Michael, great detailed response. If I need to monitor FS heartbeat, it seems I could only do it with inbound mode since no call is associated yet. On Nov 8, 2011 11:02 AM, "Michael Collins" wrote: > The modes are both "event socket" and therefore quite scalable. The > primary difference is in how the connection gets started. > > With outbound event socket, there is always a call involved. A call comes > in to FreeSWITCH and somewhere in the dialplan there is a call to the > socket application. At that point FreeSWITCH makes an "outbound" connection > to your application which is listening for connections on the IP:Port in > question. > > With inbound event socket, FreeSWITCH is the one sitting there waiting for > your application to connect, i.e. an "inbound" socket connection from your > program. Once your program is connected it can do pretty much whatever you > want: show channels to see what's going on, originate to create new > channels, fifo list, conference list, etc. etc. BTW, fs_cli is a great > example of an inbound event socket application. fs_cli literally connects > to FS via event socket. > > Hope this helps. > > -MC > > On Tue, Nov 8, 2011 at 10:53 AM, envelopes envelopes > wrote: > >> Both modes can send command and receive events. Which mode is more >> scalable? >> And pros vs cons of each mode. >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/42baf343/attachment.html From msc at freeswitch.org Tue Nov 8 23:46:31 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 12:46:31 -0800 Subject: [Freeswitch-users] Fs event socket inbound or outbound? In-Reply-To: References: Message-ID: On Tue, Nov 8, 2011 at 11:34 AM, envelopes envelopes wrote: > Michael, great detailed response. > > If I need to monitor FS heartbeat, it seems I could only do it with > inbound mode since no call is associated yet. > Correct. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/6b521929/attachment.html From acrow at integrafin.co.uk Wed Nov 9 00:01:22 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Tue, 08 Nov 2011 21:01:22 +0000 Subject: [Freeswitch-users] mod_opal Message-ID: <4EB998A2.8030206@integrafin.co.uk> Hi all, I'm having some issues getting mod_opal installed - my intention is to use its IAX2 provision with iaxmodem and hylafax to get faxing through hylafax via a SIP trunk to a Mitel 3300 acting as an ISDN gateway. It just seems it does not want to compile - I'm using the Sirius release of PTLIB and OPAL, but it did also have a go with Lalande. So my first question would be - is mod_opal maintained any more? Or is there and equivalent "sipmodem" project that would replace iaxmodem? Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From andy at fabulous4.co.uk Wed Nov 9 00:45:55 2011 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 8 Nov 2011 21:45:55 -0000 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> <021f01cc9e45$c8a859b0$59f90d10$@fabulous4.co.uk> Message-ID: <026e01cc9e5f$cc928d60$65b7a820$@fabulous4.co.uk> Hi Michael, Here's the scenario: The 'speaker' calls in to the switch and is invited to begin speaking after the beep just as if they were recording an answering message and the audio is recorded (in our case via a shoutcast server). This bit is working. These messages can last several minutes and in some cases hours so I would like to add to this the ability for 'listeners' to call in to the switch and be patched in to the above call to listen only to the message live in real time. The listeners will not be allowed to interact with the call in any way. Because this is optional however there won't always be listeners, sometimes it will be a simple dial in and record for the speaker. Hope I'm making more sense now. Many thanks for persevering Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 08 November 2011 18:53 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best way to create a one to many 'broadcast' What is the "main call" in this scenario? Sorry, I'm more confused than ever. Perhaps you could restate the question. I also respond well to pictures. :P -MC On Tue, Nov 8, 2011 at 10:39 AM, Andy Ayers wrote: Hi Michael, Apologies for the confusion there will only ever be a single audio source, is the conference still the best way or can the other callers eavesdrop somehow on the main call? Many thanks for your help Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 08 November 2011 18:15 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best way to create a one to many 'broadcast' What are the conditions under which some will participate? Will there be a defined subset of callers who can participate and another subset who will be listen only? I don't see how you can avoid using a conference if more than one audio source will be used and multiple other endpoints will be listening. -MC On Tue, Nov 8, 2011 at 8:37 AM, Andy Ayers wrote: Hi Alec, Yes sometimes, hence the challenge. Andy On 8 November 2011 16:34, Alec Taylor wrote: Sounds like you want to just use SHOUTcast directly, without using FreeSwitch. Will there be anyone participating in the conference call? On Wed, Nov 9, 2011 at 3:31 AM, Andy Ayers wrote: > Many thanks for the quick response Michael, I'd rather not use the recorded > stream if I can help it as I'm pushing the recorded stream out to a > shoutcast server and recording it there. Is there a way to acheive what I > need by connecting the calls instead? > > On 8 November 2011 16:18, Michael Collins wrote: >> >> You can use mod_local_stream. Just set up a stream like the hold music on >> but using the recorded message. Note: a local stream just plays over and >> over again in a loop, so each call won't necessarily start at the beginning >> of the message. >> -MC >> >> On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers wrote: >>> >>> Hi, >>> >>> >>> >>> I would like to use freeswitch to host a one to many broadcast style >>> conference call. The challenge I have is that there won't always be any >>> listeners so the conference application isn't quite working for me. >>> >>> >>> >>> Essentially, the speaker dials in with something important to say, by >>> default the system will simply record what they say as an archived message >>> and then they hang up. This works fine but I would like to extend the >>> functionality so that other people dialling in have the option to connect >>> into this call and listen to the message live. >>> >>> >>> >>> Can you advise me on the best way of achieving this: conference, bridge, >>> eavesdrop or something else? >>> >>> >>> >>> Many thanks >>> >>> Andy >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/4fbdb194/attachment-0001.html From Ryan at ocens.com Wed Nov 9 01:31:06 2011 From: Ryan at ocens.com (Ryan Watkins) Date: Tue, 8 Nov 2011 22:31:06 +0000 Subject: [Freeswitch-users] FreeSWITCH Module Question Message-ID: <44E5C0A9D48A3246966A4AE04692014D10270A2F@CH1PRD0604MB109.namprd06.prod.outlook.com> Hello FreeSWITCH folks! First of all, thank you for all your work on this wonderful product... it truly is a great utility. I've been doing some studying of the FreeSWITCH 1.0.6 book, and have been looking at the wiki here... but I'm looking for a specific function to see if it's available... The feature I'm looking for is kind of a combination of Group Calling and Conference Calling.... What we would like to implement is a single dial number that when called, calls and creates a conference (as opposed to the "first answer wins" function of Group Calling) Is there a module available for FreeSWITCH that would provide this function? Thanks! Ryan Watkins Networking & Customer Support OCENS 19655 1st AVE S. #203 Seattle, WA 98148 Satellite Systems and Services: Iridium, Inmarsat, Globalstar, KVH ________________________________________________________ www.ocens.com | support.ocens.com Office: (206) 878-8270 | Cell: (360) 521-7334 | Fax: (206) 878-8314 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/2f36aed2/attachment.html From msc at freeswitch.org Wed Nov 9 01:39:26 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 14:39:26 -0800 Subject: [Freeswitch-users] FreeSWITCH Module Question In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D10270A2F@CH1PRD0604MB109.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D10270A2F@CH1PRD0604MB109.namprd06.prod.outlook.com> Message-ID: You probably just want this: http://wiki.freeswitch.org/wiki/Conference_set_auto_outcall Let us know if that serves your purpose... -MC On Tue, Nov 8, 2011 at 2:31 PM, Ryan Watkins wrote: > Hello FreeSWITCH folks!**** > > ** ** > > First of all, thank you for all your work on this wonderful product? it > truly is a great utility.**** > > ** ** > > I?ve been doing some studying of the FreeSWITCH 1.0.6 book, and have been > looking at the wiki here? but I?m looking for a specific function to see if > it?s available?**** > > ** ** > > The feature I?m looking for is kind of a combination of Group Calling and > Conference Calling?. What we would like to implement is a single dial > number that when called, calls and creates a conference (as opposed to the > ?first answer wins? function of Group Calling) Is there a module available > for FreeSWITCH that would provide this function?**** > > ** ** > > Thanks!**** > > ** ** > > *Ryan Watkins* > > Networking & Customer Support**** > > ** ** > > *OCENS* > > 19655 1st AVE S. #203**** > > Seattle, WA 98148**** > > ** ** > > Satellite Systems and Services: Iridium, Inmarsat, Globalstar, KVH**** > > ________________________________________________________**** > > www.ocens.com | support.ocens.com**** > > *Office:* (206) 878-8270 | *Cell:* (360) 521-7334 | *Fax:* (206) 878-8314* > *** > > ** ** > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/f10865ae/attachment.html From freeswitch-list at puzzled.xs4all.nl Wed Nov 9 02:48:56 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 09 Nov 2011 00:48:56 +0100 Subject: [Freeswitch-users] mod_opal In-Reply-To: <4EB998A2.8030206@integrafin.co.uk> References: <4EB998A2.8030206@integrafin.co.uk> Message-ID: <4EB9BFE8.3050701@puzzled.xs4all.nl> On 11/08/2011 10:01 PM, Alex Crow wrote: > Hi all, > > I'm having some issues getting mod_opal installed - my intention is to > use its IAX2 provision with iaxmodem and hylafax to get faxing through > hylafax via a SIP trunk to a Mitel 3300 acting as an ISDN gateway. I wasn't aware that IAX2 was in working order in opal/mod_opal. Have you verified this? > It just seems it does not want to compile - I'm using the Sirius release > of PTLIB and OPAL, but it did also have a go with Lalande. > > So my first question would be - is mod_opal maintained any more? Or is > there and equivalent "sipmodem" project that would replace iaxmodem? A few months back on a CentOS 5.6 x86 box I built ptlib 2.6.8, opal 3.6.8 and mod_opal compiled fine. Which distro are you using and what are the errors you are seeing? Regards, Patrick From lloyd.aloysius at gmail.com Wed Nov 9 03:05:53 2011 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 8 Nov 2011 19:05:53 -0500 Subject: [Freeswitch-users] Voice mail migration Message-ID: HI, I am in the process of moving my current freeswitch server to a new server. What is the best way to migrate the VoiceMail from old to new server? Any help is appreciated. Thanks and regards, Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/8e8c5f04/attachment.html From msc at freeswitch.org Wed Nov 9 03:32:48 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 16:32:48 -0800 Subject: [Freeswitch-users] Voice mail migration In-Reply-To: References: Message-ID: Is your domain the same? Basically you have the voicemail.db file and you have /usr/local/freeswitch/storage/voicemail/default/xxxx where xxx is the domain. Shouldn't be too hard to copy everything over. -MC On Tue, Nov 8, 2011 at 4:05 PM, Lloyd Aloysius wrote: > HI, > > I am in the process of moving my current freeswitch server to a new > server. What is the best way to migrate the VoiceMail from old to new > server? > > Any help is appreciated. > > Thanks and regards, > > Lloyd > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/7fe5c1eb/attachment.html From lloyd.aloysius at gmail.com Wed Nov 9 03:39:49 2011 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 8 Nov 2011 19:39:49 -0500 Subject: [Freeswitch-users] Voice mail migration In-Reply-To: References: Message-ID: Micheal, Thank you for the reply. Yes same domain . But New server have a new version of FreeSWITCH. Do I need to maintain the same version FreeSWITCH for my new server. Thanks and regards, Lloyd On Tue, Nov 8, 2011 at 7:32 PM, Michael Collins wrote: > Is your domain the same? Basically you have the voicemail.db file and you > have /usr/local/freeswitch/storage/voicemail/default/xxxx where xxx is the > domain. Shouldn't be too hard to copy everything over. > > -MC > > On Tue, Nov 8, 2011 at 4:05 PM, Lloyd Aloysius wrote: > >> HI, >> >> I am in the process of moving my current freeswitch server to a new >> server. What is the best way to migrate the VoiceMail from old to new >> server? >> >> Any help is appreciated. >> >> Thanks and regards, >> >> Lloyd >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/0e115314/attachment-0001.html From msc at freeswitch.org Wed Nov 9 03:43:26 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Nov 2011 16:43:26 -0800 Subject: [Freeswitch-users] Voice mail migration In-Reply-To: References: Message-ID: unless the previous server is really old, I doubt there will be any trouble migrating. I would run a test and see how it goes. -MC On Tue, Nov 8, 2011 at 4:39 PM, Lloyd Aloysius wrote: > Micheal, > > Thank you for the reply. Yes same domain . But New server have a new > version of FreeSWITCH. > > Do I need to maintain the same version FreeSWITCH for my new server. > > > Thanks and regards, > > Lloyd > > > > On Tue, Nov 8, 2011 at 7:32 PM, Michael Collins wrote: > >> Is your domain the same? Basically you have the voicemail.db file and you >> have /usr/local/freeswitch/storage/voicemail/default/xxxx where xxx is the >> domain. Shouldn't be too hard to copy everything over. >> >> -MC >> >> On Tue, Nov 8, 2011 at 4:05 PM, Lloyd Aloysius wrote: >> >>> HI, >>> >>> I am in the process of moving my current freeswitch server to a new >>> server. What is the best way to migrate the VoiceMail from old to new >>> server? >>> >>> Any help is appreciated. >>> >>> Thanks and regards, >>> >>> Lloyd >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/bd4ae1e4/attachment.html From lloyd.aloysius at sunteltech.ca Wed Nov 9 04:21:44 2011 From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius) Date: Tue, 8 Nov 2011 20:21:44 -0500 Subject: [Freeswitch-users] Voice mail migration In-Reply-To: References: Message-ID: Thank you for the reply. Old server running FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) New Server ... Planning to run from the git. Thanks and regards, Lloyd On Tue, Nov 8, 2011 at 7:43 PM, Michael Collins wrote: > unless the previous server is really old, I doubt there will be any > trouble migrating. I would run a test and see how it goes. > > -MC > > > On Tue, Nov 8, 2011 at 4:39 PM, Lloyd Aloysius wrote: > >> Micheal, >> >> Thank you for the reply. Yes same domain . But New server have a new >> version of FreeSWITCH. >> >> Do I need to maintain the same version FreeSWITCH for my new server. >> >> >> Thanks and regards, >> >> Lloyd >> >> >> >> On Tue, Nov 8, 2011 at 7:32 PM, Michael Collins wrote: >> >>> Is your domain the same? Basically you have the voicemail.db file and >>> you have /usr/local/freeswitch/storage/voicemail/default/xxxx where xxx is >>> the domain. Shouldn't be too hard to copy everything over. >>> >>> -MC >>> >>> On Tue, Nov 8, 2011 at 4:05 PM, Lloyd Aloysius >> > wrote: >>> >>>> HI, >>>> >>>> I am in the process of moving my current freeswitch server to a new >>>> server. What is the best way to migrate the VoiceMail from old to new >>>> server? >>>> >>>> Any help is appreciated. >>>> >>>> Thanks and regards, >>>> >>>> Lloyd >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/b348cf5e/attachment.html From curriegrad2004 at gmail.com Wed Nov 9 04:52:45 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 8 Nov 2011 17:52:45 -0800 Subject: [Freeswitch-users] Voice mail migration In-Reply-To: References: Message-ID: There shouldn't be any surprises then. The DB structure from july till now for mod_voicemail hasn't changed that much anyways. However the core_db did change slightly, there is the new basic_call table at the core db which replaced another table I can't remember off the top of my head atm. On Tue, Nov 8, 2011 at 5:21 PM, Lloyd Aloysius wrote: > Thank you for the reply. Old server running > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) > New Server ... Planning to run from the git. > > > Thanks and regards, > > Lloyd > > > On Tue, Nov 8, 2011 at 7:43 PM, Michael Collins wrote: >> >> unless the previous server is really old, I doubt there will be any >> trouble migrating. I would run a test and see how it goes. >> -MC >> >> On Tue, Nov 8, 2011 at 4:39 PM, Lloyd Aloysius >> wrote: >>> >>> Micheal, >>> Thank you for the reply. Yes same domain . But New server have a new >>> version of FreeSWITCH. >>> Do I need to maintain the same version FreeSWITCH for my new server. >>> >>> Thanks and regards, >>> >>> Lloyd >>> >>> >>> On Tue, Nov 8, 2011 at 7:32 PM, Michael Collins >>> wrote: >>>> >>>> Is your domain the same? Basically you have the voicemail.db file and >>>> you have /usr/local/freeswitch/storage/voicemail/default/xxxx where xxx is >>>> the domain. Shouldn't be too hard to copy everything over. >>>> -MC >>>> >>>> On Tue, Nov 8, 2011 at 4:05 PM, Lloyd Aloysius >>>> wrote: >>>>> >>>>> HI, >>>>> I am in the process of moving my current freeswitch server to a new >>>>> server. What is the best way to migrate the VoiceMail from old to new >>>>> server? >>>>> Any help is appreciated. >>>>> >>>>> Thanks and regards, >>>>> >>>>> Lloyd >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From alec.taylor6 at gmail.com Wed Nov 9 05:14:30 2011 From: alec.taylor6 at gmail.com (Alec Taylor) Date: Wed, 9 Nov 2011 13:14:30 +1100 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: <026e01cc9e5f$cc928d60$65b7a820$@fabulous4.co.uk> References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> <021f01cc9e45$c8a859b0$59f90d10$@fabulous4.co.uk> <026e01cc9e5f$cc928d60$65b7a820$@fabulous4.co.uk> Message-ID: Hasn't the monthly FreeSWITCH meeting solved this very problem? The call is broadcast online &etc On Wed, Nov 9, 2011 at 8:45 AM, Andy Ayers wrote: > Hi Michael, > > > > Here?s the scenario: > > > > The ?speaker? calls in to the switch and is invited to begin speaking after > the beep just as if they were recording an answering message and the audio > is recorded (in our case via a shoutcast server). This bit is working. > > > > These messages can last several minutes and in some cases hours so I would > like to add to this the ability for ?listeners? to call in to the switch and > be patched in to the above call to listen only to the message live in real > time. The listeners will not be allowed to interact with the call in any > way. > > > > Because this is optional however there won?t always be listeners, sometimes > it will be a simple dial in and record for the speaker. > > > > Hope I?m making more sense now. > > > > Many thanks for persevering > > Andy > > > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael > Collins > Sent: 08 November 2011 18:53 > > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Best way to create a one to many 'broadcast' > > > > What is the "main call" in this scenario? Sorry, I'm more confused than > ever. Perhaps you could restate the question. I also respond well to > pictures. :P > > > > -MC > > On Tue, Nov 8, 2011 at 10:39 AM, Andy Ayers wrote: > > Hi Michael, > > > > Apologies for the confusion there will only ever be a single audio source, > is the conference still the best way or can the other callers eavesdrop > somehow on the main call? > > > > Many thanks for your help > > Andy > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael > Collins > Sent: 08 November 2011 18:15 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Best way to create a one to many 'broadcast' > > > > What are the conditions under which some will participate? Will there be a > defined subset of callers who can participate and another subset who will be > listen only? I don't see how you can avoid using a conference if more than > one audio source will be used and multiple other endpoints will be > listening. > > > > -MC > > On Tue, Nov 8, 2011 at 8:37 AM, Andy Ayers wrote: > > Hi Alec, > > Yes sometimes, hence the challenge. > > Andy > > > > On 8 November 2011 16:34, Alec Taylor wrote: > > Sounds like you want to just use SHOUTcast directly, without using > FreeSwitch. > > Will there be anyone participating in the conference call? > > On Wed, Nov 9, 2011 at 3:31 AM, Andy Ayers wrote: >> Many thanks for the quick response Michael, I'd rather not use the >> recorded >> stream if I can help it as I'm pushing the recorded stream out to a >> shoutcast server and recording it there. Is there a way to acheive what I >> need by connecting the calls instead? >> >> On 8 November 2011 16:18, Michael Collins wrote: >>> >>> You can use mod_local_stream. Just set up a stream like the hold music on >>> but using the recorded message. Note: a local stream just plays over and >>> over again in a loop, so each call won't necessarily start at the >>> beginning >>> of the message. >>> -MC >>> >>> On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers wrote: >>>> >>>> Hi, >>>> >>>> >>>> >>>> I would like to use freeswitch to host a one to many broadcast style >>>> conference call. The challenge I have is that there won?t always be any >>>> listeners so the conference application isn?t quite working for me. >>>> >>>> >>>> >>>> Essentially, the speaker dials in with something important to say, by >>>> default the system will simply record what they say as an archived >>>> message >>>> and then they hang up. This works fine but I would like to extend the >>>> functionality so that other people dialling in have the option to >>>> connect >>>> into this call and listen to the message live. >>>> >>>> >>>> >>>> Can you advise me on the best way of achieving this: conference, bridge, >>>> eavesdrop or something else? >>>> >>>> >>>> >>>> Many thanks >>>> >>>> Andy >>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From lloyd.aloysius at sunteltech.ca Wed Nov 9 06:28:57 2011 From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius) Date: Tue, 8 Nov 2011 22:28:57 -0500 Subject: [Freeswitch-users] Voice mail migration In-Reply-To: References: Message-ID: How to get the FreeSWITCH git for the following version FreeSWITCH Version 1.0.head *(git-0675b59 2011-06-06 21-28-14 -0500)* Thanks Lloyd * * On Tue, Nov 8, 2011 at 8:52 PM, curriegrad2004 wrote: > There shouldn't be any surprises then. The DB structure from july till > now for mod_voicemail hasn't changed that much anyways. However the > core_db did change slightly, there is the new basic_call table at the > core db which replaced another table I can't remember off the top of > my head atm. > > On Tue, Nov 8, 2011 at 5:21 PM, Lloyd Aloysius > wrote: > > Thank you for the reply. Old server running > > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) > > New Server ... Planning to run from the git. > > > > > > Thanks and regards, > > > > Lloyd > > > > > > On Tue, Nov 8, 2011 at 7:43 PM, Michael Collins > wrote: > >> > >> unless the previous server is really old, I doubt there will be any > >> trouble migrating. I would run a test and see how it goes. > >> -MC > >> > >> On Tue, Nov 8, 2011 at 4:39 PM, Lloyd Aloysius < > lloyd.aloysius at gmail.com> > >> wrote: > >>> > >>> Micheal, > >>> Thank you for the reply. Yes same domain . But New server have a new > >>> version of FreeSWITCH. > >>> Do I need to maintain the same version FreeSWITCH for my new server. > >>> > >>> Thanks and regards, > >>> > >>> Lloyd > >>> > >>> > >>> On Tue, Nov 8, 2011 at 7:32 PM, Michael Collins > >>> wrote: > >>>> > >>>> Is your domain the same? Basically you have the voicemail.db file and > >>>> you have /usr/local/freeswitch/storage/voicemail/default/xxxx where > xxx is > >>>> the domain. Shouldn't be too hard to copy everything over. > >>>> -MC > >>>> > >>>> On Tue, Nov 8, 2011 at 4:05 PM, Lloyd Aloysius > >>>> wrote: > >>>>> > >>>>> HI, > >>>>> I am in the process of moving my current freeswitch server to a new > >>>>> server. What is the best way to migrate the VoiceMail from old to new > >>>>> server? > >>>>> Any help is appreciated. > >>>>> > >>>>> Thanks and regards, > >>>>> > >>>>> Lloyd > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>> > >>>> > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111108/102f265e/attachment.html From curriegrad2004 at gmail.com Wed Nov 9 06:59:26 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 8 Nov 2011 19:59:26 -0800 Subject: [Freeswitch-users] Voice mail migration In-Reply-To: References: Message-ID: Run a "make current". That should do the git pull and update the build to the latest git for you. On Tue, Nov 8, 2011 at 7:28 PM, Lloyd Aloysius wrote: > How to get the FreeSWITCH git for the following version > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) > Thanks > Lloyd > > > > > > > On Tue, Nov 8, 2011 at 8:52 PM, curriegrad2004 > wrote: >> >> There shouldn't be any surprises then. The DB structure from july till >> now for mod_voicemail hasn't changed that much anyways. However the >> core_db did change slightly, there is the new basic_call table at the >> core db which replaced another table I can't remember off the top of >> my head atm. >> >> On Tue, Nov 8, 2011 at 5:21 PM, Lloyd Aloysius >> wrote: >> > Thank you for the reply. Old server running >> > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) >> > New Server ... Planning to run from the git. >> > >> > >> > Thanks and regards, >> > >> > Lloyd >> > >> > >> > On Tue, Nov 8, 2011 at 7:43 PM, Michael Collins >> > wrote: >> >> >> >> unless the previous server is really old, I doubt there will be any >> >> trouble migrating. I would run a test and see how it goes. >> >> -MC >> >> >> >> On Tue, Nov 8, 2011 at 4:39 PM, Lloyd Aloysius >> >> >> >> wrote: >> >>> >> >>> Micheal, >> >>> Thank you for the reply. Yes same domain . But New server have a new >> >>> version of FreeSWITCH. >> >>> Do I need to maintain the same version FreeSWITCH for my new server. >> >>> >> >>> Thanks and regards, >> >>> >> >>> Lloyd >> >>> >> >>> >> >>> On Tue, Nov 8, 2011 at 7:32 PM, Michael Collins >> >>> wrote: >> >>>> >> >>>> Is your domain the same? Basically you have the voicemail.db file and >> >>>> you have /usr/local/freeswitch/storage/voicemail/default/xxxx where >> >>>> xxx is >> >>>> the domain. Shouldn't be too hard to copy everything over. >> >>>> -MC >> >>>> >> >>>> On Tue, Nov 8, 2011 at 4:05 PM, Lloyd Aloysius >> >>>> wrote: >> >>>>> >> >>>>> HI, >> >>>>> I am in the process of moving my current freeswitch server to a new >> >>>>> server. What is the best way to migrate the VoiceMail from old to >> >>>>> new >> >>>>> server? >> >>>>> Any help is appreciated. >> >>>>> >> >>>>> Thanks and regards, >> >>>>> >> >>>>> Lloyd >> >>>>> >> >>>>> FreeSWITCH-users mailing list >> >>>>> FreeSWITCH-users at lists.freeswitch.org >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>>> >> >>>>> >> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>> http://www.freeswitch.org >> >>>>> >> >>>> >> >>>> >> >>>> >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>>> >> >>> >> >>> >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From acrow at integrafin.co.uk Wed Nov 9 10:10:19 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 09 Nov 2011 07:10:19 +0000 Subject: [Freeswitch-users] mod_opal In-Reply-To: <4EB9BFE8.3050701@puzzled.xs4all.nl> References: <4EB998A2.8030206@integrafin.co.uk> <4EB9BFE8.3050701@puzzled.xs4all.nl> Message-ID: <4EBA275B.2030600@integrafin.co.uk> On 08/11/11 23:48, Patrick Lists wrote: > On 11/08/2011 10:01 PM, Alex Crow wrote: >> Hi all, >> >> I'm having some issues getting mod_opal installed - my intention is to >> use its IAX2 provision with iaxmodem and hylafax to get faxing through >> hylafax via a SIP trunk to a Mitel 3300 acting as an ISDN gateway. > I wasn't aware that IAX2 was in working order in opal/mod_opal. Have you > verified this? > Patrick No, I presumed from the Wiki that it supports IAX2, as it says on the page. Is this not the case? If so, is there any other way to hook Hylafax up to Asterisk for normal (not t.38) faxing, eg via ISDN or POTS? >> It just seems it does not want to compile - I'm using the Sirius release >> of PTLIB and OPAL, but it did also have a go with Lalande. >> >> So my first question would be - is mod_opal maintained any more? Or is >> there and equivalent "sipmodem" project that would replace iaxmodem? > A few months back on a CentOS 5.6 x86 box I built ptlib 2.6.8, opal > 3.6.8 and mod_opal compiled fine. Which distro are you using and what > are the errors you are seeing? I wiped my git source tree and started again. It built. However the buildopal.sh does not work. It pulls the trunk of ptlib which is too new. I had to get them and build by hand. Also the Wiki docs are out of date. Lalande is not new enough to build mod_opal, at least for me. > Regards, > Patrick Thanks Alex > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From kiruthika.bkite at gmail.com Wed Nov 9 10:28:01 2011 From: kiruthika.bkite at gmail.com (kiruthika sri) Date: Wed, 9 Nov 2011 12:58:01 +0530 Subject: [Freeswitch-users] FreeTDM PRI tapping issue In-Reply-To: References: Message-ID: Resolved SEGFAULT by changing the following line, crv = tap_pri_get_crv(pritap->pri, e->ring.call) into crv = tap_pri_get_crv(pritap->pri, e->proceeding.call) under the case PRI_EVENT_PROGRESS from ftmod_pritap.c source. On Sat, Nov 5, 2011 at 9:55 AM, kiruthika sri wrote: > > > ---------- Forwarded message ---------- > From: kiruthika sri > Date: Thu, Nov 3, 2011 at 10:11 AM > Subject: Re: [Freeswitch-users] FreeTDM PRI tapping issue > To: FreeSWITCH Users Help > > > Getting segfault after getting RINGING event in the first call itself. > > > On Wed, Nov 2, 2011 at 9:01 PM, Moises Silva wrote: > >> getting segfault right away? after a few calls? days? >> >> On Wed, Nov 2, 2011 at 12:35 AM, kiruthika sri > > wrote: >> >>> >>> Hi, >>> >>> I am trying to use the ftmod_pritap module for passive call recording. >>> >>> I did the followings. >>> * Installed wanpipe 3.5.23 for sangoma card AFT A102. >>> * Installed tap 1.4 (customized version of librpi for passive tapping) >>> * Installed freeswitch and freetdm with --with-pritap option >>> * Configured wanpipe, freetdm and freeswitch. >>> >>> Facing the following problem : >>> * FreeSWITCH is getting killed >>> * Getting SIGSEGV >>> >>> # wanrouter messages >>> >>> freeswitch[19720]: segfault at 8 ip 00ae00d1 sp b6ead208 error 4 in >>> libpri.so.1.4[ac5000+4c000] >>> >>> # gdb freeswitch core >>> >>> Program terminated with signal 11, Segmentation fault. >>> #0 0x00ae00d1 in q931_call_getcrv (ctrl=0x99fd9c0, call=0x0, >>> callmode=0xb6ead24c) at q931.c:5335 >>> 5335 *callmode = call->cr & 0x7; >>> >>> Kindly do the help on this. >>> >>> Thanks in advance. >>> >>> Regards >>> Kiruthika.U >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/46fcf206/attachment-0001.html From Stefan.Weigel at allianz-warranty.com Wed Nov 9 11:58:31 2011 From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan) Date: Wed, 9 Nov 2011 09:58:31 +0100 Subject: [Freeswitch-users] call groups - sort order Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36FA8@AZWSMS03.azwarranty.int> Hi list, from the WIKI page: group call:groupname[:order] How are the entries sorted ? How can I make sure the list returned by 'group call' is sorted in a certain order ? I want to make sure a certain extension is ringed first, but sometimes another extension is listed first. Thanks and best regards Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/ba738743/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 5972 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/ba738743/attachment.bin From miha at softnet.si Wed Nov 9 12:13:19 2011 From: miha at softnet.si (Miha Zoubek) Date: Wed, 09 Nov 2011 10:13:19 +0100 Subject: [Freeswitch-users] Failed adding Freeswitch-Other-Leg-Id Message-ID: <4EBA442F.8060900@softnet.si> Hi, I have configured mod_radius_cdr. Why I am getting this error? What does it mean? 2011-11-09 08:58:34.044939 [ERR] mod_radius_cdr.c:218 [mod_radius_cdr] Failed adding Freeswitch-Other-Leg-Id: c81927a2-9173-4b69-ad1b-e9501ba1385d BR, Miha From miha at softnet.si Wed Nov 9 12:40:38 2011 From: miha at softnet.si (Miha Zoubek) Date: Wed, 09 Nov 2011 10:40:38 +0100 Subject: [Freeswitch-users] Failed adding Freeswitch-Other-Leg-Id In-Reply-To: <4EBA442F.8060900@softnet.si> References: <4EBA442F.8060900@softnet.si> Message-ID: <4EBA4A96.5020206@softnet.si> On 11/9/2011 10:13 AM, Miha Zoubek wrote: > Hi, > > I have configured mod_radius_cdr. Why I am getting this error? What does > it mean? > > 2011-11-09 08:58:34.044939 [ERR] mod_radius_cdr.c:218 [mod_radius_cdr] > Failed adding Freeswitch-Other-Leg-Id: c81927a2-9173-4b69-ad1b-e9501ba1385d > > > BR, > Miha > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Figure it out! BR, Miha From freeswitch-list at puzzled.xs4all.nl Wed Nov 9 12:56:45 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 09 Nov 2011 10:56:45 +0100 Subject: [Freeswitch-users] mod_opal In-Reply-To: <4EBA275B.2030600@integrafin.co.uk> References: <4EB998A2.8030206@integrafin.co.uk> <4EB9BFE8.3050701@puzzled.xs4all.nl> <4EBA275B.2030600@integrafin.co.uk> Message-ID: <4EBA4E5D.9060405@puzzled.xs4all.nl> On 11/09/2011 08:10 AM, Alex Crow wrote: [snip] > No, I presumed from the Wiki that it supports IAX2, as it says on the > page. Is this not the case? As far as I know IAX2 is unsupported/does not work with FreeSWITCH. Hence my question. > If so, is there any other way to hook Hylafax up to Asterisk for normal > (not t.38) faxing, eg via ISDN or POTS? My old Asterisk box has ISDN and sends and receives faxes fine: ISDN -> chan_capi -> asterisk -> iaxmodem -> hylafax But you do realize this is the FreeSWITCH mailing list right? Since you are referring to Asterisk :) If you are interested in FreeSWTICH & sending/receiving faxes then have a look at: http://wiki.freeswitch.org/wiki/Mod_spandsp [snip] > I wiped my git source tree and started again. It built. However the > buildopal.sh does not work. It pulls the trunk of ptlib which is too new. I never used buildopal.sh. Sorry I forgot to mention that. That script indeed seems to do the wrong thing. > I had to get them and build by hand. Also the Wiki docs are out of date. > Lalande is not new enough to build mod_opal, at least for me. I used Sirius (ptlib 2.8/opal 3.8) and could build mod_opal fine. Iirc someone on this list mentioned he was successfully using ptlib 2.10 and opal 3.10 (the Luyten release). Regards, Patrick From acrow at integrafin.co.uk Wed Nov 9 14:05:57 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 09 Nov 2011 11:05:57 +0000 Subject: [Freeswitch-users] mod_opal In-Reply-To: <4EBA4E5D.9060405@puzzled.xs4all.nl> References: <4EB998A2.8030206@integrafin.co.uk> <4EB9BFE8.3050701@puzzled.xs4all.nl> <4EBA275B.2030600@integrafin.co.uk> <4EBA4E5D.9060405@puzzled.xs4all.nl> Message-ID: <4EBA5E95.30606@integrafin.co.uk> On 09/11/11 09:56, Patrick Lists wrote: > As far as I know IAX2 is unsupported/does not work with FreeSWITCH. > Hence my question. > Patrick, The Wiki page for mod_opal seems to suggest otherwise but does not show how to configure it, which is why I'm confused. Anyone else know about it? > > My old Asterisk box has ISDN and sends and receives faxes fine: > ISDN -> chan_capi -> asterisk -> iaxmodem -> hylafax > > But you do realize this is the FreeSWITCH mailing list right? Since you > are referring to Asterisk :) Do'h! Of course I meant FreeSwitch! > If you are interested in FreeSWTICH& sending/receiving faxes then have > a look at: http://wiki.freeswitch.org/wiki/Mod_spandsp Really has to be Hylafax, as we need desktop faxing, not via email. We have a large user base using WHFC and some using Excel macros to send faxes! It looks like I may have to look at using t38modem and gatewaying t38 to audio towards the ISDN. > I never used buildopal.sh. Sorry I forgot to mention that. That script > indeed seems to do the wrong thing. > Yes. > I used Sirius (ptlib 2.8/opal 3.8) and could build mod_opal fine. Iirc > someone on this list mentioned he was successfully using ptlib 2.10 and > opal 3.10 (the Luyten release). > OK, good to know. > Regards, > Patrick Many thanks, Alex From admin at blindi.net Wed Nov 9 17:12:50 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Wed, 9 Nov 2011 15:12:50 +0100 (CET) Subject: [Freeswitch-users] I missing conference moderator features Message-ID: Hi all, I have a few suggestions as to improve the usability conferences as moderator. For example: you press a key, and become these menuoptions: with 1 and 3, they can search through the channels in lissen only mode. press 2 for the current channel. press 4 to bridge the current channel with your self. press 5 to mute the current channel. press 6 to kick the channel from the conference. Press 7 to move the current channel to another conference-number. press the 8 to the subscriber number of the user to hear. So you can simply remove disturb as moderator of the conference soon. eavdrop has only limiting functions. Unfortunately, I find no application where this enables channel-search mode. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From miha at softnet.si Wed Nov 9 17:17:02 2011 From: miha at softnet.si (Miha Zoubek) Date: Wed, 09 Nov 2011 15:17:02 +0100 Subject: [Freeswitch-users] freeswitch radiusclinet (mod_radius_cdr) problems Message-ID: <4EBA8B5E.2050900@softnet.si> Hi, radiusclient on freeswitch in sending more that once same response. So I am getting in my sql tables for every call two inputs which is wrong. How can I deal whit this issue? I have paste a log from freeradius server in pastebin that you can see what is freeswitch sending. http://pastebin.freeswitch.org/17730 Thank you! BR, Miha From mstockton at harqen.com Wed Nov 9 17:52:47 2011 From: mstockton at harqen.com (Matt Stockton) Date: Wed, 9 Nov 2011 08:52:47 -0600 Subject: [Freeswitch-users] mod_rtmp compile error on Ubuntu 11.10 Message-ID: Has anyone successfully built mod_rtmp on Ubuntu 11.10? I'm seeing the following errors: making install mod_rtmp Compiling rtmp.c... rtmp.c: In function ?rtmp_send_message?: rtmp.c:529:18: error: variable ?status? set but not used [-Werror=unused-but-set-variable] cc1: all warnings being treated as errors I successfully compiled when I inserted a hack to use the variables into rtmp.c and mod_rtmp.c, but wondering what the right way to fix this is. Thanks, Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/99c48046/attachment.html From curriegrad2004 at gmail.com Wed Nov 9 18:10:57 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 9 Nov 2011 07:10:57 -0800 Subject: [Freeswitch-users] mod_rtmp compile error on Ubuntu 11.10 In-Reply-To: References: Message-ID: First, file this bug in JIRA. Then, if you really want to, go in the makefile of mod_rtmp and comment out that -Werror flag on the offending file. I'm guessing you're using GCC 4.6.x series, are you? On Wed, Nov 9, 2011 at 6:52 AM, Matt Stockton wrote: > Has anyone successfully built mod_rtmp on Ubuntu 11.10? I'm seeing the > following errors: > making install mod_rtmp > Compiling rtmp.c... > rtmp.c: In function ?rtmp_send_message?: > rtmp.c:529:18: error: variable ?status? set but not used > [-Werror=unused-but-set-variable] > cc1: all warnings being treated as errors > I successfully compiled when I inserted a hack to use the variables into > rtmp.c and mod_rtmp.c, but wondering what the right way to fix this is. > Thanks, > 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 > > From trob at freemail.hu Wed Nov 9 18:11:29 2011 From: trob at freemail.hu (=?ISO-8859-2?Q?T=F3th_R=F3bert?=) Date: Wed, 09 Nov 2011 16:11:29 +0100 Subject: [Freeswitch-users] BLF to show something Message-ID: <4EBA9821.60203@freemail.hu> I tried to send event this ways: http://wiki.freeswitch.org/wiki/User:Agx#About_PUBLISH_.2F_PRESENCE But i had no success, the lamp does not change. I expected it chaneg for for some seconds/minutes until next subscribe message. But it does not. And i have tried to send events by LUA, the results are the same: nothing. Do you have an idea? Thanks Rob On Mon, Nov 7, 2011 at 22:19, Anthony Minessale wrote: >/ /you can sit in ESL and listen for PRESENCE_PROBE events and respond with >/ /PRESENCE_IN all you want FYI >/ /On Mon, Nov 7, 2011 at 3:17 AM, Fran?ois Delawarde >/ /> wrote: >/ />/ On Mon, 2011-11-07 at 05:34 +0100, T?th R?bert wrote: />/ />/ > I got two idea: />/ />/ > - sub to dp+ext where ext is the extent you want to monitor />/ />/ > - use the new mapping feature to map it to a proto. />/ />/ />/ />/ A really cool feature would be to have some type of "presenceplan" (like />/ />/ dialplan or chatplan) called upon SUBSCRIBE. />/ />/ />/ />/ It would allow to setup mappings to local extensions/queues/..., given />/ />/ any type of condition on the origin and destination, refuse the />/ />/ subscription given ACLs, send a status fetched from ODBC, or why not />/ />/ even call an external program to fetch a status every time we send a />/ />/ NOTIFY. />/ />/ />/ />/ More or less, "The Ultimate Presence Server"... />/ />/ />/ />/ I would note that the new mapping feature also seems quite powerful and />/ />/ fun! />/ />/ />/ />/ Fran?ois./ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/331ad18f/attachment.html From lloyd.aloysius at gmail.com Wed Nov 9 19:01:59 2011 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Wed, 9 Nov 2011 11:01:59 -0500 Subject: [Freeswitch-users] Voice mail migration In-Reply-To: References: Message-ID: I am not looking for a new version. I am looking for a old version FreeSWITCH Version 1.0.head *(git-0675b59 2011-06-06 21-28-14 -0500)* * * How to use git to get this version into my server. I am not a git expert. Any help is appreciated. Thanks Lloyd On Tue, Nov 8, 2011 at 10:59 PM, curriegrad2004 wrote: > Run a "make current". That should do the git pull and update the build > to the latest git for you. > > On Tue, Nov 8, 2011 at 7:28 PM, Lloyd Aloysius > wrote: > > How to get the FreeSWITCH git for the following version > > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) > > Thanks > > Lloyd > > > > > > > > > > > > > > On Tue, Nov 8, 2011 at 8:52 PM, curriegrad2004 > > > wrote: > >> > >> There shouldn't be any surprises then. The DB structure from july till > >> now for mod_voicemail hasn't changed that much anyways. However the > >> core_db did change slightly, there is the new basic_call table at the > >> core db which replaced another table I can't remember off the top of > >> my head atm. > >> > >> On Tue, Nov 8, 2011 at 5:21 PM, Lloyd Aloysius > >> wrote: > >> > Thank you for the reply. Old server running > >> > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) > >> > New Server ... Planning to run from the git. > >> > > >> > > >> > Thanks and regards, > >> > > >> > Lloyd > >> > > >> > > >> > On Tue, Nov 8, 2011 at 7:43 PM, Michael Collins > >> > wrote: > >> >> > >> >> unless the previous server is really old, I doubt there will be any > >> >> trouble migrating. I would run a test and see how it goes. > >> >> -MC > >> >> > >> >> On Tue, Nov 8, 2011 at 4:39 PM, Lloyd Aloysius > >> >> > >> >> wrote: > >> >>> > >> >>> Micheal, > >> >>> Thank you for the reply. Yes same domain . But New server have a new > >> >>> version of FreeSWITCH. > >> >>> Do I need to maintain the same version FreeSWITCH for my new server. > >> >>> > >> >>> Thanks and regards, > >> >>> > >> >>> Lloyd > >> >>> > >> >>> > >> >>> On Tue, Nov 8, 2011 at 7:32 PM, Michael Collins > > >> >>> wrote: > >> >>>> > >> >>>> Is your domain the same? Basically you have the voicemail.db file > and > >> >>>> you have /usr/local/freeswitch/storage/voicemail/default/xxxx where > >> >>>> xxx is > >> >>>> the domain. Shouldn't be too hard to copy everything over. > >> >>>> -MC > >> >>>> > >> >>>> On Tue, Nov 8, 2011 at 4:05 PM, Lloyd Aloysius > >> >>>> wrote: > >> >>>>> > >> >>>>> HI, > >> >>>>> I am in the process of moving my current freeswitch server to a > new > >> >>>>> server. What is the best way to migrate the VoiceMail from old to > >> >>>>> new > >> >>>>> server? > >> >>>>> Any help is appreciated. > >> >>>>> > >> >>>>> Thanks and regards, > >> >>>>> > >> >>>>> Lloyd > >> >>>>> > >> >>>>> FreeSWITCH-users mailing list > >> >>>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>>> > >> >>>>> > >> >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>>> http://www.freeswitch.org > >> >>>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> FreeSWITCH-users mailing list > >> >>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>> > >> >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>> http://www.freeswitch.org > >> >>>> > >> >>> > >> >>> > >> >>> > >> >>> FreeSWITCH-users mailing list > >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> > >> >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> http://www.freeswitch.org > >> >>> > >> >> > >> >> > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > > >> > > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/5c6961e1/attachment-0001.html From david.ponzone at ipeva.fr Wed Nov 9 19:03:43 2011 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 9 Nov 2011 17:03:43 +0100 Subject: [Freeswitch-users] How to prevent RFC2833-relaying Message-ID: <8A1257D8-1943-4B9B-A506-988391854760@ipeva.fr> Hello all, I have a FS box between customers and carriers, and I receive RFC-2833 from carriers, and FS relays them to customers. Classic. I am looking for a way to tell FS, for specific calls, not to relay the RFC-2833 events to customers. Is there a such thing in the code ? Thank you David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/a17d9f59/attachment.html From mstockton at harqen.com Wed Nov 9 19:07:02 2011 From: mstockton at harqen.com (Matt Stockton) Date: Wed, 9 Nov 2011 10:07:02 -0600 Subject: [Freeswitch-users] mod_rtmp compile error on Ubuntu 11.10 In-Reply-To: References: Message-ID: Okay thanks for the info. I was looking for the right place to comment out the gcc args but couldn't quite follow where I should comment it out. I checked in the mod_rtmp Makefile and didn't see anything obvious to comment out. Also, grepped for the -Werror=unused-but-set-variable in the code base after bootstrapping and configuring and every place that had it seemed to be commented out already. On Wed, Nov 9, 2011 at 9:10 AM, curriegrad2004 wrote: > First, file this bug in JIRA. Then, if you really want to, go in the > makefile of mod_rtmp and comment out that -Werror flag on the > offending file. I'm guessing you're using GCC 4.6.x series, are you? > > On Wed, Nov 9, 2011 at 6:52 AM, Matt Stockton > wrote: > > Has anyone successfully built mod_rtmp on Ubuntu 11.10? I'm seeing the > > following errors: > > making install mod_rtmp > > Compiling rtmp.c... > > rtmp.c: In function ?rtmp_send_message?: > > rtmp.c:529:18: error: variable ?status? set but not used > > [-Werror=unused-but-set-variable] > > cc1: all warnings being treated as errors > > I successfully compiled when I inserted a hack to use the variables into > > rtmp.c and mod_rtmp.c, but wondering what the right way to fix this is. > > Thanks, > > 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/20111109/bd5711a8/attachment.html From chris.chen2004 at gmail.com Wed Nov 9 19:11:12 2011 From: chris.chen2004 at gmail.com (Chris Chen) Date: Wed, 9 Nov 2011 11:11:12 -0500 Subject: [Freeswitch-users] Voice mail migration In-Reply-To: References: Message-ID: Hi Lloyd, just do the following step, you will be where you want to be *git checkout 0675b59* Note: checking out '0675b59'. You are in 'detached HEAD' state. You can look around, make experimental changes and commit them, and you can discard any commits you make in this state without impacting any branches by performing another checkout. If you want to create a new branch to retain commits you create, you may do so (now or later) by using -b with the checkout command again. Example: git checkout -b new_branch_name HEAD is now at 0675b59... FS-3321 release rwlock on error On Wed, Nov 9, 2011 at 11:01 AM, Lloyd Aloysius wrote: > I am not looking for a new version. I am looking for a old version FreeSWITCH > Version 1.0.head *(git-0675b59 2011-06-06 21-28-14 -0500)* > * > * > How to use git to get this version into my server. I am not a git expert. > > Any help is appreciated. > > Thanks > Lloyd > > > > On Tue, Nov 8, 2011 at 10:59 PM, curriegrad2004 wrote: > >> Run a "make current". That should do the git pull and update the build >> to the latest git for you. >> >> On Tue, Nov 8, 2011 at 7:28 PM, Lloyd Aloysius >> wrote: >> > How to get the FreeSWITCH git for the following version >> > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) >> > Thanks >> > Lloyd >> > >> > >> > >> > >> > >> > >> > On Tue, Nov 8, 2011 at 8:52 PM, curriegrad2004 < >> curriegrad2004 at gmail.com> >> > wrote: >> >> >> >> There shouldn't be any surprises then. The DB structure from july till >> >> now for mod_voicemail hasn't changed that much anyways. However the >> >> core_db did change slightly, there is the new basic_call table at the >> >> core db which replaced another table I can't remember off the top of >> >> my head atm. >> >> >> >> On Tue, Nov 8, 2011 at 5:21 PM, Lloyd Aloysius >> >> wrote: >> >> > Thank you for the reply. Old server running >> >> > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) >> >> > New Server ... Planning to run from the git. >> >> > >> >> > >> >> > Thanks and regards, >> >> > >> >> > Lloyd >> >> > >> >> > >> >> > On Tue, Nov 8, 2011 at 7:43 PM, Michael Collins >> >> > wrote: >> >> >> >> >> >> unless the previous server is really old, I doubt there will be any >> >> >> trouble migrating. I would run a test and see how it goes. >> >> >> -MC >> >> >> >> >> >> On Tue, Nov 8, 2011 at 4:39 PM, Lloyd Aloysius >> >> >> >> >> >> wrote: >> >> >>> >> >> >>> Micheal, >> >> >>> Thank you for the reply. Yes same domain . But New server have a >> new >> >> >>> version of FreeSWITCH. >> >> >>> Do I need to maintain the same version FreeSWITCH for my new >> server. >> >> >>> >> >> >>> Thanks and regards, >> >> >>> >> >> >>> Lloyd >> >> >>> >> >> >>> >> >> >>> On Tue, Nov 8, 2011 at 7:32 PM, Michael Collins < >> msc at freeswitch.org> >> >> >>> wrote: >> >> >>>> >> >> >>>> Is your domain the same? Basically you have the voicemail.db file >> and >> >> >>>> you have /usr/local/freeswitch/storage/voicemail/default/xxxx >> where >> >> >>>> xxx is >> >> >>>> the domain. Shouldn't be too hard to copy everything over. >> >> >>>> -MC >> >> >>>> >> >> >>>> On Tue, Nov 8, 2011 at 4:05 PM, Lloyd Aloysius >> >> >>>> wrote: >> >> >>>>> >> >> >>>>> HI, >> >> >>>>> I am in the process of moving my current freeswitch server to a >> new >> >> >>>>> server. What is the best way to migrate the VoiceMail from old to >> >> >>>>> new >> >> >>>>> server? >> >> >>>>> Any help is appreciated. >> >> >>>>> >> >> >>>>> Thanks and regards, >> >> >>>>> >> >> >>>>> Lloyd >> >> >>>>> >> >> >>>>> FreeSWITCH-users mailing list >> >> >>>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>>> >> >> >>>>> >> >> >>>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> http://www.freeswitch.org >> >> >>>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> FreeSWITCH-users mailing list >> >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>> >> >> >>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>> http://www.freeswitch.org >> >> >>>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> FreeSWITCH-users mailing list >> >> >>> FreeSWITCH-users at lists.freeswitch.org >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> >> >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> http://www.freeswitch.org >> >> >>> >> >> >> >> >> >> >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> >> >> >> > >> >> > >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/3c7ee175/attachment-0001.html From mirjana.ljuboja at tp.rs Wed Nov 9 13:10:49 2011 From: mirjana.ljuboja at tp.rs (Mirjana Ljuboja) Date: Wed, 9 Nov 2011 11:10:49 +0100 Subject: [Freeswitch-users] Building esljni.dll (libesljni.so for Windows) Message-ID: I want to use esl.jar (Event Socket Library) on Windows (XP), but it requires equivalent of libesljni.so from Linux. I can compile and use the library on Linux, but not on the Windows. I was hoping someone has built the .dll on Windows and could share. Instruction on how to make it would be great too, if it works. Thanks Mira -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/cbb5db62/attachment.html From hi_tanin at yahoo.com Wed Nov 9 17:24:39 2011 From: hi_tanin at yahoo.com (hi tanin) Date: Wed, 9 Nov 2011 06:24:39 -0800 (PST) Subject: [Freeswitch-users] Hunting Phase Problem Message-ID: <1320848679.32667.YahooMailNeo@web125711.mail.ne1.yahoo.com> I am very new with FreeSwitch. Can anyone tell me: ????What is mean by hunting phase? ????When and how it is use and why and when it is required? I want to know details about hunting phase, ? Also plese give me a way or tell me when and how the variables($0, $1, $2) are used? ? Please help me -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/91426c59/attachment.html From david at styleflare.com Wed Nov 9 18:37:05 2011 From: david at styleflare.com (David) Date: Wed, 09 Nov 2011 10:37:05 -0500 Subject: [Freeswitch-users] NYC FreeSWITCH User Meetup Message-ID: <4EBA9E21.7040302@styleflare.com> Hey Just wanted to invite the folks in the NYC area to a renewed FreeSWITCH users meetup. Its taking place tomorrow night Thursday Nov 10, 2011 at 6pm. Here is the link to the meetup page. http://www.meetup.com/fsusers-ny/ Darren Schreiber from the 2600hz project will be there to present the whistle project. We will have some beer and light refreshments. Its free and open to everyone; although please RSVP. Thanks. From msc at freeswitch.org Wed Nov 9 19:17:34 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Nov 2011 08:17:34 -0800 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> <021f01cc9e45$c8a859b0$59f90d10$@fabulous4.co.uk> <026e01cc9e5f$cc928d60$65b7a820$@fabulous4.co.uk> Message-ID: I don't think so, but let's all call in today and talk about it! -MC On Tue, Nov 8, 2011 at 6:14 PM, Alec Taylor wrote: > Hasn't the monthly FreeSWITCH meeting solved this very problem? > > The call is broadcast online &etc > > On Wed, Nov 9, 2011 at 8:45 AM, Andy Ayers wrote: > > Hi Michael, > > > > > > > > Here?s the scenario: > > > > > > > > The ?speaker? calls in to the switch and is invited to begin speaking > after > > the beep just as if they were recording an answering message and the > audio > > is recorded (in our case via a shoutcast server). This bit is working. > > > > > > > > These messages can last several minutes and in some cases hours so I > would > > like to add to this the ability for ?listeners? to call in to the switch > and > > be patched in to the above call to listen only to the message live in > real > > time. The listeners will not be allowed to interact with the call in any > > way. > > > > > > > > Because this is optional however there won?t always be listeners, > sometimes > > it will be a simple dial in and record for the speaker. > > > > > > > > Hope I?m making more sense now. > > > > > > > > Many thanks for persevering > > > > Andy > > > > > > > > > > > > > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael > > Collins > > Sent: 08 November 2011 18:53 > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Best way to create a one to many > 'broadcast' > > > > > > > > What is the "main call" in this scenario? Sorry, I'm more confused than > > ever. Perhaps you could restate the question. I also respond well to > > pictures. :P > > > > > > > > -MC > > > > On Tue, Nov 8, 2011 at 10:39 AM, Andy Ayers > wrote: > > > > Hi Michael, > > > > > > > > Apologies for the confusion there will only ever be a single audio > source, > > is the conference still the best way or can the other callers eavesdrop > > somehow on the main call? > > > > > > > > Many thanks for your help > > > > Andy > > > > > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael > > Collins > > Sent: 08 November 2011 18:15 > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Best way to create a one to many > 'broadcast' > > > > > > > > What are the conditions under which some will participate? Will there be > a > > defined subset of callers who can participate and another subset who > will be > > listen only? I don't see how you can avoid using a conference if more > than > > one audio source will be used and multiple other endpoints will be > > listening. > > > > > > > > -MC > > > > On Tue, Nov 8, 2011 at 8:37 AM, Andy Ayers wrote: > > > > Hi Alec, > > > > Yes sometimes, hence the challenge. > > > > Andy > > > > > > > > On 8 November 2011 16:34, Alec Taylor wrote: > > > > Sounds like you want to just use SHOUTcast directly, without using > > FreeSwitch. > > > > Will there be anyone participating in the conference call? > > > > On Wed, Nov 9, 2011 at 3:31 AM, Andy Ayers wrote: > >> Many thanks for the quick response Michael, I'd rather not use the > >> recorded > >> stream if I can help it as I'm pushing the recorded stream out to a > >> shoutcast server and recording it there. Is there a way to acheive what > I > >> need by connecting the calls instead? > >> > >> On 8 November 2011 16:18, Michael Collins wrote: > >>> > >>> You can use mod_local_stream. Just set up a stream like the hold music > on > >>> but using the recorded message. Note: a local stream just plays over > and > >>> over again in a loop, so each call won't necessarily start at the > >>> beginning > >>> of the message. > >>> -MC > >>> > >>> On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers > wrote: > >>>> > >>>> Hi, > >>>> > >>>> > >>>> > >>>> I would like to use freeswitch to host a one to many broadcast style > >>>> conference call. The challenge I have is that there won?t always be > any > >>>> listeners so the conference application isn?t quite working for me. > >>>> > >>>> > >>>> > >>>> Essentially, the speaker dials in with something important to say, by > >>>> default the system will simply record what they say as an archived > >>>> message > >>>> and then they hang up. This works fine but I would like to extend the > >>>> functionality so that other people dialling in have the option to > >>>> connect > >>>> into this call and listen to the message live. > >>>> > >>>> > >>>> > >>>> Can you advise me on the best way of achieving this: conference, > bridge, > >>>> eavesdrop or something else? > >>>> > >>>> > >>>> > >>>> Many thanks > >>>> > >>>> Andy > >>>> > >>>> > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/5eb9927b/attachment-0001.html From msc at freeswitch.org Wed Nov 9 19:20:23 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Nov 2011 08:20:23 -0800 Subject: [Freeswitch-users] I missing conference moderator features In-Reply-To: References: Message-ID: You might want to put up a bounty on these features if you consider them to be particularly useful. FYI, you can also emulate this behavior using bind_digit_action and Lua scripts if you are not a C hacker. -MC On Wed, Nov 9, 2011 at 6:12 AM, Thomas Hoellriegel wrote: > Hi all, > I have a few suggestions as to improve the usability conferences as > moderator. > > For example: > you press a key, and become these menuoptions: > with 1 and 3, they can search through the channels in lissen only mode. > press 2 for the current channel. > press 4 to bridge the current channel with your self. > press 5 to mute the current channel. > press 6 to kick the channel from the conference. > Press 7 to move the current channel to another conference-number. > press the 8 to the subscriber number of the user to hear. > > So you can simply remove disturb as moderator of the conference soon. > > eavdrop has only limiting functions. > Unfortunately, I find no application where this enables channel-search > mode. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/273a90c4/attachment.html From Hector.Geraldino at ip-soft.net Wed Nov 9 19:28:14 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Wed, 9 Nov 2011 11:28:14 -0500 Subject: [Freeswitch-users] Building esljni.dll (libesljni.so for Windows) In-Reply-To: References: Message-ID: <6A6B4C284AD15042B429EB9D904544AD0224D43FD3@NY1-EXMB-01.ip-soft.net> I don't have an answer for your question, however I would recommend you to use the Java ESL Client, which has no dependencies on any native libraries (100% java) and works pretty good. http://wiki.freeswitch.org/wiki/Java_ESL_Client From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mirjana Ljuboja Sent: Wednesday, November 09, 2011 5:11 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Building esljni.dll (libesljni.so for Windows) I want to use esl.jar (Event Socket Library) on Windows (XP), but it requires equivalent of libesljni.so from Linux. I can compile and use the library on Linux, but not on the Windows. I was hoping someone has built the .dll on Windows and could share. Instruction on how to make it would be great too, if it works. Thanks Mira -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/82206480/attachment.html From avi at avimarcus.net Wed Nov 9 19:35:07 2011 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 9 Nov 2011 18:35:07 +0200 Subject: [Freeswitch-users] How to prevent RFC2833-relaying In-Reply-To: <8A1257D8-1943-4B9B-A506-988391854760@ipeva.fr> References: <8A1257D8-1943-4B9B-A506-988391854760@ipeva.fr> Message-ID: If no one else has a better answer.. Two options come to mind: 1) Use bind_digit_action to eat the dtmf. 2) turn off the relaying. It seems it has to be done before the channel is initialized though. http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type -Avi 2011/11/9 David Ponzone > Hello all, > > I have a FS box between customers and carriers, and I receive RFC-2833 > from carriers, and FS relays them to customers. Classic. > > I am looking for a way to tell FS, for specific calls, not to relay the > RFC-2833 events to customers. > > Is there a such thing in the code ? > > Thank you > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/1dff34ca/attachment.html From msc at freeswitch.org Wed Nov 9 19:37:13 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Nov 2011 08:37:13 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello FreeSWITCHers! Here's today's agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_09 We have a few miscellaneous items to discuss and I have some updates. Also, I'm hoping to continue the discussion on SIP 101 that we started last week. I think it would be a good time to go over the Contact: header and what it's for. That will set the ground work for what the NDLB-connectile-dysfunction parameter does. Talk to you soon, -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/1407ae35/attachment.html From brian at freeswitch.org Wed Nov 9 19:37:23 2011 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Nov 2011 10:37:23 -0600 Subject: [Freeswitch-users] Building esljni.dll (libesljni.so for Windows) In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0224D43FD3@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD0224D43FD3@NY1-EXMB-01.ip-soft.net> Message-ID: <5A22347D-7674-4A62-B547-02DE424EF260@freeswitch.org> I suspect nobody has done it yet. I would go make nice with our Windows Devs in #freeswitch ;) /b On Nov 9, 2011, at 10:28 AM, Hector Geraldino wrote: > I can compile and use the library on Linux, but not on the Windows. I was hoping someone has built the .dll on Windows and could share. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/d01be5e2/attachment-0001.html From mstockton at harqen.com Wed Nov 9 19:43:27 2011 From: mstockton at harqen.com (Matt Stockton) Date: Wed, 9 Nov 2011 10:43:27 -0600 Subject: [Freeswitch-users] mod_rtmp compile error on Ubuntu 11.10 In-Reply-To: References: Message-ID: Also, I added the JIRA, thanks. http://jira.freeswitch.org/browse/FS-3681 On Wed, Nov 9, 2011 at 10:07 AM, Matt Stockton wrote: > Okay thanks for the info. I was looking for the right place to comment out > the gcc args but couldn't quite follow where I should comment it out. I > checked in the mod_rtmp Makefile and didn't see anything obvious to comment > out. Also, grepped for the -Werror=unused-but-set-variable in the code base > after bootstrapping and configuring and every place that had it seemed to > be commented out already. > > > > On Wed, Nov 9, 2011 at 9:10 AM, curriegrad2004 wrote: > >> First, file this bug in JIRA. Then, if you really want to, go in the >> makefile of mod_rtmp and comment out that -Werror flag on the >> offending file. I'm guessing you're using GCC 4.6.x series, are you? >> >> On Wed, Nov 9, 2011 at 6:52 AM, Matt Stockton >> wrote: >> > Has anyone successfully built mod_rtmp on Ubuntu 11.10? I'm seeing the >> > following errors: >> > making install mod_rtmp >> > Compiling rtmp.c... >> > rtmp.c: In function ?rtmp_send_message?: >> > rtmp.c:529:18: error: variable ?status? set but not used >> > [-Werror=unused-but-set-variable] >> > cc1: all warnings being treated as errors >> > I successfully compiled when I inserted a hack to use the variables into >> > rtmp.c and mod_rtmp.c, but wondering what the right way to fix this is. >> > Thanks, >> > 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/20111109/495e9d7a/attachment.html From david at styleflare.com Wed Nov 9 20:06:42 2011 From: david at styleflare.com (David) Date: Wed, 09 Nov 2011 12:06:42 -0500 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: <4EBAB322.4060405@styleflare.com> Sounds Good. Lets also mention the FreeSWITCH NYC Meetup tomorrow night with Darren Schreiber from 2600hz. Come by after work and grab a beer with us. Show your support for the NYC FreeSWITCH community. Here is the link. http://bit.ly/sYmtx2 Thanks. On 11/9/11 11:37 AM, Michael Collins wrote: > Hello FreeSWITCHers! > > Here's today's agenda: > http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_09 > > We have a few miscellaneous items to discuss and I have some updates. > Also, I'm hoping to continue the discussion on SIP 101 that we started > last week. I think it would be a good time to go over the Contact: > header and what it's for. That will set the ground work for what the > NDLB-connectile-dysfunction parameter does. > > Talk to you soon, > -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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/2cba69a0/attachment.html From alec.taylor6 at gmail.com Wed Nov 9 20:18:57 2011 From: alec.taylor6 at gmail.com (Alec Taylor) Date: Thu, 10 Nov 2011 04:18:57 +1100 Subject: [Freeswitch-users] Best way to create a one to many 'broadcast' In-Reply-To: References: <015601cc9e24$a0dea9a0$e29bfce0$@fabulous4.co.uk> <021f01cc9e45$c8a859b0$59f90d10$@fabulous4.co.uk> <026e01cc9e5f$cc928d60$65b7a820$@fabulous4.co.uk> Message-ID: +1 :P On Thu, Nov 10, 2011 at 3:17 AM, Michael Collins wrote: > I don't think so, but let's all call in today and talk about it! > -MC > > On Tue, Nov 8, 2011 at 6:14 PM, Alec Taylor wrote: >> >> Hasn't the monthly FreeSWITCH meeting solved this very problem? >> >> The call is broadcast online &etc >> >> On Wed, Nov 9, 2011 at 8:45 AM, Andy Ayers wrote: >> > Hi Michael, >> > >> > >> > >> > Here?s the scenario: >> > >> > >> > >> > The ?speaker? calls in to the switch and is invited to begin speaking >> > after >> > the beep just as if they were recording an answering message and the >> > audio >> > is recorded (in our case via a shoutcast server). This bit is working. >> > >> > >> > >> > These messages can last several minutes and in some cases hours so I >> > would >> > like to add to this the ability for ?listeners? to call in to the switch >> > and >> > be patched in to the above call to listen only to the message live in >> > real >> > time. The listeners will not be allowed to interact with the call in any >> > way. >> > >> > >> > >> > Because this is optional however there won?t always be listeners, >> > sometimes >> > it will be a simple dial in and record for the speaker. >> > >> > >> > >> > Hope I?m making more sense now. >> > >> > >> > >> > Many thanks for persevering >> > >> > Andy >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > From: freeswitch-users-bounces at lists.freeswitch.org >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> > Michael >> > Collins >> > Sent: 08 November 2011 18:53 >> > >> > To: FreeSWITCH Users Help >> > Subject: Re: [Freeswitch-users] Best way to create a one to many >> > 'broadcast' >> > >> > >> > >> > What is the "main call" in this scenario? Sorry, I'm more confused than >> > ever. Perhaps you could restate the question. I also respond well to >> > pictures. :P >> > >> > >> > >> > -MC >> > >> > On Tue, Nov 8, 2011 at 10:39 AM, Andy Ayers >> > wrote: >> > >> > Hi Michael, >> > >> > >> > >> > Apologies for the confusion there will only ever be a single audio >> > source, >> > is the conference still the best way or can the other callers eavesdrop >> > somehow on the main call? >> > >> > >> > >> > Many thanks for your help >> > >> > Andy >> > >> > >> > >> > >> > >> > From: freeswitch-users-bounces at lists.freeswitch.org >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> > Michael >> > Collins >> > Sent: 08 November 2011 18:15 >> > To: FreeSWITCH Users Help >> > Subject: Re: [Freeswitch-users] Best way to create a one to many >> > 'broadcast' >> > >> > >> > >> > What are the conditions under which some will participate? Will there be >> > a >> > defined subset of callers who can participate and another subset who >> > will be >> > listen only? I don't see how you can avoid using a conference if more >> > than >> > one audio source will be used and multiple other endpoints will be >> > listening. >> > >> > >> > >> > -MC >> > >> > On Tue, Nov 8, 2011 at 8:37 AM, Andy Ayers wrote: >> > >> > Hi Alec, >> > >> > Yes sometimes, hence the challenge. >> > >> > Andy >> > >> > >> > >> > On 8 November 2011 16:34, Alec Taylor wrote: >> > >> > Sounds like you want to just use SHOUTcast directly, without using >> > FreeSwitch. >> > >> > Will there be anyone participating in the conference call? >> > >> > On Wed, Nov 9, 2011 at 3:31 AM, Andy Ayers wrote: >> >> Many thanks for the quick response Michael, I'd rather not use the >> >> recorded >> >> stream if I can help it as I'm pushing the recorded stream out to a >> >> shoutcast server and recording it there. Is there a way to acheive what >> >> I >> >> need by connecting the calls instead? >> >> >> >> On 8 November 2011 16:18, Michael Collins wrote: >> >>> >> >>> You can use mod_local_stream. Just set up a stream like the hold music >> >>> on >> >>> but using the recorded message. Note: a local stream just plays over >> >>> and >> >>> over again in a loop, so each call won't necessarily start at the >> >>> beginning >> >>> of the message. >> >>> -MC >> >>> >> >>> On Tue, Nov 8, 2011 at 6:42 AM, Andy Ayers >> >>> wrote: >> >>>> >> >>>> Hi, >> >>>> >> >>>> >> >>>> >> >>>> I would like to use freeswitch to host a one to many broadcast style >> >>>> conference call. The challenge I have is that there won?t always be >> >>>> any >> >>>> listeners so the conference application isn?t quite working for me. >> >>>> >> >>>> >> >>>> >> >>>> Essentially, the speaker dials in with something important to say, by >> >>>> default the system will simply record what they say as an archived >> >>>> message >> >>>> and then they hang up. This works fine but I would like to extend the >> >>>> functionality so that other people dialling in have the option to >> >>>> connect >> >>>> into this call and listen to the message live. >> >>>> >> >>>> >> >>>> >> >>>> Can you advise me on the best way of achieving this: conference, >> >>>> bridge, >> >>>> eavesdrop or something else? >> >>>> >> >>>> >> >>>> >> >>>> Many thanks >> >>>> >> >>>> Andy >> >>>> >> >>>> >> >>>> >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>>> >> >>> >> >>> >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Wed Nov 9 20:44:33 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Nov 2011 09:44:33 -0800 Subject: [Freeswitch-users] Hunting Phase Problem In-Reply-To: <1320848679.32667.YahooMailNeo@web125711.mail.ne1.yahoo.com> References: <1320848679.32667.YahooMailNeo@web125711.mail.ne1.yahoo.com> Message-ID: Tanin, Welcome to the FreeSWITCH community! We hope you enjoy your stay. :) Okay, these are basic but very good questions. If you want a really in depth discussion I recommend getting the FreeSWITCH book and reviewing chapters 5 and 8 which discuss dialplan processing in great detail. However, here is a very brief answer to your question: There are two basic "phases" to dialplan processing: "hunting" and "executing". During the hunting phase, the dialplan processor parses the dialplan looking for extensions with matching conditions. When a condition matches, any inside that condition are added to the "task list". (There are various ways to control this hunting behavior, but let's save that for a later discussion.) Once the hunting phase is done, there is a list of actions that needs to be executed. It's at this point that variables get "expanded". For example this condition yields a value for $1: $1 will contain whatever is matched in the expression. In this case, it will be a four-digit number, like "1000", "1001", etc. (For more information on this topic see http://wiki.freeswitch.org/wiki/Regex) See this pastebin where I have separated the hunting from the executing: http://pastebin.freeswitch.org/17732 It's a call to extension 1009. You first see the dialplan hunting starting at line #3. Notice all the debug lines that begin with "Dialplan:" - these represent the dialplan parser going over each extension's tags looking for matches. You'll see either PASS or FAIL on a match or no match. (Compare line #76 to line #78.) If there's a match then you'll see any actions being added to the task list. Look starting at line #79 and you'll see all the actions being added for this extension. NOTE: if you look in conf/dialplan/default.xml at the "Local_Extension" dialplan extension you'll see a direct correspondence to the tags there and the actions being added here in the log. Once the dialplan is done hunting it moves on to the execute phase. (See log entries after line #105.) Notice all the lines beginning with "EXECUTE:" - these represent the actions on the task list actually being executed, one at a time and in the order they were added. That's it! Well, not exactly, but this will get you started. There are a number of tricks and a few caveats, but you'll get to those in due course. Feel free to join #freeswitch on irc.freenode.net and chat with us in realtime. Also, we have a conference call every Wednesday. See the main wiki page for more information. -MC On Wed, Nov 9, 2011 at 6:24 AM, hi tanin wrote: > I am very new with FreeSwitch. Can anyone tell me: > What is mean by hunting phase? > When and how it is use and why and when it is required? > I want to know details about hunting phase, > > Also plese give me a way or tell me when and how the variables($0, $1, $2) > are used? > > Please help me > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/7cf7130b/attachment-0001.html From adrian.fuentes at hovanetworks.com Wed Nov 9 19:20:36 2011 From: adrian.fuentes at hovanetworks.com (Adrian Fuentes) Date: Wed, 9 Nov 2011 10:20:36 -0600 Subject: [Freeswitch-users] FreeSwitch as SBC Message-ID: <1554E94A-5A57-4033-9CC3-E42056D69A20@hovanetworks.com> hello all! can help describe what you can do FreeSwitch as SBC? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/227df71f/attachment.html From david.ponzone at ipeva.fr Wed Nov 9 20:48:33 2011 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 9 Nov 2011 18:48:33 +0100 Subject: [Freeswitch-users] How to prevent RFC2833-relaying In-Reply-To: References: <8A1257D8-1943-4B9B-A506-988391854760@ipeva.fr> Message-ID: <7C0F9004-CFDE-4834-971F-FAAFAE0F23FA@ipeva.fr> Avi, 1) I can definitely try that! 2) I think there is a mistake in the wiki. In the dtmf-type page, it says that it can't be done with the channel variable, but there is a page for dtmf_type where it looks like it can be set from the dialplan. I'll give it a try anyway. I also found a pass_rfc2833 variable, but it only accepts "true" and "false" ("true" means FS is transparent, "false" means FS will decode/recode the DTMF). I would need a "block" value. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 09/11/2011 ? 17:35, Avi Marcus a ?crit : > If no one else has a better answer.. > Two options come to mind: > 1) Use bind_digit_action to eat the dtmf. > 2) turn off the relaying. It seems it has to be done before the channel is initialized though. > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type > > -Avi > > > 2011/11/9 David Ponzone > Hello all, > > I have a FS box between customers and carriers, and I receive RFC-2833 from carriers, and FS relays them to customers. Classic. > > I am looking for a way to tell FS, for specific calls, not to relay the RFC-2833 events to customers. > > Is there a such thing in the code ? > > Thank you > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/3ba2cff6/attachment.html From acrow at integrafin.co.uk Wed Nov 9 21:17:57 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 09 Nov 2011 18:17:57 +0000 Subject: [Freeswitch-users] Anyone using t38modem with FS (gatewaying from audio<>t38) and Hylafax successfully? Message-ID: <4EBAC3D5.5020407@integrafin.co.uk> Hi list, I am trying to figure out, after abandoning trying to get iaxmodem working in FS, if it is possible to use t38modem to connect FS to Hylafax, mainly for the purpose of sending and receiving non-t38 faxes via an ISDN card or non-t38-aware gateway (I'm currently using our Mitel 3300 as the gateway and voice calls between the two systems are fine). I have managed after a lot of effort to get t38modem compiled and running, and it seems to react to incoming fax calls coming in via the Mitel gateway from a normal fax machine, but the fax machine does not hear any reply to its initiation tones and the call hangs up in the end. Ivan on this list has suggested that I don't really need to do the gatewaying as t38modem can work with audio over SIP as well, so I tried removing all the t38 lines from the dialplan but with no success. My fax dialplan looks like this: t38modem is invoked thusly: t38modem -tttt -o /var/log/t38modem.log --no-h323 -u t38modem --sip-register t38modem at 192.168.20.245,password --sip-listen udp$127.0.0.1:6060 --ptty +/dev/ttyT38-1 --route "modem:.*=sip:@192.168.20.245" --route "sip:.*=modem:" Where 192.168.20.245 is the ip of the FS box. Any ideas? I can provide logs if required. Thanks, Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From anthony.minessale at gmail.com Wed Nov 9 21:23:18 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Nov 2011 12:23:18 -0600 Subject: [Freeswitch-users] mod_rtmp compile error on Ubuntu 11.10 In-Reply-To: References: Message-ID: fixed On Wed, Nov 9, 2011 at 10:43 AM, Matt Stockton wrote: > Also, I added the JIRA, thanks. http://jira.freeswitch.org/browse/FS-3681 > > > On Wed, Nov 9, 2011 at 10:07 AM, Matt Stockton wrote: > >> Okay thanks for the info. I was looking for the right place to comment >> out the gcc args but couldn't quite follow where I should comment it out. I >> checked in the mod_rtmp Makefile and didn't see anything obvious to comment >> out. Also, grepped for the -Werror=unused-but-set-variable in the code base >> after bootstrapping and configuring and every place that had it seemed to >> be commented out already. >> >> >> >> On Wed, Nov 9, 2011 at 9:10 AM, curriegrad2004 wrote: >> >>> First, file this bug in JIRA. Then, if you really want to, go in the >>> makefile of mod_rtmp and comment out that -Werror flag on the >>> offending file. I'm guessing you're using GCC 4.6.x series, are you? >>> >>> On Wed, Nov 9, 2011 at 6:52 AM, Matt Stockton >>> wrote: >>> > Has anyone successfully built mod_rtmp on Ubuntu 11.10? I'm seeing the >>> > following errors: >>> > making install mod_rtmp >>> > Compiling rtmp.c... >>> > rtmp.c: In function ?rtmp_send_message?: >>> > rtmp.c:529:18: error: variable ?status? set but not used >>> > [-Werror=unused-but-set-variable] >>> > cc1: all warnings being treated as errors >>> > I successfully compiled when I inserted a hack to use the variables >>> into >>> > rtmp.c and mod_rtmp.c, but wondering what the right way to fix this is. >>> > Thanks, >>> > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/c5695cad/attachment-0001.html From mstockton at harqen.com Wed Nov 9 21:26:14 2011 From: mstockton at harqen.com (Matt Stockton) Date: Wed, 9 Nov 2011 12:26:14 -0600 Subject: [Freeswitch-users] mod_rtmp compile error on Ubuntu 11.10 In-Reply-To: References: Message-ID: Anthony, Thank you very much! Much appreciated. On Wed, Nov 9, 2011 at 12:23 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > fixed > > > On Wed, Nov 9, 2011 at 10:43 AM, Matt Stockton wrote: > >> Also, I added the JIRA, thanks. >> http://jira.freeswitch.org/browse/FS-3681 >> >> >> On Wed, Nov 9, 2011 at 10:07 AM, Matt Stockton wrote: >> >>> Okay thanks for the info. I was looking for the right place to comment >>> out the gcc args but couldn't quite follow where I should comment it out. I >>> checked in the mod_rtmp Makefile and didn't see anything obvious to comment >>> out. Also, grepped for the -Werror=unused-but-set-variable in the code base >>> after bootstrapping and configuring and every place that had it seemed to >>> be commented out already. >>> >>> >>> >>> On Wed, Nov 9, 2011 at 9:10 AM, curriegrad2004 >> > wrote: >>> >>>> First, file this bug in JIRA. Then, if you really want to, go in the >>>> makefile of mod_rtmp and comment out that -Werror flag on the >>>> offending file. I'm guessing you're using GCC 4.6.x series, are you? >>>> >>>> On Wed, Nov 9, 2011 at 6:52 AM, Matt Stockton >>>> wrote: >>>> > Has anyone successfully built mod_rtmp on Ubuntu 11.10? I'm seeing the >>>> > following errors: >>>> > making install mod_rtmp >>>> > Compiling rtmp.c... >>>> > rtmp.c: In function ?rtmp_send_message?: >>>> > rtmp.c:529:18: error: variable ?status? set but not used >>>> > [-Werror=unused-but-set-variable] >>>> > cc1: all warnings being treated as errors >>>> > I successfully compiled when I inserted a hack to use the variables >>>> into >>>> > rtmp.c and mod_rtmp.c, but wondering what the right way to fix this >>>> is. >>>> > Thanks, >>>> > 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/185a0c5b/attachment.html From brian at freeswitch.org Wed Nov 9 21:21:24 2011 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Nov 2011 12:21:24 -0600 Subject: [Freeswitch-users] Anyone using t38modem with FS (gatewaying from audio<>t38) and Hylafax successfully? In-Reply-To: <4EBAC3D5.5020407@integrafin.co.uk> References: <4EBAC3D5.5020407@integrafin.co.uk> Message-ID: <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> You're missing the ever critical t38_passthru=true variable remove all else about t38 since those are valid for termination/originate with FS itself... don't need t38_gateway at all.... then don't worry about the rest. /b On Nov 9, 2011, at 12:17 PM, Alex Crow wrote: > > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/c5b0a78f/attachment.html From acrow at integrafin.co.uk Wed Nov 9 22:52:30 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 09 Nov 2011 19:52:30 +0000 Subject: [Freeswitch-users] Anyone using t38modem with FS (gatewaying from audio<>t38) and Hylafax successfully? In-Reply-To: <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> Message-ID: <4EBAD9FE.3070401@integrafin.co.uk> On 09/11/11 18:21, Brian West wrote: > You're missing the ever critical t38_passthru=true variable remove all > else about t38 since those are valid for termination/originate with FS > itself... don't need t38_gateway at all.... then don't worry about the > rest. > > /b Brian, Really? Even if the incoming call is not a t38 but just PCMU over RTP? If this works I'll pay the wiki penalty! Cheers Alex From jerry.richards at teotech.com Wed Nov 9 23:35:38 2011 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 9 Nov 2011 12:35:38 -0800 Subject: [Freeswitch-users] Holding Call With Bypass Media True Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6963278B3E607@VA3DIAXVS351.RED001.local> Hi All, I am seeing an issue where Freeswitch is disconnecting a call when one endpoint tries to put the call on-hold (inbound-bypass-media is true and one phone has a secondary extension of the other phone). I'm just wondering if there is a rule/restriction for bypass media I'm not aware of. If I change the configuration to inbound-proxy-media true then the call is put on-hold correctly. If I set it back to inbound-bypass-media true and then remove the secondary extension from Phone A (below), then the call is put on-hold correctly. Here is the scenario: 1) Phone A registers primary extension 1002 and secondary extension 1003 2) Phone B registers only primary extension 1003 3) A calls B 4) B answers 5) A holds call 6) Freeswitch sends BYE to both A and B with reason DESTINATION_OUT_OF_ORDER (Also it says: "Re-INVITE to a no-media channel that is not in a bridge") Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/d01e3686/attachment.html From acrow at integrafin.co.uk Thu Nov 10 00:29:27 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 09 Nov 2011 21:29:27 +0000 Subject: [Freeswitch-users] Anyone using t38modem with FS (gatewaying from audio<>t38) and Hylafax successfully? In-Reply-To: <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> Message-ID: <4EBAF0B7.6030703@integrafin.co.uk> On 09/11/11 18:21, Brian West wrote: > You're missing the ever critical t38_passthru=true variable remove all > else about t38 since those are valid for termination/originate with FS > itself... don't need t38_gateway at all.... then don't worry about the > rest. > > /b > > On Nov 9, 2011, at 12:17 PM, Alex Crow wrote: > >> >> >> >> >> >> >> >> >> >> >> > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian, I am afraid is is still not working. Freeswitch seems to abort the call with these logs: 2011-11-09 21:16:08.468151 [DEBUG] switch_core_session.c:788 Send signal sofia/internal/5470 at 10.10.0.2 [BREAK] 2011-11-09 21:16:08.488151 [DEBUG] sofia_glue.c:166 sofia/internal/5470 at 10.10.0.2 image media sdp: v=0 o=FreeSWITCH 1320855245 1320855247 IN IP4 192.168.20.245 s=FreeSWITCH c=IN IP4 192.168.20.245 t=0 0 m=image 18122 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:500 a=T38FaxMaxDatagram:500 a=T38FaxUdpEC:t38UDPRedundancy 2011-11-09 21:16:08.488151 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/5470 at 10.10.0.2 [BREAK] 2011-11-09 21:16:08.488151 [DEBUG] sofia.c:5283 Channel sofia/internal/5470 at 10.10.0.2 entering state [calling][0] 2011-11-09 21:16:08.508150 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/5470 at 10.10.0.2 [BREAK] 2011-11-09 21:16:08.508150 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/5470 at 10.10.0.2 [BREAK] 2011-11-09 21:16:08.508150 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/5470 at 10.10.0.2 [BREAK] 2011-11-09 21:16:08.508150 [DEBUG] sofia.c:4868 Passing 415 Unsupported Media Type to other leg 2011-11-09 21:16:08.508150 [DEBUG] switch_core_session.c:788 Send signal sofia/internal/sip:t38modem at 192.168.20.245:6060 [BREAK] 2011-11-09 21:16:08.508150 [DEBUG] sofia.c:5283 Channel sofia/internal/5470 at 10.10.0.2 entering state [ready][415] 2011-11-09 21:16:08.508150 [DEBUG] sofia_glue.c:4730 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2011-11-09 21:16:08.508150 [DEBUG] sofia_glue.c:2782 Already using PCMA 2011-11-09 21:16:08.508150 [DEBUG] sofia_glue.c:4851 Set 2833 dtmf send/recv payload to 101 2011-11-09 21:16:08.528161 [DEBUG] mod_sofia.c:2298 Responding with 415 [Unsupported Media Type] 2011-11-09 21:16:08.528161 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/sip:t38modem at 192.168.20.245:6060 [BREAK] 2011-11-09 21:16:08.528161 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/sip:t38modem at 192.168.20.245:6060 [BREAK] 2011-11-09 21:16:08.528161 [DEBUG] sofia.c:5283 Channel sofia/internal/sip:t38modem at 192.168.20.245:6060 entering state [ready][415] 2011-11-09 21:16:08.528161 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/sip:t38modem at 192.168.20.245:6060 [BREAK] 2011-11-09 21:17:01.788147 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/5470 at 10.10.0.2 [BREAK] 2011-11-09 21:17:01.808150 [DEBUG] switch_channel.c:2826 (sofia/internal/5470 at 10.10.0.2) Callstate Change ACTIVE -> HANGUP Any ideas to get around this would be most gratefully received. Best regards, Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/b3a1d988/attachment-0001.html From anthony.minessale at gmail.com Thu Nov 10 00:33:53 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Nov 2011 15:33:53 -0600 Subject: [Freeswitch-users] Holding Call With Bypass Media True In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6963278B3E607@VA3DIAXVS351.RED001.local> References: <2BF7FB90DF25EA4485949F3AF2B9D6963278B3E607@VA3DIAXVS351.RED001.local> Message-ID: It might be some nat alg stuff confusing the legs coming from the same phone. On Wed, Nov 9, 2011 at 2:35 PM, Jerry Richards wrote: > Hi All, > I am seeing an issue where Freeswitch is disconnecting a call when one > endpoint tries to put the call on-hold (inbound-bypass-media is true and > one phone has a secondary extension of the other phone). I'm just > wondering if there is a rule/restriction for bypass media I'm not aware of. > > If I change the configuration to inbound-proxy-media true then the call is > put on-hold correctly. If I set it back to inbound-bypass-media true and > then remove the secondary extension from Phone A (below), then the call is > put on-hold correctly. > > Here is the scenario: > 1) Phone A registers primary extension 1002 and secondary extension 1003 > 2) Phone B registers only primary extension 1003 > 3) A calls B > 4) B answers > 5) A holds call > 6) Freeswitch sends BYE to both A and B with reason > DESTINATION_OUT_OF_ORDER > (Also it says: "Re-INVITE to a no-media channel that is not in a > bridge") > > Thanks, > 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/b6b10e1b/attachment.html From brian at freeswitch.org Thu Nov 10 00:39:52 2011 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Nov 2011 15:39:52 -0600 Subject: [Freeswitch-users] Anyone using t38modem with FS (gatewaying from audio<>t38) and Hylafax successfully? In-Reply-To: <4EBAF0B7.6030703@integrafin.co.uk> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> Message-ID: <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> That kinda says it all. /b On Nov 9, 2011, at 3:29 PM, Alex Crow wrote: > 2011-11-09 21:16:08.528161 [DEBUG] mod_sofia.c:2298 Responding with 415 [Unsupported Media Type] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/dd505b62/attachment.html From sdame at 207me.com Thu Nov 10 01:27:55 2011 From: sdame at 207me.com (Stephen Dame) Date: Wed, 9 Nov 2011 17:27:55 -0500 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: <002701cc9f2e$d47dfbc0$7d79f340$@com> Michael, Is there a public archive of recordings for the weekly conference calls? I attended some this summer and learned a lot about the project, but work schedule prevents attendance now. Thanks Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, November 09, 2011 11:37 AM To: freeswitch-users at lists.freeswitch.org; freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Hello FreeSWITCHers! Here's today's agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_09 We have a few miscellaneous items to discuss and I have some updates. Also, I'm hoping to continue the discussion on SIP 101 that we started last week. I think it would be a good time to go over the Contact: header and what it's for. That will set the ground work for what the NDLB-connectile-dysfunction parameter does. Talk to you soon, -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/2ae69be8/attachment.html From avi at avimarcus.net Thu Nov 10 01:41:02 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 10 Nov 2011 00:41:02 +0200 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: <002701cc9f2e$d47dfbc0$7d79f340$@com> References: <002701cc9f2e$d47dfbc0$7d79f340$@com> Message-ID: The link to the recording goes on: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call This weeks doesn't seem to be up yet. -Avi On Thu, Nov 10, 2011 at 12:27 AM, Stephen Dame wrote: > Michael,**** > > ** ** > > Is there a public archive of recordings for the weekly conference calls? I > attended some this summer and learned a lot about the project, but work > schedule prevents attendance now.**** > > ** ** > > Thanks**** > > Stephen**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Wednesday, November 09, 2011 11:37 AM > *To:* freeswitch-users at lists.freeswitch.org; > freeswitch-dev at lists.freeswitch.org > *Subject:* [Freeswitch-users] FreeSWITCH Conference Call Today**** > > ** ** > > Hello FreeSWITCHers!**** > > ** ** > > Here's today's agenda:**** > > http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_09**** > > ** ** > > We have a few miscellaneous items to discuss and I have some updates. > Also, I'm hoping to continue the discussion on SIP 101 that we started last > week. I think it would be a good time to go over the Contact: header and > what it's for. That will set the ground work for what the > NDLB-connectile-dysfunction parameter does.**** > > ** ** > > Talk to you soon,**** > > -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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/28349b4e/attachment.html From sdame at 207me.com Thu Nov 10 01:50:00 2011 From: sdame at 207me.com (Stephen Dame) Date: Wed, 9 Nov 2011 17:50:00 -0500 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: <002701cc9f2e$d47dfbc0$7d79f340$@com> References: <002701cc9f2e$d47dfbc0$7d79f340$@com> Message-ID: <003801cc9f31$ea578580$bf069080$@com> Never mind, I just checked and found them. http://wiki.freeswitch.org/wiki/Weekly_Conference_Call Thanks stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stephen Dame Sent: Wednesday, November 09, 2011 5:28 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] FreeSWITCH Conference Call Today Michael, Is there a public archive of recordings for the weekly conference calls? I attended some this summer and learned a lot about the project, but work schedule prevents attendance now. Thanks Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, November 09, 2011 11:37 AM To: freeswitch-users at lists.freeswitch.org; freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Hello FreeSWITCHers! Here's today's agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_09 We have a few miscellaneous items to discuss and I have some updates. Also, I'm hoping to continue the discussion on SIP 101 that we started last week. I think it would be a good time to go over the Contact: header and what it's for. That will set the ground work for what the NDLB-connectile-dysfunction parameter does. Talk to you soon, -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/12517874/attachment-0001.html From admin at blindi.net Thu Nov 10 02:41:39 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Thu, 10 Nov 2011 00:41:39 +0100 (CET) Subject: [Freeswitch-users] I missing conference moderator features In-Reply-To: References: Message-ID: Hi Michael, thanks for your Reply. Although I've read a lot of lua. But i do not found a channel search funtion. I can only find the application eavesdrop the similar feature. For example to select a channel listen only (*key). The problem is that there seems to be no functions to define keys assignments. Or to store the current channel in a variable. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From vetali100 at gmail.com Thu Nov 10 03:21:12 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Wed, 9 Nov 2011 16:21:12 -0800 Subject: [Freeswitch-users] make current fails with 'config.sub not found' Message-ID: Today decided to upgrade FS, after almost 5 months from last upgrade, and executed 'make current' However, it failed with the following errors: ... create mode 100644 src/mod/event_handlers/mod_cdr_mongodb/mod_cdr_mongodb.c delete mode 100644 src/mod/timers/.empty create mode 100644 swig_common.i make[1]: Leaving directory `/usr/local/src/freeswitch' make all make[1]: Entering directory `/usr/local/src/freeswitch' cd . && /bin/sh /usr/local/src/freeswitch/build/config/missing --run aclocal-1.9 cd . && /bin/sh /usr/local/src/freeswitch/build/config/missing --run automake-1.9 --gnu *configure.in:27: required file `build/config/config.sub' not found* make[1]: *** [Makefile.in] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 Could you please hint what has changed since then or what I am doing wrong? Thank you, Vitalie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/aaa3d8be/attachment.html From krice at freeswitch.org Thu Nov 10 03:42:35 2011 From: krice at freeswitch.org (Ken Rice) Date: Wed, 09 Nov 2011 18:42:35 -0600 Subject: [Freeswitch-users] make current fails with 'config.sub not found' In-Reply-To: Message-ID: You need to reboostrap... Some stuff in the build system was updated to remove generated files from tree... K On 11/9/11 6:21 PM, "Vitalie Colosov" wrote: > Today decided to upgrade FS, after almost 5 months from last upgrade, and > executed 'make current' > > However, it failed with the following errors: > > ... > ?create mode 100644 src/mod/event_handlers/mod_cdr_mongodb/mod_cdr_mongodb.c > ?delete mode 100644 src/mod/timers/.empty > ?create mode 100644 swig_common.i > make[1]: Leaving directory `/usr/local/src/freeswitch' > make all > make[1]: Entering directory `/usr/local/src/freeswitch' > cd . && /bin/sh /usr/local/src/freeswitch/build/config/missing --run > aclocal-1.9 > ?cd . && /bin/sh /usr/local/src/freeswitch/build/config/missing --run > automake-1.9 --gnu > configure.in:27 : required file > `build/config/config.sub' not found > make[1]: *** [Makefile.in] Error 1 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > > Could you please hint what has changed since then or what I am doing wrong? > > Thank you, > Vitalie > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/8bacf9dc/attachment.html From jeff at jefflenk.com Thu Nov 10 03:53:14 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 9 Nov 2011 16:53:14 -0800 (PST) Subject: [Freeswitch-users] make current fails with 'config.sub not found' In-Reply-To: References: Message-ID: <1320886393993-6980176.post@n2.nabble.com> see http://freeswitch-users.2379917.n2.nabble.com/Heads-up-autogenerated-file-quot-build-config-config-sub-quot-removed-td6964103.html -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/make-current-fails-with-config-sub-not-found-tp6980116p6980176.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vetali100 at gmail.com Thu Nov 10 04:07:39 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Wed, 9 Nov 2011 17:07:39 -0800 Subject: [Freeswitch-users] make current fails with 'config.sub not found' In-Reply-To: <1320886393993-6980176.post@n2.nabble.com> References: <1320886393993-6980176.post@n2.nabble.com> Message-ID: Thank you, this fixed my problem (automake --add-missing) Somehow I missed that notification. Thanks, Vitalie 2011/11/9 Jeff Lenk > see > > http://freeswitch-users.2379917.n2.nabble.com/Heads-up-autogenerated-file-quot-build-config-config-sub-quot-removed-td6964103.html > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/make-current-fails-with-config-sub-not-found-tp6980116p6980176.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/20111109/c1d0fff1/attachment.html From vetali100 at gmail.com Thu Nov 10 08:13:59 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Wed, 9 Nov 2011 21:13:59 -0800 Subject: [Freeswitch-users] new git - no more 'SIP auth challenge' messages in log Message-ID: Hi List, After upgrading to the latest git, I don't see anymore "SIP auth challenge" messages in the log for normal register attempts. Is there any new switch to enable these message? Log verbosity is INFO and Thanks, Vitalie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111109/f2e685ad/attachment.html From acrow at integrafin.co.uk Thu Nov 10 10:28:23 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 10 Nov 2011 07:28:23 +0000 Subject: [Freeswitch-users] Anyone using t38modem with FS (gatewaying from audio<>t38) and Hylafax successfully? In-Reply-To: <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> Message-ID: <4EBB7D17.9080208@integrafin.co.uk> Brian, Can you elaborate? It seems that the call is PCMA, as far as I can tell t38modem supports PCMA. What can I do to fix this? I thought this would work with audio faxing? Or am I completely wrong and I can only do t38 using t38modemso I do need some kind of gateway either on or outside of FS to get standard fax working? This was why I had the gateway stuff in the dial plan in the first place. Alex On 09/11/11 21:39, Brian West wrote: > That kinda says it all. > > /b > > On Nov 9, 2011, at 3:29 PM, Alex Crow wrote: > >> 2011-11-09 21:16:08.528161 [DEBUG] mod_sofia.c:2298 Responding with >> 415 [Unsupported Media Type] > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/646b8fe0/attachment-0001.html From miha at softnet.si Thu Nov 10 12:59:08 2011 From: miha at softnet.si (Miha Zoubek) Date: Thu, 10 Nov 2011 10:59:08 +0100 Subject: [Freeswitch-users] freeswitch radiusclinet (mod_radius_cdr) problems In-Reply-To: <4EBA8B5E.2050900@softnet.si> References: <4EBA8B5E.2050900@softnet.si> Message-ID: <4EBBA06C.206@softnet.si> Hi, does anybody know why NAS (freeswitch) is behaving like that? All my entries are duplicated in sql tables. If do not pick up call, I get tree entries for every call that is made (I should get nothing if call is not pickup). Thank you! BR, miha figure out why the entries are duplicate (they're probably*not* duplicate), and figure out what fields make up the "same" session On 11/9/2011 3:17 PM, Miha Zoubek wrote: > Hi, > > radiusclient on freeswitch in sending more that once same response. So I > am getting in my sql tables for every call two inputs which is wrong. > How can I deal whit this issue? > > I have paste a log from freeradius server in pastebin that you can see > what is freeswitch sending. > > http://pastebin.freeswitch.org/17730 > > > Thank you! > > BR, > Miha > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/c8e0290e/attachment.html From mitja.thomas1 at ewetel.de Thu Nov 10 13:35:38 2011 From: mitja.thomas1 at ewetel.de (Mitja Thomas) Date: Thu, 10 Nov 2011 11:35:38 +0100 Subject: [Freeswitch-users] I missing conference moderator features In-Reply-To: References: Message-ID: <4EBBA8FA.4020206@ewetel.de> Hello Thomas, I guess what Micheal means is, that there isn't a complete, ready-to-use feature like the one you're asking for. But you could write I lua script that retreives information about the channels of a conference and work with these information according to previously set bindings (bind_digit_action). What I would do (very short, very superficial): 1) Bind digits to Dialplan Actions and let these call lua Scripts. 2) In these lua scripts retrieve information about a given conference with api:execute("conference", ""..conference_id.." xml_list") 3) In one of these actions you select a channel (by skipping through the possible channels probably) and save the current_channel in a channel_variable in the moderator channel: session:setVariable("x_current_channel", curent_channel_uuid) 4) Perform Actions on that channel which you could identify by the uuid you stored in your channel variable. All that should be possible with the given FreeSWITCH api + lua (although I clearly didnt just write a script and tested it ;) ). Regards, Mitja Am 20:59, schrieb Thomas Hoellriegel: > Hi Michael, > thanks for your Reply. > Although I've read a lot of lua. > But i do not found a channel search funtion. > I can only find the application eavesdrop the similar feature. > For example to select a channel listen only (*key). > The problem is that there seems to be no functions to define keys > assignments. > Or to store the current channel in a variable. > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc From acrow at integrafin.co.uk Thu Nov 10 16:13:32 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 10 Nov 2011 13:13:32 +0000 Subject: [Freeswitch-users] Anyone using t38modem with FS (gatewaying from audio<>t38) and Hylafax successfully? In-Reply-To: <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> Message-ID: <4EBBCDFC.4020003@integrafin.co.uk> On 09/11/11 21:39, Brian West wrote: > That kinda says it all. > > /b > > On Nov 9, 2011, at 3:29 PM, Alex Crow wrote: > >> 2011-11-09 21:16:08.528161 [DEBUG] mod_sofia.c:2298 Responding with >> 415 [Unsupported Media Type] > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Replying to myself, I got it in the end, all I needed was this, which is already in the wiki more or less: That worked fine and Hylafax received the pages perfectly! Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/e57a9558/attachment.html From steveu at coppice.org Thu Nov 10 17:10:03 2011 From: steveu at coppice.org (Steve Underwood) Date: Thu, 10 Nov 2011 22:10:03 +0800 Subject: [Freeswitch-users] Anyone using t38modem with FS (gatewaying from audio<>t38) and Hylafax successfully? In-Reply-To: <4EBB7D17.9080208@integrafin.co.uk> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> Message-ID: <4EBBDB3B.2070509@coppice.org> On 11/10/2011 03:28 PM, Alex Crow wrote: > Brian, > > Can you elaborate? It seems that the call is PCMA, as far as I can > tell t38modem supports PCMA. What can I do to fix this? I thought this > would work with audio faxing? > > Or am I completely wrong and I can only do t38 using t38modemso I do > need some kind of gateway either on or outside of FS to get standard > fax working? This was why I had the gateway stuff in the dial plan in > the first place. > If t38modem supports audio FAXing, its a new feature. Certainly until recently it was a pure T.38 package. Steve From cliff at develix.com Thu Nov 10 18:58:59 2011 From: cliff at develix.com (Cliff Wells) Date: Thu, 10 Nov 2011 07:58:59 -0800 Subject: [Freeswitch-users] High load on database server In-Reply-To: <1319567530.2234.16.camel@portable-evil> References: <4EA5DA64.1060503@communicatefreely.net> <1319494283.4593.160.camel@portable-evil> <543606611-1319565003-cardhu_decombobulator_blackberry.rim.net-1845868528-@b13.c27.bise6.blackberry> <1319567530.2234.16.camel@portable-evil> Message-ID: <1320940739.12168.26.camel@portable-evil> On Tue, 2011-10-25 at 11:32 -0700, Cliff Wells wrote: > On Tue, 2011-10-25 at 17:50 +0000, Luis Jimenez wrote: > > Hi Anthony, which db engine do you prefer, this as recommendation for us, for better performance with FS? > > We use PostgreSQL 8.4. Our system currently processes around 350 > concurrent calls during peak hours and PostgreSQL barely notices (every > call accesses the database to get the dialplan for around 5000 numbers). > We also access the same database for CDR logging and > application-specific data, so the number of concurrent database > connections is actually somewhat higher than that. > > That being said, unixODBC 2.2 will often segfault under that load, so be > wary of that (it appears reliable up to about 300 connections, seems to > reliably fall over at 330). We're in the process of testing 2.3.0 which > is reportedly better in this regard. Just a short follow up on the stability of unixODBC 2.3.0 vs 2.2.x. We consistently saw segmentation faults at 330+ concurrent calls with 2.2.x. This morning I observed one of those sames systems handling over 365 concurrent calls with 2.3.0 without crashing. Nothing else has changed. Regards, Cliff From Stefan.Weigel at allianz-warranty.com Thu Nov 10 19:53:40 2011 From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan) Date: Thu, 10 Nov 2011 17:53:40 +0100 Subject: [Freeswitch-users] fwd: call groups - sort order Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36FBB@AZWSMS03.azwarranty.int> Hi list, sorry for pushing this one, but I would really appreciate any hints on this... Thanks and best regards, Stefan Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Weigel, Stefan Gesendet: Mittwoch, 9. November 2011 09:59 An: FreeSWITCH Users Help Betreff: [Freeswitch-users] call groups - sort order Hi list, from the WIKI page: group call:groupname[:order] How are the entries sorted ? How can I make sure the list returned by 'group call' is sorted in a certain order ? I want to make sure a certain extension is ringed first, but sometimes another extension is listed first. Thanks and best regards Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/d6058783/attachment-0001.html From lloyd.aloysius at sunteltech.ca Thu Nov 10 19:57:34 2011 From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius) Date: Thu, 10 Nov 2011 11:57:34 -0500 Subject: [Freeswitch-users] Voice mail migration In-Reply-To: References: Message-ID: Thanks Lloyd * * * * On Wed, Nov 9, 2011 at 11:11 AM, Chris Chen wrote: > Hi Lloyd, just do the following step, you will be where you want to be > > *git checkout 0675b59* > Note: checking out '0675b59'. > > You are in 'detached HEAD' state. You can look around, make experimental > changes and commit them, and you can discard any commits you make in this > state without impacting any branches by performing another checkout. > > If you want to create a new branch to retain commits you create, you may > do so (now or later) by using -b with the checkout command again. Example: > > git checkout -b new_branch_name > > HEAD is now at 0675b59... FS-3321 release rwlock on error > > > On Wed, Nov 9, 2011 at 11:01 AM, Lloyd Aloysius wrote: > >> I am not looking for a new version. I am looking for a old version FreeSWITCH >> Version 1.0.head *(git-0675b59 2011-06-06 21-28-14 -0500)* >> * >> * >> How to use git to get this version into my server. I am not a git expert. >> >> Any help is appreciated. >> >> Thanks >> Lloyd >> >> >> >> On Tue, Nov 8, 2011 at 10:59 PM, curriegrad2004 > > wrote: >> >>> Run a "make current". That should do the git pull and update the build >>> to the latest git for you. >>> >>> On Tue, Nov 8, 2011 at 7:28 PM, Lloyd Aloysius >>> wrote: >>> > How to get the FreeSWITCH git for the following version >>> > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) >>> > Thanks >>> > Lloyd >>> > >>> > >>> > >>> > >>> > >>> > >>> > On Tue, Nov 8, 2011 at 8:52 PM, curriegrad2004 < >>> curriegrad2004 at gmail.com> >>> > wrote: >>> >> >>> >> There shouldn't be any surprises then. The DB structure from july till >>> >> now for mod_voicemail hasn't changed that much anyways. However the >>> >> core_db did change slightly, there is the new basic_call table at the >>> >> core db which replaced another table I can't remember off the top of >>> >> my head atm. >>> >> >>> >> On Tue, Nov 8, 2011 at 5:21 PM, Lloyd Aloysius >>> >> wrote: >>> >> > Thank you for the reply. Old server running >>> >> > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) >>> >> > New Server ... Planning to run from the git. >>> >> > >>> >> > >>> >> > Thanks and regards, >>> >> > >>> >> > Lloyd >>> >> > >>> >> > >>> >> > On Tue, Nov 8, 2011 at 7:43 PM, Michael Collins >> > >>> >> > wrote: >>> >> >> >>> >> >> unless the previous server is really old, I doubt there will be any >>> >> >> trouble migrating. I would run a test and see how it goes. >>> >> >> -MC >>> >> >> >>> >> >> On Tue, Nov 8, 2011 at 4:39 PM, Lloyd Aloysius >>> >> >> >>> >> >> wrote: >>> >> >>> >>> >> >>> Micheal, >>> >> >>> Thank you for the reply. Yes same domain . But New server have a >>> new >>> >> >>> version of FreeSWITCH. >>> >> >>> Do I need to maintain the same version FreeSWITCH for my new >>> server. >>> >> >>> >>> >> >>> Thanks and regards, >>> >> >>> >>> >> >>> Lloyd >>> >> >>> >>> >> >>> >>> >> >>> On Tue, Nov 8, 2011 at 7:32 PM, Michael Collins < >>> msc at freeswitch.org> >>> >> >>> wrote: >>> >> >>>> >>> >> >>>> Is your domain the same? Basically you have the voicemail.db >>> file and >>> >> >>>> you have /usr/local/freeswitch/storage/voicemail/default/xxxx >>> where >>> >> >>>> xxx is >>> >> >>>> the domain. Shouldn't be too hard to copy everything over. >>> >> >>>> -MC >>> >> >>>> >>> >> >>>> On Tue, Nov 8, 2011 at 4:05 PM, Lloyd Aloysius >>> >> >>>> wrote: >>> >> >>>>> >>> >> >>>>> HI, >>> >> >>>>> I am in the process of moving my current freeswitch server to a >>> new >>> >> >>>>> server. What is the best way to migrate the VoiceMail from old >>> to >>> >> >>>>> new >>> >> >>>>> server? >>> >> >>>>> Any help is appreciated. >>> >> >>>>> >>> >> >>>>> Thanks and regards, >>> >> >>>>> >>> >> >>>>> Lloyd >>> >> >>>>> >>> >> >>>>> FreeSWITCH-users mailing list >>> >> >>>>> FreeSWITCH-users at lists.freeswitch.org >>> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>>>> >>> >> >>>>> >>> >> >>>>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>>>> http://www.freeswitch.org >>> >> >>>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> FreeSWITCH-users mailing list >>> >> >>>> FreeSWITCH-users at lists.freeswitch.org >>> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>>> >>> >> >>>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>>> http://www.freeswitch.org >>> >> >>>> >>> >> >>> >>> >> >>> >>> >> >>> >>> >> >>> FreeSWITCH-users mailing list >>> >> >>> FreeSWITCH-users at lists.freeswitch.org >>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >>> >> >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> http://www.freeswitch.org >>> >> >>> >>> >> >> >>> >> >> >>> >> >> >>> >> >> FreeSWITCH-users mailing list >>> >> >> FreeSWITCH-users at lists.freeswitch.org >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >> >>> >> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >> http://www.freeswitch.org >>> >> >> >>> >> > >>> >> > >>> >> > >>> >> > FreeSWITCH-users mailing list >>> >> > FreeSWITCH-users at lists.freeswitch.org >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> > http://www.freeswitch.org >>> >> > >>> >> > >>> >> >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/49ea7d82/attachment.html From jeff at jefflenk.com Thu Nov 10 20:13:54 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Thu, 10 Nov 2011 09:13:54 -0800 (PST) Subject: [Freeswitch-users] new git - no more 'SIP auth challenge' messages in log In-Reply-To: References: Message-ID: <1320945234630-6982405.post@n2.nabble.com> Please see http://jira.freeswitch.org/browse/FS-3094 for a discussion on this. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/new-git-no-more-SIP-auth-challenge-messages-in-log-tp6980582p6982405.html Sent from the freeswitch-users mailing list archive at Nabble.com. From oseslija at gmail.com Thu Nov 10 20:24:34 2011 From: oseslija at gmail.com (Ognjen Seslija) Date: Thu, 10 Nov 2011 18:24:34 +0100 Subject: [Freeswitch-users] freeswitch radiusclinet (mod_radius_cdr) problems In-Reply-To: <4EBA8B5E.2050900@softnet.si> References: <4EBA8B5E.2050900@softnet.si> Message-ID: Radius clients generally send Start and Stop records, I guess this is why you get two records. On Wed, Nov 9, 2011 at 3:17 PM, Miha Zoubek wrote: > Hi, > > radiusclient on freeswitch in sending more that once same response. So I > am getting in my sql tables for every call two inputs which is wrong. > How can I deal whit this issue? > > I have paste a log from freeradius server in pastebin that you can see > what is freeswitch sending. > > http://pastebin.freeswitch.org/17730 > > > Thank you! > > BR, > Miha > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/4da2429d/attachment-0001.html From turqmr2 at gmail.com Thu Nov 10 20:34:38 2011 From: turqmr2 at gmail.com (Jacob Smith) Date: Thu, 10 Nov 2011 12:34:38 -0500 Subject: [Freeswitch-users] Help with home setup In-Reply-To: References: <4EB9288F.6050705@gmail.com> Message-ID: <4EBC0B2E.6090108@gmail.com> Thanks for pointing me in the right direction. I now realize that Freeswitch was never running. So when you stop laughing and/or shaking your head... When I try to start freeswitch I get this error: /libexec/ld-elf.so.1: Shared object "libstdc++.so.6" not found, required by "libfreeswitch.so.1" On 11/08/2011 11:29 AM, Avi Marcus wrote: > Most basic debug: open the fs_cli (/usr/local/freeswitch/bin/fs_cli) > and see if you see the register attempts. > If not, then there's an IP or firewall issue (perhaps your FS is > listening only on a public IP or the like) > Or, it will tell you the authentication error. > > -Avi > > > On Tue, Nov 8, 2011 at 3:03 PM, Jacob Smith > wrote: > > I have been trying to get this working for the last week and would > really appreciate your help. > > I have a pfSense 2.0 server with Freeswitch and FusionPBX installed. I > followed the instructions at > http://wiki.fusionpbx.com/index.php?title=PfSense_Install and can > access > the web GUI. > > My goal is to use a Gigaset C610A-IP connected to my Google Voice > account. However, my first failure seems to be the most basic: through > FusionPBX, I created a user and extension to use with the phone but I > can not register it with FreeSwitch. So, how do I get the phone to > work > so I can test that the basic functions of FreeSwitch are working and > move on? > > 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/20111110/33f1d66f/attachment.html From covici at ccs.covici.com Thu Nov 10 21:09:27 2011 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 10 Nov 2011 13:09:27 -0500 Subject: [Freeswitch-users] new git - no more 'SIP auth challenge' messages in log In-Reply-To: <1320945234630-6982405.post@n2.nabble.com> References: <1320945234630-6982405.post@n2.nabble.com> Message-ID: <18097.1320948567@ccs.covici.com> So what is the parameter to put those messages back to warning from DEBUG10? Jeff Lenk wrote: > Please see http://jira.freeswitch.org/browse/FS-3094 > > for a discussion on this. > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/new-git-no-more-SIP-auth-challenge-messages-in-log-tp6980582p6982405.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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From Hector.Geraldino at ip-soft.net Thu Nov 10 21:14:28 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Thu, 10 Nov 2011 13:14:28 -0500 Subject: [Freeswitch-users] reporting bug for freeswitch-contrib project Message-ID: <6A6B4C284AD15042B429EB9D904544AD0224D44294@NY1-EXMB-01.ip-soft.net> Hi, What's the best way to report a bug (and a patch) for a project that is part of the freeswitch-contrib repository? I can't find any category on JIRA for those projects. Should I send the bug description and the proposed patch to the freeswitch-dev list, or is there a better way? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/f4f79187/attachment.html From jaknap34 at gmail.com Thu Nov 10 10:19:14 2011 From: jaknap34 at gmail.com (Pankaj Upadhyay) Date: Thu, 10 Nov 2011 12:49:14 +0530 Subject: [Freeswitch-users] Patch_Up_Two_Files Message-ID: Hi all , -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/78c36d1f/attachment.html From msc at freeswitch.org Thu Nov 10 21:24:14 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Nov 2011 10:24:14 -0800 Subject: [Freeswitch-users] call groups - sort order In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36FA8@AZWSMS03.azwarranty.int> References: <5003D7D3E06F514E8C682F18D223265C04D3B36FA8@AZWSMS03.azwarranty.int> Message-ID: Which wiki page is this? -MC On Wed, Nov 9, 2011 at 12:58 AM, Weigel, Stefan < Stefan.Weigel at allianz-warranty.com> wrote: > Hi list,**** > > ** ** > > from the WIKI page:**** > > ** ** > > group call:groupname[:order]**** > > ** ** > > How are the entries sorted ? How can I make sure the list returned by > ?group call? is sorted in a certain order ? I want to make sure a certain > extension is ringed first, but sometimes another extension is listed first. > **** > > ** ** > > ** ** > > Thanks and best regards**** > > ** ** > > Stefan**** > > ** ** > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/d53cf2f3/attachment.html From hynek.cihlar at gmail.com Thu Nov 10 21:28:18 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Thu, 10 Nov 2011 19:28:18 +0100 Subject: [Freeswitch-users] Playlist playback Message-ID: <2968466961287872078@unknownmsgid> Does Freeswitch support playing back of multiple audio files? I.e. issuing one command with multiple files? I am trying to build dynamic dialogs over event socket and issuing call for every part of the dialog seems as a waste. Sent from my mobile device From msc at freeswitch.org Thu Nov 10 21:28:29 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Nov 2011 10:28:29 -0800 Subject: [Freeswitch-users] reporting bug for freeswitch-contrib project In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0224D44294@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD0224D44294@NY1-EXMB-01.ip-soft.net> Message-ID: Which user in the contrib are you targeting? We might be able to ping him/her. -MC On Thu, Nov 10, 2011 at 10:14 AM, Hector Geraldino < Hector.Geraldino at ip-soft.net> wrote: > Hi,**** > > ** ** > > What?s the best way to report a bug (and a patch) for a project that is > part of the freeswitch-contrib repository? I can?t find any category on > JIRA for those projects. Should I send the bug description and the proposed > patch to the freeswitch-dev list, or is there a better way?**** > > ** ** > > 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/20111110/1da4719d/attachment-0001.html From msc at freeswitch.org Thu Nov 10 21:32:18 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Nov 2011 10:32:18 -0800 Subject: [Freeswitch-users] Playlist playback In-Reply-To: <2968466961287872078@unknownmsgid> References: <2968466961287872078@unknownmsgid> Message-ID: Yep, there are several ways. Check this out: http://wiki.freeswitch.org/wiki/Mod_file_string Essentially you can do a !-delimited list of files. Also, don't forget about the phrase macros, which allow you to get even more creative with playing multiple files... -MC On Thu, Nov 10, 2011 at 10:28 AM, Hynek Cihlar wrote: > Does Freeswitch support playing back of multiple audio files? I.e. > issuing one command with multiple files? > > I am trying to build dynamic dialogs over event socket and issuing > call for every part of the dialog seems as a waste. > > > Sent from my mobile device > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/d698e575/attachment.html From Hector.Geraldino at ip-soft.net Thu Nov 10 21:49:39 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Thu, 10 Nov 2011 13:49:39 -0500 Subject: [Freeswitch-users] reporting bug for freeswitch-contrib project In-Reply-To: References: <6A6B4C284AD15042B429EB9D904544AD0224D44294@NY1-EXMB-01.ip-soft.net> Message-ID: <6A6B4C284AD15042B429EB9D904544AD0224D442AB@NY1-EXMB-01.ip-soft.net> David Varnes/ java org.freeswitch.esl.client project From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, November 10, 2011 1:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] reporting bug for freeswitch-contrib project Which user in the contrib are you targeting? We might be able to ping him/her. -MC On Thu, Nov 10, 2011 at 10:14 AM, Hector Geraldino > wrote: Hi, What's the best way to report a bug (and a patch) for a project that is part of the freeswitch-contrib repository? I can't find any category on JIRA for those projects. Should I send the bug description and the proposed patch to the freeswitch-dev list, or is there a better way? 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/20111110/f10b0679/attachment.html From msc at freeswitch.org Thu Nov 10 22:14:50 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Nov 2011 11:14:50 -0800 Subject: [Freeswitch-users] reporting bug for freeswitch-contrib project In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0224D442AB@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD0224D44294@NY1-EXMB-01.ip-soft.net> <6A6B4C284AD15042B429EB9D904544AD0224D442AB@NY1-EXMB-01.ip-soft.net> Message-ID: Ah, I'm not familiar with him. If his email is in there somewhere then you should probably just contact him directly. You won't find very many core FreeSWITCHers who are too keen on Java. :P -MC On Thu, Nov 10, 2011 at 10:49 AM, Hector Geraldino < Hector.Geraldino at ip-soft.net> wrote: > David Varnes/ java org.freeswitch.esl.client project**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Thursday, November 10, 2011 1:28 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] reporting bug for freeswitch-contrib > project**** > > ** ** > > Which user in the contrib are you targeting? We might be able to ping > him/her. **** > > -MC**** > > On Thu, Nov 10, 2011 at 10:14 AM, Hector Geraldino < > Hector.Geraldino at ip-soft.net> wrote:**** > > Hi,**** > > **** > > What?s the best way to report a bug (and a patch) for a project that is > part of the freeswitch-contrib repository? I can?t find any category on > JIRA for those projects. Should I send the bug description and the proposed > patch to the freeswitch-dev list, or is there a better way?**** > > **** > > 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/20111110/da908fe1/attachment.html From ga at steadfasttelecom.com Thu Nov 10 22:20:51 2011 From: ga at steadfasttelecom.com (Gilad Abada) Date: Thu, 10 Nov 2011 14:20:51 -0500 Subject: [Freeswitch-users] Sofia Recover not retaining record route Message-ID: Hi all I am using sofia recover with unregistered trunks and on recovery Freeswitch seems to lose the record route and VIA headers, before the recover FreeSwitch has all the info. After the recover I get the call back but FreeSwitch is still trying to recover and then ultimately disconnects the call with this error: 2011-11-10 13:48:23.227240 [NOTICE] sofia.c:6039 Hangup sofia/external/+1XXXXXXXXXX at CARRIER.com [CS_SOFT_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] The reINVITE to "salvage" the call is missing the record-route info. I have a capture that I can send for further trouble shooting. This test was done on an incoming call. Thanks so much, Gill -- Gilad Abada SteadFast Telecommunications, Inc. V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. From anthony.minessale at gmail.com Thu Nov 10 22:35:04 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Nov 2011 13:35:04 -0600 Subject: [Freeswitch-users] Sofia Recover not retaining record route In-Reply-To: References: Message-ID: Open a JIRA http://jira.freeswitch.org and supply all the relevant info. Consider contacting consulting at freeswitch.org for commercial assistance to expedite your problem. On Thu, Nov 10, 2011 at 1:20 PM, Gilad Abada wrote: > Hi all > > I am using sofia recover with unregistered trunks and on recovery > Freeswitch seems to lose the record route and VIA headers, before the > recover FreeSwitch has all the info. After the recover I get the call > back but FreeSwitch is still trying to recover and then ultimately > disconnects the call with this error: 2011-11-10 13:48:23.227240 > [NOTICE] sofia.c:6039 Hangup sofia/external/+1XXXXXXXXXX at CARRIER.com > [CS_SOFT_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] > The reINVITE to "salvage" the call is missing the record-route info. > I have a capture that I can send for further trouble shooting. > > This test was done on an incoming call. > > Thanks so much, > Gill > > -- > Gilad Abada > > SteadFast Telecommunications, Inc. > > V: 212.589.1001 > F: 212.589.1011 > > For 35 years, Steadfast Telecommunications has been providing > state-of-the-art communications technology to businesses and > government agencies - large and small. Steadfast Telecommunications > tailors Unified Communications and Voice-Over IP Solutions to > single-site offices or multi-site and worldwide enterprises. Make > your virtual office a reality. Enjoy the freedom to travel while > remaining connected to your office. > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/4ab8dd77/attachment-0001.html From mkdutchman at gmail.com Thu Nov 10 22:29:04 2011 From: mkdutchman at gmail.com (Melvin King) Date: Thu, 10 Nov 2011 14:29:04 -0500 Subject: [Freeswitch-users] freeTDM can't configure with Digium TDM410 card Message-ID: ok, here's my setup. TDM410 card with port 1 having a line to the PSTN and ports 2-4 have the telephone lines plugged in FS version is 1.0.head (git-9dd45e3 2011-11-03 14-45-01 -0500) freetdm.conf has full rw permissions My problem is that I cannot configure freetdm to use this blasted card, no matter what I do..... Assistance appreciated this is my freetdm.conf file [span zt FXS] trunk_type => FXS fxs-channel => 1 [span zt FXO] trunk_type => FXO fxo-channel => 2-4 this is the relevant output from the logfile 2011-11-10 14:16:02.229171 [DEBUG] ftdm_config.c:52 New mod directory: /usr/local/freeswitch/mod 2011-11-10 14:16:02.229411 [DEBUG] ftdm_config.c:58 New config directory: /usr/local/freeswitch/conf 2011-11-10 14:16:02.229656 [DEBUG] ftdm_sched.c:154 Initializing scheduling API 2011-11-10 14:16:02.229732 [DEBUG] ftdm_sched.c:251 Created schedule freetdm-master 2011-11-10 14:16:02.229792 [NOTICE] ftdm_sched.c:178 Launching main schedule thread 2011-11-10 14:16:02.229938 [DEBUG] ftdm_sched.c:187 Running schedule freetdm-master in the main schedule thread 2011-11-10 14:16:02.230029 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/modules.conf. 2011-11-10 14:16:02.230288 [NOTICE] ftdm_io.c:5725 Modules configured: 1 2011-11-10 14:16:02.230494 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/freetdm.conf. 2011-11-10 14:16:02.230592 [DEBUG] ftdm_io.c:4635 Reading FreeTDM configuration file 2011-11-10 14:16:02.230687 [DEBUG] ftdm_io.c:4651 found config for span 2011-11-10 14:16:02.231224 [NOTICE] ftmod_zt.c:1323 Using DAHDI control device 2011-11-10 14:16:02.231410 [INFO] ftdm_io.c:4965 Loading IO from /usr/local/freeswitch/mod/ftmod_zt.so [zt] 2011-11-10 14:16:02.231589 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/zt.conf. 2011-11-10 14:16:02.231826 [INFO] ftmod_zt.c:585 Setting rxgain val to 0.000000 2011-11-10 14:16:02.231922 [INFO] ftmod_zt.c:593 Setting txgain val to 0.000000 2011-11-10 14:16:02.232044 [INFO] ftdm_io.c:787 Auto-loaded I/O module 'zt' 2011-11-10 14:16:02.232201 [DEBUG] ftdm_io.c:4672 created span 1 (FXS) of type zt 2011-11-10 14:16:02.232274 [DEBUG] ftdm_io.c:4690 span 1 [trunk_type]=[FXS] 2011-11-10 14:16:02.232370 [DEBUG] ftdm_io.c:4695 setting trunk type to 'FXS' 2011-11-10 14:16:02.232517 [DEBUG] ftdm_io.c:4690 span 1 [fxs-channel]=[1] 2011-11-10 14:16:02.232770 [WARNING] ftmod_zt.c:352 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 1 fd 16 (Invalid argument)] 2011-11-10 14:16:02.233387 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:17 2011-11-10 14:16:02.233595 [DEBUG] ftdm_io.c:5263 Creating new group:__default 2011-11-10 14:16:02.233746 [DEBUG] ftdm_io.c:4651 found config for span 2011-11-10 14:16:02.233931 [DEBUG] ftdm_io.c:4672 created span 2 (FXO) of type zt 2011-11-10 14:16:02.234045 [DEBUG] ftdm_io.c:4690 span 2 [trunk_type]=[FXO] 2011-11-10 14:16:02.234111 [DEBUG] ftdm_io.c:4695 setting trunk type to 'FXO' 2011-11-10 14:16:02.234173 [DEBUG] ftdm_io.c:4690 span 2 [fxo-channel]=[2-4] 2011-11-10 14:16:02.234550 [WARNING] ftmod_zt.c:352 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 2 fd 16 (Invalid argument)] 2011-11-10 14:16:02.235149 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 2 as FreeTDM device 2:1 fd:18 2011-11-10 14:16:02.235550 [WARNING] ftmod_zt.c:352 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 3 fd 16 (Invalid argument)] 2011-11-10 14:16:02.236157 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 3 as FreeTDM device 2:2 fd:19 2011-11-10 14:16:02.236544 [WARNING] ftmod_zt.c:352 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 4 fd 16 (Invalid argument)] 2011-11-10 14:16:02.237129 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 4 as FreeTDM device 2:3 fd:20 2011-11-10 14:16:02.237385 [INFO] ftdm_io.c:4887 Configured 4 channel(s) 2011-11-10 14:16:02.237644 [ERR] mod_freetdm.c:2953 open of freetdm.conf failed 2011-11-10 14:16:02.238048 [DEBUG] ftdm_sched.c:217 Waiting for main schedule thread to finish 2011-11-10 14:16:02.333760 [NOTICE] ftdm_sched.c:147 Main scheduling thread going out ... 2011-11-10 14:16:02.338252 [INFO] ftdm_io.c:601 Closing channel zt:1:1 fd:17 2011-11-10 14:16:02.338393 [INFO] ftdm_io.c:601 Closing channel zt:2:1 fd:18 2011-11-10 14:16:02.338528 [INFO] ftdm_io.c:601 Closing channel zt:2:2 fd:19 2011-11-10 14:16:02.338601 [INFO] ftdm_io.c:601 Closing channel zt:2:3 fd:20 2011-11-10 14:16:02.338720 [INFO] ftdm_io.c:5092 Unloading I/O interface zt 2011-11-10 14:16:02.338787 [INFO] ftdm_io.c:5099 Unloaded I/O interface zt 2011-11-10 14:16:02.338848 [INFO] ftdm_io.c:5118 Unloading module /usr/local/freeswitch/mod/ftmod_zt.so 2011-11-10 14:16:02.338997 [DEBUG] ftdm_dso.c:90 lib 0xa103d78 was closed with success 2011-11-10 14:16:02.339062 [INFO] ftdm_io.c:5120 Unloaded module /usr/local/freeswitch/mod/ftmod_zt.so 2011-11-10 14:16:02.339124 [DEBUG] ftdm_sched.c:552 Destroying schedule freetdm-master 2011-11-10 14:16:02.339230 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_freetdm.so **Module load routine returned an error** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/6082ccae/attachment.html From moises.silva at gmail.com Thu Nov 10 23:18:59 2011 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 10 Nov 2011 14:18:59 -0600 Subject: [Freeswitch-users] freeTDM can't configure with Digium TDM410 card In-Reply-To: References: Message-ID: On Thu, Nov 10, 2011 at 1:29 PM, Melvin King wrote: > ok, here's my setup. > > TDM410 card with port 1 having a line to the PSTN and ports 2-4 have the > telephone lines plugged in > > FS version is 1.0.head (git-9dd45e3 2011-11-03 14-45-01 -0500) > > freetdm.conf has full rw permissions > > My problem is that I cannot configure freetdm to use this blasted card, no > matter what I do..... > [device /dev/dahdi/channel chan 1 fd 16 (Invalid argument)] > 2011-11-10 14:16:02.233387 [INFO] ftmod_zt.c:404 configuring device > /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:17 > 2011-11-10 14:16:02.233595 [DEBUG] ftdm_io.c:5263 Creating new > group:__default > 2011-11-10 14:16:02.233746 [DEBUG] ftdm_io.c:4651 found config for span > 2011-11-10 14:16:02.233931 [DEBUG] ftdm_io.c:4672 created span 2 (FXO) of > type zt > 2011-11-10 14:16:02.234045 [DEBUG] ftdm_io.c:4690 span 2 [trunk_type]=[FXO] > 2011-11-10 14:16:02.234111 [DEBUG] ftdm_io.c:4695 setting trunk type to > 'FXO' > 2011-11-10 14:16:02.234173 [DEBUG] ftdm_io.c:4690 span 2 > [fxo-channel]=[2-4] > 2011-11-10 14:16:02.234550 [WARNING] ftmod_zt.c:352 this ioctl fails on > older ftdmtel but is harmless if you used ztcfg > [device /dev/dahdi/channel chan 2 fd 16 (Invalid argument)] > This most likely means you're trying to apply a configuration that is not valid for that type of channel (ie, fxs in fxo ports or viceversa). It could also be you did not create /etc/dahdi/system.conf and did not run "dahdi_cfg" *Moises Silva **Software Engineer, Development Manager*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube VegaStream is now part of Sangoma! Ask us about both Gateway Appliances and Internal Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/4cb26d05/attachment.html From mkdutchman at gmail.com Thu Nov 10 23:32:53 2011 From: mkdutchman at gmail.com (Melvin King) Date: Thu, 10 Nov 2011 15:32:53 -0500 Subject: [Freeswitch-users] freeTDM can't configure with Digium TDM410 card In-Reply-To: References: Message-ID: Now that you mention it, I do have to run dahdi_cfg manually every time the system starts up.....as though it never ran to begin with. FS is configured to automatically start. I will recheck the FXS/FXO, and see what happens. Mel On Thu, Nov 10, 2011 at 3:18 PM, Moises Silva wrote: > On Thu, Nov 10, 2011 at 1:29 PM, Melvin King wrote: > >> ok, here's my setup. >> >> TDM410 card with port 1 having a line to the PSTN and ports 2-4 have the >> telephone lines plugged in >> >> FS version is 1.0.head (git-9dd45e3 2011-11-03 14-45-01 -0500) >> >> freetdm.conf has full rw permissions >> >> My problem is that I cannot configure freetdm to use this blasted card, >> no matter what I do..... >> [device /dev/dahdi/channel chan 1 fd 16 (Invalid argument)] >> 2011-11-10 14:16:02.233387 [INFO] ftmod_zt.c:404 configuring device >> /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:17 >> 2011-11-10 14:16:02.233595 [DEBUG] ftdm_io.c:5263 Creating new >> group:__default >> 2011-11-10 14:16:02.233746 [DEBUG] ftdm_io.c:4651 found config for span >> 2011-11-10 14:16:02.233931 [DEBUG] ftdm_io.c:4672 created span 2 (FXO) of >> type zt >> 2011-11-10 14:16:02.234045 [DEBUG] ftdm_io.c:4690 span 2 >> [trunk_type]=[FXO] >> 2011-11-10 14:16:02.234111 [DEBUG] ftdm_io.c:4695 setting trunk type to >> 'FXO' >> 2011-11-10 14:16:02.234173 [DEBUG] ftdm_io.c:4690 span 2 >> [fxo-channel]=[2-4] >> 2011-11-10 14:16:02.234550 [WARNING] ftmod_zt.c:352 this ioctl fails on >> older ftdmtel but is harmless if you used ztcfg >> [device /dev/dahdi/channel chan 2 fd 16 (Invalid argument)] >> > > > This most likely means you're trying to apply a configuration that is not > valid for that type of channel (ie, fxs in fxo ports or viceversa). It > could also be you did not create /etc/dahdi/system.conf and did not run > "dahdi_cfg" > > *Moises Silva > **Software Engineer, Development Manager*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > ** > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter`| > | YouTube > > VegaStream is now part of Sangoma! > > Ask us about both Gateway Appliances > and Internal Gateways > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/73a1eb4b/attachment-0001.html From mkdutchman at gmail.com Thu Nov 10 23:48:45 2011 From: mkdutchman at gmail.com (Melvin King) Date: Thu, 10 Nov 2011 15:48:45 -0500 Subject: [Freeswitch-users] freeTDM can't configure with Digium TDM410 card In-Reply-To: References: Message-ID: ok, I did have the FXO and FXS turned around in my freetdm.conf file. However, it still gives me the "can't open file" error message. Relevant log lines below... 2011-11-10 15:43:30.283719 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_flush_cache' 2011-11-10 15:43:30.283910 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_locate' 2011-11-10 15:43:30.284100 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_wrap' 2011-11-10 15:43:30.286276 [DEBUG] ftdm_config.c:52 New mod directory: /usr/local/freeswitch/mod 2011-11-10 15:43:30.286572 [DEBUG] ftdm_config.c:58 New config directory: /usr/local/freeswitch/conf 2011-11-10 15:43:30.286747 [DEBUG] ftdm_sched.c:154 Initializing scheduling API 2011-11-10 15:43:30.286828 [DEBUG] ftdm_sched.c:251 Created schedule freetdm-master 2011-11-10 15:43:30.286888 [NOTICE] ftdm_sched.c:178 Launching main schedule thread 2011-11-10 15:43:30.287029 [DEBUG] ftdm_sched.c:187 Running schedule freetdm-master in the main schedule thread 2011-11-10 15:43:30.287122 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/modules.conf. 2011-11-10 15:43:30.287570 [NOTICE] ftdm_io.c:5725 Modules configured: 1 2011-11-10 15:43:30.287695 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/freetdm.conf. 2011-11-10 15:43:30.287789 [DEBUG] ftdm_io.c:4635 Reading FreeTDM configuration file 2011-11-10 15:43:30.287889 [DEBUG] ftdm_io.c:4651 found config for span 2011-11-10 15:43:30.288578 [NOTICE] ftmod_zt.c:1323 Using DAHDI control device 2011-11-10 15:43:30.288742 [INFO] ftdm_io.c:4965 Loading IO from /usr/local/freeswitch/mod/ftmod_zt.so [zt] 2011-11-10 15:43:30.288833 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/zt.conf. 2011-11-10 15:43:30.289011 [INFO] ftmod_zt.c:585 Setting rxgain val to 0.000000 2011-11-10 15:43:30.289096 [INFO] ftmod_zt.c:593 Setting txgain val to 0.000000 2011-11-10 15:43:30.289215 [INFO] ftdm_io.c:787 Auto-loaded I/O module 'zt' 2011-11-10 15:43:30.289467 [DEBUG] ftdm_io.c:4672 created span 1 (FXO) of type zt 2011-11-10 15:43:30.289565 [DEBUG] ftdm_io.c:4690 span 1 [trunk_type]=[FXO] 2011-11-10 15:43:30.289636 [DEBUG] ftdm_io.c:4695 setting trunk type to 'FXO' 2011-11-10 15:43:30.289705 [DEBUG] ftdm_io.c:4690 span 1 [fxo-channel]=[1] 2011-11-10 15:43:30.289939 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:18 2011-11-10 15:43:30.290059 [DEBUG] ftdm_io.c:5263 Creating new group:__default 2011-11-10 15:43:30.290240 [DEBUG] ftdm_io.c:4651 found config for span 2011-11-10 15:43:30.290540 [DEBUG] ftdm_io.c:4672 created span 2 (FXS) of type zt 2011-11-10 15:43:30.290640 [DEBUG] ftdm_io.c:4690 span 2 [trunk_type]=[FXS] 2011-11-10 15:43:30.290707 [DEBUG] ftdm_io.c:4695 setting trunk type to 'FXS' 2011-11-10 15:43:30.290775 [DEBUG] ftdm_io.c:4690 span 2 [fxs-channel]=[2-4] 2011-11-10 15:43:30.291014 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 2 as FreeTDM device 2:1 fd:19 2011-11-10 15:43:30.291265 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 3 as FreeTDM device 2:2 fd:20 2011-11-10 15:43:30.291737 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 4 as FreeTDM device 2:3 fd:21 2011-11-10 15:43:30.291966 [INFO] ftdm_io.c:4887 Configured 4 channel(s) 2011-11-10 15:43:30.292140 [ERR] mod_freetdm.c:2953 open of freetdm.conf failed 2011-11-10 15:43:30.295782 [DEBUG] ftdm_sched.c:217 Waiting for main schedule thread to finish 2011-11-10 15:43:30.390235 [NOTICE] ftdm_sched.c:147 Main scheduling thread going out ... 2011-11-10 15:43:30.396054 [INFO] ftdm_io.c:601 Closing channel zt:1:1 fd:18 2011-11-10 15:43:30.396142 [INFO] ftdm_io.c:601 Closing channel zt:2:1 fd:19 2011-11-10 15:43:30.396174 [INFO] ftdm_io.c:601 Closing channel zt:2:2 fd:20 2011-11-10 15:43:30.396207 [INFO] ftdm_io.c:601 Closing channel zt:2:3 fd:21 2011-11-10 15:43:30.396281 [INFO] ftdm_io.c:5092 Unloading I/O interface zt 2011-11-10 15:43:30.396306 [INFO] ftdm_io.c:5099 Unloaded I/O interface zt 2011-11-10 15:43:30.396387 [INFO] ftdm_io.c:5118 Unloading module /usr/local/freeswitch/mod/ftmod_zt.so 2011-11-10 15:43:30.396525 [DEBUG] ftdm_dso.c:90 lib 0x8aa5e48 was closed with success 2011-11-10 15:43:30.396543 [INFO] ftdm_io.c:5120 Unloaded module /usr/local/freeswitch/mod/ftmod_zt.so 2011-11-10 15:43:30.396565 [DEBUG] ftdm_sched.c:552 Destroying schedule freetdm-master 2011-11-10 15:43:30.396636 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_freetdm.so **Module load routine returned an error** On Thu, Nov 10, 2011 at 3:32 PM, Melvin King wrote: > Now that you mention it, I do have to run dahdi_cfg manually every time > the system starts up.....as though it never ran to begin with. FS is > configured to automatically start. > > I will recheck the FXS/FXO, and see what happens. > > Mel > > On Thu, Nov 10, 2011 at 3:18 PM, Moises Silva wrote: > >> On Thu, Nov 10, 2011 at 1:29 PM, Melvin King wrote: >> >>> ok, here's my setup. >>> >>> TDM410 card with port 1 having a line to the PSTN and ports 2-4 have the >>> telephone lines plugged in >>> >>> FS version is 1.0.head (git-9dd45e3 2011-11-03 14-45-01 -0500) >>> >>> freetdm.conf has full rw permissions >>> >>> My problem is that I cannot configure freetdm to use this blasted card, >>> no matter what I do..... >>> [device /dev/dahdi/channel chan 1 fd 16 (Invalid argument)] >>> 2011-11-10 14:16:02.233387 [INFO] ftmod_zt.c:404 configuring device >>> /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:17 >>> 2011-11-10 14:16:02.233595 [DEBUG] ftdm_io.c:5263 Creating new >>> group:__default >>> 2011-11-10 14:16:02.233746 [DEBUG] ftdm_io.c:4651 found config for span >>> 2011-11-10 14:16:02.233931 [DEBUG] ftdm_io.c:4672 created span 2 (FXO) >>> of type zt >>> 2011-11-10 14:16:02.234045 [DEBUG] ftdm_io.c:4690 span 2 >>> [trunk_type]=[FXO] >>> 2011-11-10 14:16:02.234111 [DEBUG] ftdm_io.c:4695 setting trunk type to >>> 'FXO' >>> 2011-11-10 14:16:02.234173 [DEBUG] ftdm_io.c:4690 span 2 >>> [fxo-channel]=[2-4] >>> 2011-11-10 14:16:02.234550 [WARNING] ftmod_zt.c:352 this ioctl fails on >>> older ftdmtel but is harmless if you used ztcfg >>> [device /dev/dahdi/channel chan 2 fd 16 (Invalid argument)] >>> >> >> >> This most likely means you're trying to apply a configuration that is not >> valid for that type of channel (ie, fxs in fxo ports or viceversa). It >> could also be you did not create /etc/dahdi/system.conf and did not run >> "dahdi_cfg" >> >> *Moises Silva >> **Software Engineer, Development Manager*** >> >> msilva at sangoma.com >> >> Sangoma Technologies >> >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >> >> >> t. +1 800 388 2475 (N. America) >> >> t. +1 905 474 1990 x128 >> >> f. +1 905 474 9223 >> >> >> >> ** >> >> Products >> | Solutions >> | Events >> | Contact >> | Wiki >> | Facebook >> | Twitter`| >> | YouTube >> >> VegaStream is now part of Sangoma! >> >> Ask us about both Gateway Appliances >> and Internal Gateways >> >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/75c5f413/attachment.html From mkdutchman at gmail.com Fri Nov 11 00:14:01 2011 From: mkdutchman at gmail.com (Melvin King) Date: Thu, 10 Nov 2011 16:14:01 -0500 Subject: [Freeswitch-users] freeTDM can't configure with Digium TDM410 card In-Reply-To: References: Message-ID: After some more testing/checking/experimenting/swearing this line seems to be the first indication that something is wrong 2011-11-10 16:08:11.785839 [ERR] mod_freetdm.c:2953 open of freetdm.conf failed And it definitely is not a permissions problem, the file has full r/w permissions, it seems more like it can't find the file. Mel On Thu, Nov 10, 2011 at 3:48 PM, Melvin King wrote: > ok, I did have the FXO and FXS turned around in my freetdm.conf file. > However, it still gives me the "can't open file" error message. Relevant > log lines below... > > 2011-11-10 15:43:30.283719 [NOTICE] switch_loadable_module.c:298 Adding > API Function 'xml_flush_cache' > 2011-11-10 15:43:30.283910 [NOTICE] switch_loadable_module.c:298 Adding > API Function 'xml_locate' > 2011-11-10 15:43:30.284100 [NOTICE] switch_loadable_module.c:298 Adding > API Function 'xml_wrap' > 2011-11-10 15:43:30.286276 [DEBUG] ftdm_config.c:52 New mod directory: > /usr/local/freeswitch/mod > 2011-11-10 15:43:30.286572 [DEBUG] ftdm_config.c:58 New config directory: > /usr/local/freeswitch/conf > 2011-11-10 15:43:30.286747 [DEBUG] ftdm_sched.c:154 Initializing > scheduling API > 2011-11-10 15:43:30.286828 [DEBUG] ftdm_sched.c:251 Created schedule > freetdm-master > 2011-11-10 15:43:30.286888 [NOTICE] ftdm_sched.c:178 Launching main > schedule thread > 2011-11-10 15:43:30.287029 [DEBUG] ftdm_sched.c:187 Running schedule > freetdm-master in the main schedule thread > 2011-11-10 15:43:30.287122 [DEBUG] ftdm_config.c:80 Configuration file is > /usr/local/freeswitch/conf/modules.conf. > 2011-11-10 15:43:30.287570 [NOTICE] ftdm_io.c:5725 Modules configured: 1 > 2011-11-10 15:43:30.287695 [DEBUG] ftdm_config.c:80 Configuration file is > /usr/local/freeswitch/conf/freetdm.conf. > 2011-11-10 15:43:30.287789 [DEBUG] ftdm_io.c:4635 Reading FreeTDM > configuration file > 2011-11-10 15:43:30.287889 [DEBUG] ftdm_io.c:4651 found config for span > 2011-11-10 15:43:30.288578 [NOTICE] ftmod_zt.c:1323 Using DAHDI control > device > 2011-11-10 15:43:30.288742 [INFO] ftdm_io.c:4965 Loading IO from > /usr/local/freeswitch/mod/ftmod_zt.so [zt] > 2011-11-10 15:43:30.288833 [DEBUG] ftdm_config.c:80 Configuration file is > /usr/local/freeswitch/conf/zt.conf. > 2011-11-10 15:43:30.289011 [INFO] ftmod_zt.c:585 Setting rxgain val to > 0.000000 > 2011-11-10 15:43:30.289096 [INFO] ftmod_zt.c:593 Setting txgain val to > 0.000000 > 2011-11-10 15:43:30.289215 [INFO] ftdm_io.c:787 Auto-loaded I/O module 'zt' > 2011-11-10 15:43:30.289467 [DEBUG] ftdm_io.c:4672 created span 1 (FXO) of > type zt > 2011-11-10 15:43:30.289565 [DEBUG] ftdm_io.c:4690 span 1 [trunk_type]=[FXO] > 2011-11-10 15:43:30.289636 [DEBUG] ftdm_io.c:4695 setting trunk type to > 'FXO' > 2011-11-10 15:43:30.289705 [DEBUG] ftdm_io.c:4690 span 1 [fxo-channel]=[1] > 2011-11-10 15:43:30.289939 [INFO] ftmod_zt.c:404 configuring device > /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:18 > 2011-11-10 15:43:30.290059 [DEBUG] ftdm_io.c:5263 Creating new > group:__default > 2011-11-10 15:43:30.290240 [DEBUG] ftdm_io.c:4651 found config for span > 2011-11-10 15:43:30.290540 [DEBUG] ftdm_io.c:4672 created span 2 (FXS) of > type zt > 2011-11-10 15:43:30.290640 [DEBUG] ftdm_io.c:4690 span 2 [trunk_type]=[FXS] > 2011-11-10 15:43:30.290707 [DEBUG] ftdm_io.c:4695 setting trunk type to > 'FXS' > 2011-11-10 15:43:30.290775 [DEBUG] ftdm_io.c:4690 span 2 > [fxs-channel]=[2-4] > 2011-11-10 15:43:30.291014 [INFO] ftmod_zt.c:404 configuring device > /dev/dahdi/channel channel 2 as FreeTDM device 2:1 fd:19 > 2011-11-10 15:43:30.291265 [INFO] ftmod_zt.c:404 configuring device > /dev/dahdi/channel channel 3 as FreeTDM device 2:2 fd:20 > 2011-11-10 15:43:30.291737 [INFO] ftmod_zt.c:404 configuring device > /dev/dahdi/channel channel 4 as FreeTDM device 2:3 fd:21 > 2011-11-10 15:43:30.291966 [INFO] ftdm_io.c:4887 Configured 4 channel(s) > 2011-11-10 15:43:30.292140 [ERR] mod_freetdm.c:2953 open of freetdm.conf > failed > 2011-11-10 15:43:30.295782 [DEBUG] ftdm_sched.c:217 Waiting for main > schedule thread to finish > 2011-11-10 15:43:30.390235 [NOTICE] ftdm_sched.c:147 Main scheduling > thread going out ... > 2011-11-10 15:43:30.396054 [INFO] ftdm_io.c:601 Closing channel zt:1:1 > fd:18 > 2011-11-10 15:43:30.396142 [INFO] ftdm_io.c:601 Closing channel zt:2:1 > fd:19 > 2011-11-10 15:43:30.396174 [INFO] ftdm_io.c:601 Closing channel zt:2:2 > fd:20 > 2011-11-10 15:43:30.396207 [INFO] ftdm_io.c:601 Closing channel zt:2:3 > fd:21 > 2011-11-10 15:43:30.396281 [INFO] ftdm_io.c:5092 Unloading I/O interface zt > 2011-11-10 15:43:30.396306 [INFO] ftdm_io.c:5099 Unloaded I/O interface zt > 2011-11-10 15:43:30.396387 [INFO] ftdm_io.c:5118 Unloading module > /usr/local/freeswitch/mod/ftmod_zt.so > 2011-11-10 15:43:30.396525 [DEBUG] ftdm_dso.c:90 lib 0x8aa5e48 was closed > with success > 2011-11-10 15:43:30.396543 [INFO] ftdm_io.c:5120 Unloaded module > /usr/local/freeswitch/mod/ftmod_zt.so > 2011-11-10 15:43:30.396565 [DEBUG] ftdm_sched.c:552 Destroying schedule > freetdm-master > 2011-11-10 15:43:30.396636 [CRIT] switch_loadable_module.c:1281 Error > Loading module /usr/local/freeswitch/mod/mod_freetdm.so > > **Module load routine returned an error** > > On Thu, Nov 10, 2011 at 3:32 PM, Melvin King wrote: > >> Now that you mention it, I do have to run dahdi_cfg manually every time >> the system starts up.....as though it never ran to begin with. FS is >> configured to automatically start. >> >> I will recheck the FXS/FXO, and see what happens. >> >> Mel >> >> On Thu, Nov 10, 2011 at 3:18 PM, Moises Silva wrote: >> >>> On Thu, Nov 10, 2011 at 1:29 PM, Melvin King wrote: >>> >>>> ok, here's my setup. >>>> >>>> TDM410 card with port 1 having a line to the PSTN and ports 2-4 have >>>> the telephone lines plugged in >>>> >>>> FS version is 1.0.head (git-9dd45e3 2011-11-03 14-45-01 -0500) >>>> >>>> freetdm.conf has full rw permissions >>>> >>>> My problem is that I cannot configure freetdm to use this blasted card, >>>> no matter what I do..... >>>> [device /dev/dahdi/channel chan 1 fd 16 (Invalid argument)] >>>> 2011-11-10 14:16:02.233387 [INFO] ftmod_zt.c:404 configuring device >>>> /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:17 >>>> 2011-11-10 14:16:02.233595 [DEBUG] ftdm_io.c:5263 Creating new >>>> group:__default >>>> 2011-11-10 14:16:02.233746 [DEBUG] ftdm_io.c:4651 found config for span >>>> 2011-11-10 14:16:02.233931 [DEBUG] ftdm_io.c:4672 created span 2 (FXO) >>>> of type zt >>>> 2011-11-10 14:16:02.234045 [DEBUG] ftdm_io.c:4690 span 2 >>>> [trunk_type]=[FXO] >>>> 2011-11-10 14:16:02.234111 [DEBUG] ftdm_io.c:4695 setting trunk type to >>>> 'FXO' >>>> 2011-11-10 14:16:02.234173 [DEBUG] ftdm_io.c:4690 span 2 >>>> [fxo-channel]=[2-4] >>>> 2011-11-10 14:16:02.234550 [WARNING] ftmod_zt.c:352 this ioctl fails on >>>> older ftdmtel but is harmless if you used ztcfg >>>> [device /dev/dahdi/channel chan 2 fd 16 (Invalid argument)] >>>> >>> >>> >>> This most likely means you're trying to apply a configuration that is >>> not valid for that type of channel (ie, fxs in fxo ports or viceversa). It >>> could also be you did not create /etc/dahdi/system.conf and did not run >>> "dahdi_cfg" >>> >>> *Moises Silva >>> **Software Engineer, Development Manager*** >>> >>> msilva at sangoma.com >>> >>> Sangoma Technologies >>> >>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >>> >>> >>> t. +1 800 388 2475 (N. America) >>> >>> t. +1 905 474 1990 x128 >>> >>> f. +1 905 474 9223 >>> >>> >>> >>> ** >>> >>> Products >>> | Solutions >>> | Events >>> | Contact >>> | Wiki >>> | Facebook >>> | Twitter`| >>> | YouTube >>> >>> VegaStream is now part of Sangoma! >>> >>> Ask us about both Gateway Appliances >>> and Internal Gateways >>> >>> >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/88d9e0e5/attachment-0001.html From wstephen80 at gmail.com Fri Nov 11 00:17:11 2011 From: wstephen80 at gmail.com (Stephen Wilde) Date: Thu, 10 Nov 2011 22:17:11 +0100 Subject: [Freeswitch-users] Correct way to hangup a channel Message-ID: I have the necessity to hangup a channel inside a media bug callback in a my custom module. Currently, what I do is: static switch_bool_t my_media_callback(switch_media_bug_t *bug, void *user_data, switch_abc_type_t type) { struct my_custom_helper * my_helper = (struct my_custom_helper *) user_data; switch (type) { case SWITCH_ABC_TYPE_READ_REPLACE: case SWITCH_ABC_TYPE_WRITE_REPLACE: { switch_frame_t *frame; if (sth->read) { frame = switch_core_media_bug_get_read_replace_frame(bug); } else { frame = switch_core_media_bug_get_write_replace_frame(bug); } /* .... media analysis .... */ if (to_hangup) { switch_channel_t *channel = switch_core_session_get_channel(my_helper->session); if (channel) { switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING); } } } return SWITCH_TRUE; } The dropped channel is a b-leg, previously bridged with an a-leg (incoming call). Probably in my code there is an error because the code works fine and the channel is correctly dropped but inside Freeswitch remain some "zombies" calls. If I issue a fs_cli "show channels" I see that there are some calls dropped by my module showed in "ACTIVE" state. What is wrong with my channel hangup? Thanks in advance Stephen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/b39c2a9d/attachment.html From anthony.minessale at gmail.com Fri Nov 11 02:34:06 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Nov 2011 17:34:06 -0600 Subject: [Freeswitch-users] Correct way to hangup a channel In-Reply-To: References: Message-ID: look for something that calls either switch_core_session_locate switch_core_session_get_partner or something that calls switch_core_session_read_lock(session); in all of these cases its necessary to call switch_core_session_rwunlock(session); before letting the session pointer go out of scope. On Thu, Nov 10, 2011 at 3:17 PM, Stephen Wilde wrote: > I have the necessity to hangup a channel inside a media bug callback in a > my custom module. > > Currently, what I do is: > > static switch_bool_t my_media_callback(switch_media_bug_t *bug, void > *user_data, switch_abc_type_t type) > { > struct my_custom_helper * my_helper = (struct my_custom_helper *) > user_data; > > switch (type) { > case SWITCH_ABC_TYPE_READ_REPLACE: > case SWITCH_ABC_TYPE_WRITE_REPLACE: > { > switch_frame_t *frame; > > if (sth->read) { > frame = switch_core_media_bug_get_read_replace_frame(bug); > } else { > frame = switch_core_media_bug_get_write_replace_frame(bug); > } > > /* > .... > media analysis > .... > */ > > if (to_hangup) { > switch_channel_t *channel = > switch_core_session_get_channel(my_helper->session); > > if (channel) { > switch_channel_hangup(channel, > SWITCH_CAUSE_NORMAL_CLEARING); > } > } > } > > return SWITCH_TRUE; > } > > > The dropped channel is a b-leg, previously bridged with an a-leg (incoming > call). > > Probably in my code there is an error because the code works fine and the > channel is correctly dropped but inside Freeswitch remain some "zombies" > calls. > If I issue a fs_cli "show channels" I see that there are some calls > dropped by my module showed in "ACTIVE" state. > What is wrong with my channel hangup? > > Thanks in advance > Stephen > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/c80ca192/attachment.html From anthony.minessale at gmail.com Fri Nov 11 02:42:02 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Nov 2011 17:42:02 -0600 Subject: [Freeswitch-users] I missing conference moderator features In-Reply-To: References: Message-ID: FreeSWITCH shares the UNIX philosophy of do one thing and do it well (most of the time) The conference module aims to provide the most basic features possible for others to build on. There are several ways to map keystrokes to various code and plugins for you to make what you ask for but adding it to the conference module itself pollutes the module. (I already feel bad about having the pin feature built in) if anything changes to mod conference should be attempts to reduce complexity rather than increase it. On Wed, Nov 9, 2011 at 8:12 AM, Thomas Hoellriegel wrote: > Hi all, > I have a few suggestions as to improve the usability conferences as > moderator. > > For example: > you press a key, and become these menuoptions: > with 1 and 3, they can search through the channels in lissen only mode. > press 2 for the current channel. > press 4 to bridge the current channel with your self. > press 5 to mute the current channel. > press 6 to kick the channel from the conference. > Press 7 to move the current channel to another conference-number. > press the 8 to the subscriber number of the user to hear. > > So you can simply remove disturb as moderator of the conference soon. > > eavdrop has only limiting functions. > Unfortunately, I find no application where this enables channel-search > mode. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111110/caaff0ab/attachment.html From avi at avimarcus.net Fri Nov 11 03:26:07 2011 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 11 Nov 2011 02:26:07 +0200 Subject: [Freeswitch-users] Call Broadcasts - when to start playing? Message-ID: I've got a simple message that a business needs to play for all his contacts. How do you folks do broadcasts? There's the commerical mod_amd but I don't think I'm getting that just yet. wait_for_silence for 2 seconds (to wait out the answering machine?), then play the message? How many "intervals" is 2 seconds? Or mix wait_for_silence with mod_avmd via ESL so either one triggers it? Or does mod_avmd do both the beep (voicemail still beeps??) and the silence? Thanks for your input, -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111111/d9601c48/attachment.html From hynek.cihlar at gmail.com Fri Nov 11 10:05:50 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Fri, 11 Nov 2011 08:05:50 +0100 Subject: [Freeswitch-users] Playlist playback In-Reply-To: References: <2968466961287872078@unknownmsgid> Message-ID: Michael, tx for the suggestions. The mod_file_string is what I was looking for. Unfortunately my target language is not supported so I'm implementing my own phrase support, although very limited. Hynek On Thu, Nov 10, 2011 at 7:32 PM, Michael Collins wrote: > Yep, there are several ways. Check this out: > http://wiki.freeswitch.org/wiki/Mod_file_string > > Essentially you can do a !-delimited list of files. Also, don't forget > about the phrase macros, which allow you to get even more creative with > playing multiple files... > > -MC > > On Thu, Nov 10, 2011 at 10:28 AM, Hynek Cihlar wrote: > >> Does Freeswitch support playing back of multiple audio files? I.e. >> issuing one command with multiple files? >> >> I am trying to build dynamic dialogs over event socket and issuing >> call for every part of the dialog seems as a waste. >> >> >> Sent from my mobile device >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111111/99558f86/attachment-0001.html From Stefan.Weigel at allianz-warranty.com Fri Nov 11 10:23:56 2011 From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan) Date: Fri, 11 Nov 2011 08:23:56 +0100 Subject: [Freeswitch-users] call groups - sort order In-Reply-To: References: <5003D7D3E06F514E8C682F18D223265C04D3B36FA8@AZWSMS03.azwarranty.int> Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36FBC@AZWSMS03.azwarranty.int> Hi Michael, http://wiki.freeswitch.org/wiki/Mod_db Best regards, Stefan Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins Gesendet: Donnerstag, 10. November 2011 19:24 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] call groups - sort order Which wiki page is this? -MC On Wed, Nov 9, 2011 at 12:58 AM, Weigel, Stefan > wrote: Hi list, from the WIKI page: group call:groupname[:order] How are the entries sorted ? How can I make sure the list returned by 'group call' is sorted in a certain order ? I want to make sure a certain extension is ringed first, but sometimes another extension is listed first. Thanks and best regards Stefan FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111111/ac4a5363/attachment.html From miha at softnet.si Fri Nov 11 11:02:14 2011 From: miha at softnet.si (Miha Zoubek) Date: Fri, 11 Nov 2011 09:02:14 +0100 Subject: [Freeswitch-users] freeswitch radiusclinet (mod_radius_cdr) problems In-Reply-To: References: <4EBA8B5E.2050900@softnet.si> Message-ID: <4EBCD686.9040305@softnet.si> Hi @ Ognjen, bellow you can see that my Accounting-Request(4), Accounting-Response(5) are twice send. Any idea? BR, Miha 71.449050 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=235, l=265) 71.517347 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=235, l=20) 73.536126 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Request(1) (id=236, l=210) 73.567412 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Accept(2) (id=236, l=20) 73.572794 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=237, l=321) 73.574156 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=237, l=20) 83.482760 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=238, l=401) 83.485670 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=238, l=20) 83.514594 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=239, l=402) 83.516404 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=239, l=20) On 11/10/2011 6:24 PM, Ognjen Seslija wrote: > Radius clients generally send Start and Stop records, I guess this is > why you get two records. > > On Wed, Nov 9, 2011 at 3:17 PM, Miha Zoubek > wrote: > > Hi, > > radiusclient on freeswitch in sending more that once same > response. So I > am getting in my sql tables for every call two inputs which is wrong. > How can I deal whit this issue? > > I have paste a log from freeradius server in pastebin that you can see > what is freeswitch sending. > > http://pastebin.freeswitch.org/17730 > > > Thank you! > > BR, > Miha > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111111/3358f9be/attachment.html From fieldpeak at gmail.com Fri Nov 11 11:59:31 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Fri, 11 Nov 2011 16:59:31 +0800 Subject: [Freeswitch-users] paramter "NDLB-force-rport" did not effect In-Reply-To: References: Message-ID: sorry for the confusion, now i understood the rport, and now i use NDLB-connectile-dysfunction for each users, this problem was resolved, thanks all for kindly help!BR, Charles 2011/11/9 Brian West > What is the user agent? > > /b > > On Nov 8, 2011, at 1:30 AM, fieldpeak wrote: > > Dear Friends, > > i met a problem that call a registered user behind NAT, i need FS send > INVITE message to the port of the NAT device from which the registered > user sent out REGISTRATION but not according to the CONTACT header. > > see the wiki, i enable below paramter, > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111111/d85ad329/attachment.html From acrow at integrafin.co.uk Fri Nov 11 13:06:54 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 11 Nov 2011 10:06:54 +0000 Subject: [Freeswitch-users] {Half-solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EBBDB3B.2070509@coppice.org> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> Message-ID: <4EBCF3BE.3030009@integrafin.co.uk> On 10/11/11 14:10, Steve Underwood wrote: > On 11/10/2011 03:28 PM, Alex Crow wrote: >> Brian, >> >> Can you elaborate? It seems that the call is PCMA, as far as I can >> tell t38modem supports PCMA. What can I do to fix this? I thought this >> would work with audio faxing? >> >> Or am I completely wrong and I can only do t38 using t38modemso I do >> need some kind of gateway either on or outside of FS to get standard >> fax working? This was why I had the gateway stuff in the dial plan in >> the first place. >> > If t38modem supports audio FAXing, its a new feature. Certainly until > recently it was a pure T.38 package. > > Steve > Steve, I couldn't get it to do so for inbound faxes, I had to transcode as it only seems to work with T38 input, but it is now working: However on the outbound side it seems to be insisting on sending audio (ALAW/ULAW), but even if I don't transcode the fax never completes. T38 modem keeps giving me: 2011/11/11 09:51:03.494 Media Patch:0xb597db70 RTP_UDP Session 1, Control port on remote not ready. Whatever dialplan I try I can't seem to make this go away. Anyone got t38modem working outbound? Cheers Alex > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From Prometheus001 at gmx.net Fri Nov 11 14:07:56 2011 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 11 Nov 2011 12:07:56 +0100 Subject: [Freeswitch-users] Conference and room numbers in a multi tenant environment Message-ID: <4EBD020C.5080105@gmx.net> Hello, I created 2 conferences * 1000 at domain1 * 1000 at doamin2 When I called * conference 1000 at domain1 from user 100 at domain1 and * conference 1000 at domain2 from user 100 at domain2 both users are in the same conference in my case. Question: Is 1000 at domain1 distinct from 1000 at domain2 or do I have to use seperate room numbers? Best regards Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111111/bc608c70/attachment.html From acrow at integrafin.co.uk Fri Nov 11 15:52:19 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 11 Nov 2011 12:52:19 +0000 Subject: [Freeswitch-users] {Almost there} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EBCF3BE.3030009@integrafin.co.uk> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> Message-ID: <4EBD1A83.2020807@integrafin.co.uk> If anyone's interested, I've get outbound fax semi-working with this setup: t38modem is started like this: t38modem -ttt -o /var/log/t38modem.log --no-h323 -u t38modem --sip-register t38modem at 192.168.20.245,secret --sip-listen udp\$*:6060 --ptty +/dev/ttyT38-1 --route "modem:.*=sip:@192.168.20.245;OPAL-Force-Fax-Mode=true" --route "sip:.*=modem:" The remaining problem is that on the receiving end we only have the top centimetre or two of the image. I think this may be a tiff library issue but I'm not sure. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From mkdutchman at gmail.com Fri Nov 11 16:05:47 2011 From: mkdutchman at gmail.com (Melvin King) Date: Fri, 11 Nov 2011 08:05:47 -0500 Subject: [Freeswitch-users] freeTDM can't configure with Digium TDM410 card In-Reply-To: References: Message-ID: It looks like I've found the solution, and perhaps the problem I was able to manually load the ftdm module after freeswitch was started, so all I did was move the line that loads the ftdm module (inside the modules.conf.xml) to the bottom, in effect loading the module after all the other modules were loaded. Presto, problem solved. I guess module loading order does matter. On a somewhat unrelated note, the same thing is happening with the dahdi hardware, dahdi_cfg is being run before the hardware drivers are loaded, so that probably needs the same solution. Mel On Thu, Nov 10, 2011 at 4:14 PM, Melvin King wrote: > After some more testing/checking/experimenting/swearing this line seems to > be the first indication that something is wrong > > 2011-11-10 16:08:11.785839 [ERR] mod_freetdm.c:2953 open of freetdm.conf > failed > > And it definitely is not a permissions problem, the file has full r/w > permissions, it seems more like it can't find the file. > > Mel > > > On Thu, Nov 10, 2011 at 3:48 PM, Melvin King wrote: > >> ok, I did have the FXO and FXS turned around in my freetdm.conf file. >> However, it still gives me the "can't open file" error message. Relevant >> log lines below... >> >> 2011-11-10 15:43:30.283719 [NOTICE] switch_loadable_module.c:298 Adding >> API Function 'xml_flush_cache' >> 2011-11-10 15:43:30.283910 [NOTICE] switch_loadable_module.c:298 Adding >> API Function 'xml_locate' >> 2011-11-10 15:43:30.284100 [NOTICE] switch_loadable_module.c:298 Adding >> API Function 'xml_wrap' >> 2011-11-10 15:43:30.286276 [DEBUG] ftdm_config.c:52 New mod directory: >> /usr/local/freeswitch/mod >> 2011-11-10 15:43:30.286572 [DEBUG] ftdm_config.c:58 New config directory: >> /usr/local/freeswitch/conf >> 2011-11-10 15:43:30.286747 [DEBUG] ftdm_sched.c:154 Initializing >> scheduling API >> 2011-11-10 15:43:30.286828 [DEBUG] ftdm_sched.c:251 Created schedule >> freetdm-master >> 2011-11-10 15:43:30.286888 [NOTICE] ftdm_sched.c:178 Launching main >> schedule thread >> 2011-11-10 15:43:30.287029 [DEBUG] ftdm_sched.c:187 Running schedule >> freetdm-master in the main schedule thread >> 2011-11-10 15:43:30.287122 [DEBUG] ftdm_config.c:80 Configuration file is >> /usr/local/freeswitch/conf/modules.conf. >> 2011-11-10 15:43:30.287570 [NOTICE] ftdm_io.c:5725 Modules configured: 1 >> 2011-11-10 15:43:30.287695 [DEBUG] ftdm_config.c:80 Configuration file is >> /usr/local/freeswitch/conf/freetdm.conf. >> 2011-11-10 15:43:30.287789 [DEBUG] ftdm_io.c:4635 Reading FreeTDM >> configuration file >> 2011-11-10 15:43:30.287889 [DEBUG] ftdm_io.c:4651 found config for span >> 2011-11-10 15:43:30.288578 [NOTICE] ftmod_zt.c:1323 Using DAHDI control >> device >> 2011-11-10 15:43:30.288742 [INFO] ftdm_io.c:4965 Loading IO from >> /usr/local/freeswitch/mod/ftmod_zt.so [zt] >> 2011-11-10 15:43:30.288833 [DEBUG] ftdm_config.c:80 Configuration file is >> /usr/local/freeswitch/conf/zt.conf. >> 2011-11-10 15:43:30.289011 [INFO] ftmod_zt.c:585 Setting rxgain val to >> 0.000000 >> 2011-11-10 15:43:30.289096 [INFO] ftmod_zt.c:593 Setting txgain val to >> 0.000000 >> 2011-11-10 15:43:30.289215 [INFO] ftdm_io.c:787 Auto-loaded I/O module >> 'zt' >> 2011-11-10 15:43:30.289467 [DEBUG] ftdm_io.c:4672 created span 1 (FXO) of >> type zt >> 2011-11-10 15:43:30.289565 [DEBUG] ftdm_io.c:4690 span 1 >> [trunk_type]=[FXO] >> 2011-11-10 15:43:30.289636 [DEBUG] ftdm_io.c:4695 setting trunk type to >> 'FXO' >> 2011-11-10 15:43:30.289705 [DEBUG] ftdm_io.c:4690 span 1 [fxo-channel]=[1] >> 2011-11-10 15:43:30.289939 [INFO] ftmod_zt.c:404 configuring device >> /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:18 >> 2011-11-10 15:43:30.290059 [DEBUG] ftdm_io.c:5263 Creating new >> group:__default >> 2011-11-10 15:43:30.290240 [DEBUG] ftdm_io.c:4651 found config for span >> 2011-11-10 15:43:30.290540 [DEBUG] ftdm_io.c:4672 created span 2 (FXS) of >> type zt >> 2011-11-10 15:43:30.290640 [DEBUG] ftdm_io.c:4690 span 2 >> [trunk_type]=[FXS] >> 2011-11-10 15:43:30.290707 [DEBUG] ftdm_io.c:4695 setting trunk type to >> 'FXS' >> 2011-11-10 15:43:30.290775 [DEBUG] ftdm_io.c:4690 span 2 >> [fxs-channel]=[2-4] >> 2011-11-10 15:43:30.291014 [INFO] ftmod_zt.c:404 configuring device >> /dev/dahdi/channel channel 2 as FreeTDM device 2:1 fd:19 >> 2011-11-10 15:43:30.291265 [INFO] ftmod_zt.c:404 configuring device >> /dev/dahdi/channel channel 3 as FreeTDM device 2:2 fd:20 >> 2011-11-10 15:43:30.291737 [INFO] ftmod_zt.c:404 configuring device >> /dev/dahdi/channel channel 4 as FreeTDM device 2:3 fd:21 >> 2011-11-10 15:43:30.291966 [INFO] ftdm_io.c:4887 Configured 4 channel(s) >> 2011-11-10 15:43:30.292140 [ERR] mod_freetdm.c:2953 open of freetdm.conf >> failed >> 2011-11-10 15:43:30.295782 [DEBUG] ftdm_sched.c:217 Waiting for main >> schedule thread to finish >> 2011-11-10 15:43:30.390235 [NOTICE] ftdm_sched.c:147 Main scheduling >> thread going out ... >> 2011-11-10 15:43:30.396054 [INFO] ftdm_io.c:601 Closing channel zt:1:1 >> fd:18 >> 2011-11-10 15:43:30.396142 [INFO] ftdm_io.c:601 Closing channel zt:2:1 >> fd:19 >> 2011-11-10 15:43:30.396174 [INFO] ftdm_io.c:601 Closing channel zt:2:2 >> fd:20 >> 2011-11-10 15:43:30.396207 [INFO] ftdm_io.c:601 Closing channel zt:2:3 >> fd:21 >> 2011-11-10 15:43:30.396281 [INFO] ftdm_io.c:5092 Unloading I/O interface >> zt >> 2011-11-10 15:43:30.396306 [INFO] ftdm_io.c:5099 Unloaded I/O interface zt >> 2011-11-10 15:43:30.396387 [INFO] ftdm_io.c:5118 Unloading module >> /usr/local/freeswitch/mod/ftmod_zt.so >> 2011-11-10 15:43:30.396525 [DEBUG] ftdm_dso.c:90 lib 0x8aa5e48 was closed >> with success >> 2011-11-10 15:43:30.396543 [INFO] ftdm_io.c:5120 Unloaded module >> /usr/local/freeswitch/mod/ftmod_zt.so >> 2011-11-10 15:43:30.396565 [DEBUG] ftdm_sched.c:552 Destroying schedule >> freetdm-master >> 2011-11-10 15:43:30.396636 [CRIT] switch_loadable_module.c:1281 Error >> Loading module /usr/local/freeswitch/mod/mod_freetdm.so >> >> **Module load routine returned an error** >> >> On Thu, Nov 10, 2011 at 3:32 PM, Melvin King wrote: >> >>> Now that you mention it, I do have to run dahdi_cfg manually every time >>> the system starts up.....as though it never ran to begin with. FS is >>> configured to automatically start. >>> >>> I will recheck the FXS/FXO, and see what happens. >>> >>> Mel >>> >>> On Thu, Nov 10, 2011 at 3:18 PM, Moises Silva wrote: >>> >>>> On Thu, Nov 10, 2011 at 1:29 PM, Melvin King wrote: >>>> >>>>> ok, here's my setup. >>>>> >>>>> TDM410 card with port 1 having a line to the PSTN and ports 2-4 have >>>>> the telephone lines plugged in >>>>> >>>>> FS version is 1.0.head (git-9dd45e3 2011-11-03 14-45-01 -0500) >>>>> >>>>> freetdm.conf has full rw permissions >>>>> >>>>> My problem is that I cannot configure freetdm to use this blasted >>>>> card, no matter what I do..... >>>>> [device /dev/dahdi/channel chan 1 fd 16 (Invalid argument)] >>>>> 2011-11-10 14:16:02.233387 [INFO] ftmod_zt.c:404 configuring device >>>>> /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:17 >>>>> 2011-11-10 14:16:02.233595 [DEBUG] ftdm_io.c:5263 Creating new >>>>> group:__default >>>>> 2011-11-10 14:16:02.233746 [DEBUG] ftdm_io.c:4651 found config for span >>>>> 2011-11-10 14:16:02.233931 [DEBUG] ftdm_io.c:4672 created span 2 (FXO) >>>>> of type zt >>>>> 2011-11-10 14:16:02.234045 [DEBUG] ftdm_io.c:4690 span 2 >>>>> [trunk_type]=[FXO] >>>>> 2011-11-10 14:16:02.234111 [DEBUG] ftdm_io.c:4695 setting trunk type >>>>> to 'FXO' >>>>> 2011-11-10 14:16:02.234173 [DEBUG] ftdm_io.c:4690 span 2 >>>>> [fxo-channel]=[2-4] >>>>> 2011-11-10 14:16:02.234550 [WARNING] ftmod_zt.c:352 this ioctl fails >>>>> on older ftdmtel but is harmless if you used ztcfg >>>>> [device /dev/dahdi/channel chan 2 fd 16 (Invalid argument)] >>>>> >>>> >>>> >>>> This most likely means you're trying to apply a configuration that is >>>> not valid for that type of channel (ie, fxs in fxo ports or viceversa). It >>>> could also be you did not create /etc/dahdi/system.conf and did not run >>>> "dahdi_cfg" >>>> >>>> *Moises Silva >>>> **Software Engineer, Development Manager*** >>>> >>>> msilva at sangoma.com >>>> >>>> Sangoma Technologies >>>> >>>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >>>> >>>> >>>> t. +1 800 388 2475 (N. America) >>>> >>>> t. +1 905 474 1990 x128 >>>> >>>> f. +1 905 474 9223 >>>> >>>> >>>> >>>> ** >>>> >>>> Products >>>> | Solutions >>>> | Events >>>> | Contact >>>> | Wiki >>>> | Facebook >>>> | Twitter`| >>>> | YouTube >>>> >>>> VegaStream is now part of Sangoma! >>>> >>>> Ask us about both Gateway Appliances >>>> and Internal Gateways >>>> >>>> >>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111111/4ee7f1cc/attachment-0001.html From wstephen80 at gmail.com Fri Nov 11 16:42:39 2011 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 11 Nov 2011 14:42:39 +0100 Subject: [Freeswitch-users] Correct way to hangup a channel In-Reply-To: References: Message-ID: Thank you Anthony, solved! I have in my code a call to a "switch_core_session_locate" so, as you say, after adding a "switch_core_session_rwunlock" the problem disappear and there are no more "zombie" sessions in my Freeswitch. Stephen On Fri, Nov 11, 2011 at 12:34 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > look for something that calls either switch_core_session_locate > switch_core_session_get_partner or something that calls > switch_core_session_read_lock(session); in all of these cases its > necessary to call switch_core_session_rwunlock(session); before letting the > session pointer go out of scope. > > > On Thu, Nov 10, 2011 at 3:17 PM, Stephen Wilde wrote: > >> I have the necessity to hangup a channel inside a media bug callback in a >> my custom module. >> >> Currently, what I do is: >> >> static switch_bool_t my_media_callback(switch_media_bug_t *bug, void >> *user_data, switch_abc_type_t type) >> { >> struct my_custom_helper * my_helper = (struct my_custom_helper *) >> user_data; >> >> switch (type) { >> case SWITCH_ABC_TYPE_READ_REPLACE: >> case SWITCH_ABC_TYPE_WRITE_REPLACE: >> { >> switch_frame_t *frame; >> >> if (sth->read) { >> frame = switch_core_media_bug_get_read_replace_frame(bug); >> } else { >> frame = >> switch_core_media_bug_get_write_replace_frame(bug); >> } >> >> /* >> .... >> media analysis >> .... >> */ >> >> if (to_hangup) { >> switch_channel_t *channel = >> switch_core_session_get_channel(my_helper->session); >> >> if (channel) { >> switch_channel_hangup(channel, >> SWITCH_CAUSE_NORMAL_CLEARING); >> } >> } >> } >> >> return SWITCH_TRUE; >> } >> >> >> The dropped channel is a b-leg, previously bridged with an a-leg >> (incoming call). >> >> Probably in my code there is an error because the code works fine and the >> channel is correctly dropped but inside Freeswitch remain some "zombies" >> calls. >> If I issue a fs_cli "show channels" I see that there are some calls >> dropped by my module showed in "ACTIVE" state. >> What is wrong with my channel hangup? >> >> Thanks in advance >> Stephen >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111111/f9b314e0/attachment.html From acrow at integrafin.co.uk Fri Nov 11 16:48:32 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 11 Nov 2011 13:48:32 +0000 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EBD1A83.2020807@integrafin.co.uk> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> Message-ID: <4EBD27B0.4000504@integrafin.co.uk> On 11/11/11 12:52, Alex Crow wrote: > If anyone's interested, I've get outbound fax semi-working with this setup: > > > > > > > > > > > t38modem is started like this: > > t38modem -ttt -o /var/log/t38modem.log --no-h323 -u t38modem > --sip-register t38modem at 192.168.20.245,secret --sip-listen udp\$*:6060 > --ptty +/dev/ttyT38-1 --route > "modem:.*=sip:@192.168.20.245;OPAL-Force-Fax-Mode=true" --route > "sip:.*=modem:" > > > The remaining problem is that on the receiving end we only have the top > centimetre or two of the image. I think this may be a tiff library issue > but I'm not sure. > > Cheers > > Alex > In the end I had to disable ECM in the Hylafax modem definition for t38modem, this was what was causing the image truncation. Odd, because both t38modem and spandsp ostensibly support ECM. I'm wondering if I was sending audio out directly over TDM rather than via a SIP gateway using G.711 then ECM would be OK... probably a timing issue. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From tculjaga at gmail.com Fri Nov 11 16:51:53 2011 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 11 Nov 2011 14:51:53 +0100 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EBD27B0.4000504@integrafin.co.uk> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> Message-ID: always disable ECM for fax calls over IP... its the 1st rule of thumb for VoIP faxing :=) On Fri, Nov 11, 2011 at 2:48 PM, Alex Crow wrote: > On 11/11/11 12:52, Alex Crow wrote: > > If anyone's interested, I've get outbound fax semi-working with this > setup: > > > > > > > > > > > > > > > > > > > > > > t38modem is started like this: > > > > t38modem -ttt -o /var/log/t38modem.log --no-h323 -u t38modem > > --sip-register t38modem at 192.168.20.245,secret --sip-listen udp\$*:6060 > > --ptty +/dev/ttyT38-1 --route > > "modem:.*=sip:@192.168.20.245;OPAL-Force-Fax-Mode=true" --route > > "sip:.*=modem:" > > > > > > The remaining problem is that on the receiving end we only have the top > > centimetre or two of the image. I think this may be a tiff library issue > > but I'm not sure. > > > > Cheers > > > > Alex > > > > In the end I had to disable ECM in the Hylafax modem definition for > t38modem, this was what was causing the image truncation. Odd, because > both t38modem and spandsp ostensibly support ECM. I'm wondering if I was > sending audio out directly over TDM rather than via a SIP gateway using > G.711 then ECM would be OK... probably a timing issue. > > Cheers > Alex > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under > number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on > the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111111/557dad42/attachment.html From brian at freeswitch.org Fri Nov 11 17:59:46 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Nov 2011 08:59:46 -0600 Subject: [Freeswitch-users] {Almost there} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EBD1A83.2020807@integrafin.co.uk> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> Message-ID: <5A118B9A-9FAF-4465-A1B9-58EBFC38B217@freeswitch.org> does t38modem not do t38? /b On Nov 11, 2011, at 6:52 AM, Alex Crow wrote: > If anyone's interested, I've get outbound fax semi-working with this setup: > > > > > > > > > > > t38modem is started like this: > > t38modem -ttt -o /var/log/t38modem.log --no-h323 -u t38modem > --sip-register t38modem at 192.168.20.245,secret --sip-listen udp\$*:6060 > --ptty +/dev/ttyT38-1 --route > "modem:.*=sip:@192.168.20.245;OPAL-Force-Fax-Mode=true" --route > "sip:.*=modem:" > > > The remaining problem is that on the receiving end we only have the top > centimetre or two of the image. I think this may be a tiff library issue > but I'm not sure. > > Cheers > > Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111111/3b50bd50/attachment-0001.html From steveu at coppice.org Fri Nov 11 18:32:34 2011 From: steveu at coppice.org (Steve Underwood) Date: Fri, 11 Nov 2011 23:32:34 +0800 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> Message-ID: <4EBD4012.7040802@coppice.org> Turning off ECM is equivalent to saying "I use FAX, but I couldn't give a damn if they are received". Steve On 11/11/2011 09:51 PM, Tihomir Culjaga wrote: > always disable ECM for fax calls over IP... its the 1st rule of thumb > for VoIP faxing :=) > > On Fri, Nov 11, 2011 at 2:48 PM, Alex Crow > wrote: > > On 11/11/11 12:52, Alex Crow wrote: > > If anyone's interested, I've get outbound fax semi-working with > this setup: > > > > > > > > > > > > > > > > > > > > > > t38modem is started like this: > > > > t38modem -ttt -o /var/log/t38modem.log --no-h323 -u t38modem > > --sip-register t38modem at 192.168.20.245 > ,secret --sip-listen udp\$*:6060 > > --ptty +/dev/ttyT38-1 --route > > "modem:.*=sip:@192.168.20.245 > ;OPAL-Force-Fax-Mode=true" --route > > "sip:.*=modem:" > > > > > > The remaining problem is that on the receiving end we only have > the top > > centimetre or two of the image. I think this may be a tiff > library issue > > but I'm not sure. > > > > Cheers > > > > Alex > > > > In the end I had to disable ECM in the Hylafax modem definition for > t38modem, this was what was causing the image truncation. Odd, because > both t38modem and spandsp ostensibly support ECM. I'm wondering if > I was > sending audio out directly over TDM rather than via a SIP gateway > using > G.711 then ECM would be OK... probably a timing issue. > > Cheers > Alex > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales > under number: 3727592) > Authorised and regulated by the Financial Services Authority > (entered on the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 Fri Nov 11 19:18:48 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Nov 2011 10:18:48 -0600 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EBD4012.7040802@coppice.org> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> Message-ID: <82A6EE94-026E-44FD-9D33-586FB7077FC3@freeswitch.org> Disabling ECM by FAR the WRONG thing to do and you have this mindset that is required its only required when you were hanging on by a thread doing G.711 Faxing over the public internet and you didn't want one tiny blip to screw the fax over. I still don't understand why Alex is doing t.38 gateway from audio to t.38? Either way ECM works fine never had any issues with it. /b On Nov 11, 2011, at 9:32 AM, Steve Underwood wrote: > Turning off ECM is equivalent to saying "I use FAX, but I couldn't give > a damn if they are received". > > Steve > > > On 11/11/2011 09:51 PM, Tihomir Culjaga wrote: >> always disable ECM for fax calls over IP... its the 1st rule of thumb >> for VoIP faxing :=) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111111/366a2364/attachment.html From msc at freeswitch.org Fri Nov 11 20:14:22 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Nov 2011 09:14:22 -0800 Subject: [Freeswitch-users] call groups - sort order In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36FBC@AZWSMS03.azwarranty.int> References: <5003D7D3E06F514E8C682F18D223265C04D3B36FA8@AZWSMS03.azwarranty.int> <5003D7D3E06F514E8C682F18D223265C04D3B36FBC@AZWSMS03.azwarranty.int> Message-ID: Stefan, I looked in mod_db.c and if I understand what I'm reading there isn't an explicit sort when pulling this data out of the database table. Here's the line in question: sql = switch_mprintf("select url,'%q' from group_data where groupname='%q'", how, argv[1]); I would recommend that you log in to sqlite3 and issue this command manually and see what you get. I don't know enough about SQLite to be able to tell you if it enforces any kind of order when doing a select on a table when there's no ORDER BY clause. -MC On Thu, Nov 10, 2011 at 11:23 PM, Weigel, Stefan < Stefan.Weigel at allianz-warranty.com> wrote: > Hi Michael,**** > > ** ** > > http://wiki.freeswitch.org/wiki/Mod_db**** > > ** ** > > ** ** > > Best regards,**** > > ** ** > > Stefan**** > > ** ** > > ** ** > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Michael > Collins > *Gesendet:* Donnerstag, 10. November 2011 19:24 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] call groups - sort order**** > > ** ** > > Which wiki page is this?**** > > -MC**** > > On Wed, Nov 9, 2011 at 12:58 AM, Weigel, Stefan < > Stefan.Weigel at allianz-warranty.com> wrote:**** > > Hi list,**** > > **** > > from the WIKI page:**** > > **** > > group call:groupname[:order]**** > > **** > > How are the entries sorted ? How can I make sure the list returned by > ?group call? is sorted in a certain order ? I want to make sure a certain > extension is ringed first, but sometimes another extension is listed first. > **** > > **** > > **** > > Thanks and best regards**** > > **** > > Stefan**** > > **** > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111111/69137003/attachment.html From x.liu at hw.ac.uk Fri Nov 11 21:59:53 2011 From: x.liu at hw.ac.uk (xl127) Date: Fri, 11 Nov 2011 18:59:53 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> Message-ID: <4EBD70A9.6090505@hw.ac.uk> Hello, I found this is a question that was asked before by others, but I didn't find the answer. Anyway, I am using FS ESL outbound mode connecting to my IVR app, using FS's "speak" and "detect_speech" to access Nuance MRCP V1 server. I want the user to be able to barge-in during the system's speaking. How could I implement it? I tried to specify "kill-on-barge-in=true" in the mrcp config profile. The barge-in doesn't work. With or without setting kill-on-barge-in, FS stops responding to my phone call and eventually it hangs up my call if I speak somthing (do the barge-in) during the system's speaking. I made a turn-by-turn loop in my app, the ASR/TTS works fine if I do not do barge-in ( I wait until the TTS finishes then I start to speak) Any advices please? Thanks! Xing -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 From msc at freeswitch.org Fri Nov 11 23:18:35 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Nov 2011 12:18:35 -0800 Subject: [Freeswitch-users] Conference and room numbers in a multi tenant environment In-Reply-To: <4EBD020C.5080105@gmx.net> References: <4EBD020C.5080105@gmx.net> Message-ID: I believe the different domains share the same conference.conf.xml file, so you'll need either different room numbers or different conference profile names for each domain. -MC On Fri, Nov 11, 2011 at 3:07 AM, Peter P GMX wrote: > ** > Hello, > > I created 2 conferences > > - 1000 at domain1 > - 1000 at doamin2 > > When I called > > - conference 1000 at domain1 from user 100 at domain1 and > - conference 1000 at domain2 from user 100 at domain2 > > both users are in the same conference in my case. > > Question: Is 1000 at domain1 distinct from 1000 at domain2 or do I have to use > seperate room numbers? > > Best regards > Peter > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111111/e42b6bff/attachment-0001.html From krice at freeswitch.org Sat Nov 12 00:43:36 2011 From: krice at freeswitch.org (Ken Rice) Date: Fri, 11 Nov 2011 15:43:36 -0600 Subject: [Freeswitch-users] Conference and room numbers in a multi tenant environment In-Reply-To: Message-ID: Conference 1000 at domain1 and conf 1000 at domain2 are not conferences in different domains in freeswitch, the are conferences with different conference profiles The easiest way to get what you want is to do 1000-domain1 at profile and 1000-domain2 at profile On 11/11/11 2:18 PM, "Michael Collins" wrote: > I believe the different domains share the same conference.conf.xml file, so > you'll need either different room numbers or different conference profile > names for each domain. > > -MC > > On Fri, Nov 11, 2011 at 3:07 AM, Peter P GMX wrote: >> >> Hello, >> >> I created 2 conferences >> >> * 1000 at domain1 >> * 1000 at doamin2 >> When I called >> >> * conference 1000 at domain1 from user 100 at domain1 and >> * conference 1000 at domain2 from user 100 at domain2 >> both users are in the same conference in my case. >> >> Question: Is 1000 at domain1 distinct from 1000 at domain2 or do I have to use >> seperate room numbers? >> >> Best regards >> Peter >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111111/9db19c89/attachment.html From acrow at integrafin.co.uk Sat Nov 12 00:46:07 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 11 Nov 2011 21:46:07 +0000 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: <82A6EE94-026E-44FD-9D33-586FB7077FC3@freeswitch.org> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> <82A6EE94-026E-44FD-9D33-586FB7077FC3@freeswitch.org> Message-ID: <4EBD979F.20100@integrafin.co.uk> Brian, I am *not* going out to/in from the world at large from/to the FS box via T38. I don't have a T38 capable gateway at the moment, and this is for an emergency box to replace a machine with several analog modems attached to the Mitel PBX. Therefore I have to translate t38modem's T38 I/O into audio over SIP to send to/rcv from our (non-T38-understanding) Mitel PBX which connects via E1 trunks to the PSTN. I know it's not ideal but it does seem to work OK so far. We have an older fax server (analog lines) where the modem does not support ECM. We virtually never have any problems sending or receiving faxes from this machine. However if you or anyone else can help get ECM working in this setup I'd be happy, but I have my doubts due to the G.711 step between the FS machine and the Mitel, probably causing jitter. To clarify, here is the topology: Hylafax<---Audio---->t38modem<---T38--->Freeswitch/t38gateway<----G.711/SIP--->Mitel 3300(T38 unsupported, grr)<---ISDN30---->PSTN I have to do the transcode as the Mitel is the weak link. If I could pop an E1 card in the FS box I'm sure things would work a lot better, but we don't have a spare card or indeed a spare circuit right now! Hope this make sense.... Alex On 11/11/11 16:18, Brian West wrote: > Disabling ECM by FAR the WRONG thing to do and you have this mindset > that is required its only required when you were hanging on by a > thread doing G.711 Faxing over the public internet and you didn't want > one tiny blip to screw the fax over. I still don't understand why > Alex is doing t.38 gateway from audio to t.38? Either way ECM works > fine never had any issues with it. > > /b > > On Nov 11, 2011, at 9:32 AM, Steve Underwood wrote: > >> Turning off ECM is equivalent to saying "I use FAX, but I couldn't give >> a damn if they are received". >> >> Steve >> >> >> On 11/11/2011 09:51 PM, Tihomir Culjaga wrote: >>> always disable ECM for fax calls over IP... its the 1st rule of thumb >>> for VoIP faxing :=) > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111111/2ee90325/attachment.html From acrow at integrafin.co.uk Sat Nov 12 00:50:05 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 11 Nov 2011 21:50:05 +0000 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EBD4012.7040802@coppice.org> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> Message-ID: <4EBD988D.3040506@integrafin.co.uk> On 11/11/11 15:32, Steve Underwood wrote: > Turning off ECM is equivalent to saying "I use FAX, but I couldn't give > a damn if they are received". > > Steve > Steve, From this experience it was the opposite - with ECM on, both ends seemed to be 100% happy that the fax transaction was completed, however the piece of paper only showed 5% of the image - on each page of a 2 page fax! See my last posting to list/Brian for the topology, and yes, I know it is not ideal. I think investing in a gateway may be the better solution. Who knows, maybe when we upgrade the Mitel it will have this functionality (for a license fee I strongly imagine!). Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From michal.bielicki at seventhsignal.de Sat Nov 12 02:20:10 2011 From: michal.bielicki at seventhsignal.de (Michal Bielicki) Date: Sat, 12 Nov 2011 00:20:10 +0100 Subject: [Freeswitch-users] call groups - sort order In-Reply-To: References: <5003D7D3E06F514E8C682F18D223265C04D3B36FA8@AZWSMS03.azwarranty.int> <5003D7D3E06F514E8C682F18D223265C04D3B36FBC@AZWSMS03.azwarranty.int> Message-ID: You can of course just add a order by to the statement in the source :) Am 11.11.2011 um 18:14 schrieb Michael Collins: > Stefan, > > I looked in mod_db.c and if I understand what I'm reading there isn't an explicit sort when pulling this data out of the database table. Here's the line in question: > > sql = switch_mprintf("select url,'%q' from group_data where groupname='%q'", how, argv[1]); > > I would recommend that you log in to sqlite3 and issue this command manually and see what you get. I don't know enough about SQLite to be able to tell you if it enforces any kind of order when doing a select on a table when there's no ORDER BY clause. > > -MC > > On Thu, Nov 10, 2011 at 11:23 PM, Weigel, Stefan wrote: > Hi Michael, > > > > http://wiki.freeswitch.org/wiki/Mod_db > > > > > > Best regards, > > > > Stefan > > > > > > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins > Gesendet: Donnerstag, 10. November 2011 19:24 > An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] call groups - sort order > > > > Which wiki page is this? > > -MC > > On Wed, Nov 9, 2011 at 12:58 AM, Weigel, Stefan wrote: > > Hi list, > > > > from the WIKI page: > > > > group call:groupname[:order] > > > > How are the entries sorted ? How can I make sure the list returned by ?group call? is sorted in a certain order ? I want to make sure a certain extension is ringed first, but sometimes another extension is listed first. > > > > > > Thanks and best regards > > > > Stefan > > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 Michal Bielicki Gesch?ftsf?hrer / CEO Seventh Signal Ltd. & Co. KG Weigandufer 45, B?ro 115, D-12059 Berlin Voice: +49 30 60988730 Amtsgericht Charlottenburg HRA 44413 B Ust.-ID: DE266981999 Gesch?ftsf?hrer: Michal Bielicki Pers?nlich Haftende Gesellschafterin: Seventh Signal Ltd, 69 Great Hampton St. Birmingham, B18 6EW, GB, Company Nr.: 06889439 WWW.: http://www.seventhsignal.de ---- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111112/9ca18cf7/attachment-0001.html From shouldbeq931 at gmail.com Sat Nov 12 03:20:51 2011 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Sat, 12 Nov 2011 00:20:51 +0000 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EBD979F.20100@integrafin.co.uk> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> <82A6EE94-026E-44FD-9D33-586FB7077FC3@freeswitch.org> <4EBD979F.20100@integrafin.co.uk> Message-ID: On Fri, Nov 11, 2011 at 9:46 PM, Alex Crow wrote: > Brian, > > I am *not* going out to/in from the world at large from/to the FS box via > T38. I don't have a T38 capable gateway at the moment, and this is for an > emergency box to replace a machine with several analog modems attached to > the Mitel PBX. > > Therefore I have to translate t38modem's T38 I/O into audio over SIP to send > to/rcv from our (non-T38-understanding) Mitel PBX which connects via E1 > trunks to the PSTN. > > I know it's not ideal but it does seem to work OK so far. > > We have an older fax server (analog lines) where the modem does not support > ECM. We virtually never have any problems sending or receiving faxes from > this machine. However if you or anyone else can help get ECM working in this > setup I'd be happy, but I have my doubts due to the G.711 step between the > FS machine and the Mitel, probably causing jitter. > > > To clarify, here is the topology: > > Hylafax<---Audio---->t38modem<---T38--->Freeswitch/t38gateway<----G.711/SIP--->Mitel > 3300(T38 unsupported, grr)<---ISDN30---->PSTN > > I have to do the transcode as the Mitel is the weak link. > > If I could pop an E1 card in the FS box I'm sure things would work a lot > better, but we don't have a spare card or indeed a spare circuit right now! > > Hope this make sense.... > > Alex > > I would use an analogue or PRI/BRI (depending on volume) card in the Mitel going to modems/dialogic card(s) in the Hylafax server. You might also want to look at IAX modem instead of T38 From acrow at integrafin.co.uk Sat Nov 12 12:36:42 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Sat, 12 Nov 2011 09:36:42 +0000 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> <82A6EE94-026E-44FD-9D33-586FB7077FC3@freeswitch.org> <4EBD979F.20100@integrafin.co.uk> Message-ID: <4EBE3E2A.6040104@integrafin.co.uk> > I would use an analogue or PRI/BRI (depending on volume) card in the > Mitel going to modems/dialogic card(s) in the Hylafax server. > > You might also want to look at IAX modem instead of T38 > shouldbe, We have an engineer coming in 2 weeks to do a full software upgrade on the Mitel. This means it should then be capable of t38! I will check before then if we need to add or upgrade the DSP. The reason we didn't get cards is that this is a stand-in solution should the ailing fax server go down, this gives us more time to arrange a permanent solution. We don't have any spare servers at the moment that will accommodate a modem card. The ill server (which currently answers about 6 fax numbers) and the main one (which has 16) use exactly what you say, they both have modems, in the first case 2 Multitech 8-port cards and in the second 16 external Multitech modems (unwieldy!) which connect to an Analog Service Unit on the Mitel. The ASU is I suppose what is called a "Channel Bank" in the SIP world. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From saeedahmad1981 at gmail.com Sat Nov 12 14:42:00 2011 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Sat, 12 Nov 2011 12:42:00 +0100 Subject: [Freeswitch-users] FreeSwitch as SBC In-Reply-To: <1554E94A-5A57-4033-9CC3-E42056D69A20@hovanetworks.com> References: <1554E94A-5A57-4033-9CC3-E42056D69A20@hovanetworks.com> Message-ID: http://wiki.freeswitch.org/wiki/SBC_Setup On Wed, Nov 9, 2011 at 5:20 PM, Adrian Fuentes < adrian.fuentes at hovanetworks.com> wrote: > hello all! > > can help describe what you can do FreeSwitch as SBC? > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111112/4c50521c/attachment.html From Prometheus001 at gmx.net Sat Nov 12 15:12:46 2011 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 12 Nov 2011 13:12:46 +0100 Subject: [Freeswitch-users] Conference and room numbers in a multi tenant environment In-Reply-To: References: Message-ID: <4EBE62BE.4080006@gmx.net> Thanks, I will do it that way then. Best regards Peter Am 11.11.2011 22:43, schrieb Ken Rice: > Conference 1000 at domain1 and conf 1000 at domain2 are not conferences in > different domains in freeswitch, the are conferences with different > conference profiles > > The easiest way to get what you want is to do 1000-domain1 at profile and > 1000-domain2 at profile > > > On 11/11/11 2:18 PM, "Michael Collins" wrote: > > I believe the different domains share the same conference.conf.xml > file, so you'll need either different room numbers or different > conference profile names for each domain. > > -MC > > On Fri, Nov 11, 2011 at 3:07 AM, Peter P GMX > wrote: > > > Hello, > > I created 2 conferences > > * 1000 at domain1 > * 1000 at doamin2 > > When I called > > * conference 1000 at domain1 from user 100 at domain1 and > * conference 1000 at domain2 from user 100 at domain2 > > both users are in the same conference in my case. > > Question: Is 1000 at domain1 distinct from 1000 at domain2 or do I > have to use seperate room numbers? > > Best regards > Peter > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111112/2cbec933/attachment.html From shouldbeq931 at gmail.com Sat Nov 12 17:18:53 2011 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Sat, 12 Nov 2011 14:18:53 +0000 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EBE3E2A.6040104@integrafin.co.uk> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> <82A6EE94-026E-44FD-9D33-586FB7077FC3@freeswitch.org> <4EBD979F.20100@integrafin.co.uk> <4EBE3E2A.6040104@integrafin.co.uk> Message-ID: On Sat, Nov 12, 2011 at 9:36 AM, Alex Crow wrote: > >> I would use an analogue or PRI/BRI (depending on volume) card in the >> Mitel going to modems/dialogic card(s) in the Hylafax server. >> >> You might also want to look at IAX modem instead of T38 >> > > shouldbe, > > We have an engineer coming in 2 weeks to do a full software upgrade on > the Mitel. This means it should then be capable of t38! > > I will check before then if we need to add or upgrade the DSP. > > The reason we didn't get cards is that this is a stand-in solution > should the ailing fax server go down, this gives us more time to arrange > a permanent solution. We don't have any spare servers at the moment that > will accommodate a modem card. > > The ill server (which currently answers about 6 fax numbers) and the > main one (which has 16) use exactly what you say, they both have modems, > in the first case 2 Multitech 8-port cards and in the second 16 external > Multitech modems (unwieldy!) which connect to an Analog Service Unit on > the Mitel. The ASU is I suppose what is called a "Channel Bank" in the > SIP world. > > Cheers > > Alex > Hi Alex, Although t38 would be one solution, I would tend to go with BRI/PRI dialogic cards in a Hylafax server connected to a trunk on the 3300. Because the "modems" on the Dialogic cards can get full information (the inbound CLI and the DDI that the number was "to") over the ISDN trunk, as well as being able to set the local identifier, the tagline format and the CLI, you can have a single BRI card replacing multiple modems. as a sanitised example for setting the local identifier on received calls... cat dynamic.sh #!/bin/sh RCALLID1=$2 case "$RCALLID1" in 3001) echo "LocalIdentifier: +44.10.12343001";; # New Main Number 3299) echo "LocalIdentifier: +44.10.12343299";; # team 1 Number 5900) echo "LocalIdentifier: Please_Use_New_Number";; # Old Main Number 5911) echo "LocalIdentifier: +44.10.12355911";; # team 2 Number 5947) echo "LocalIdentifier: +44.10.12355947";; # team 3 Number 6851) echo "LocalIdentifier: +44.10.12346851";; # team 4 Number 6911) echo "LocalIdentifier: +44.10.12346911";; # team 5 Number 6947) echo "LocalIdentifier: +44.10.12346947";; # team 6 Number 6951) echo "LocalIdentifier: +44.10.12346951";; # team 7 Number 6952) echo "LocalIdentifier: +44.10.12346952";; # team 8 Number 6991) echo "LocalIdentifier: +44.10.12346991";; # team 9 Number 6998) echo "LocalIdentifier: +44.10.12346998";; # team 10 Number esac It is called by adding "DynamicConfig: /var/spool/hylafax/etc/Dynamic.sh"to /var/spool/hylafax/etc/config.ttyds01 and /var/spool/hylafax/etc/config.ttyds02 The above was a single Dialogic BRI card that replaced 12 modems and fax machines just the other side of the Thames, granted the volume at that site is "low", but you can have multiple BRI cards in a single server and there are 2 and 4 port BRI cards available as well as PRI cards if you have "high" volumes. For inbound calls you can also build in HA by having one trunk on the Mitel side with multiple ports, and each port (or set of ports) going to a different Hylafax server. Although SIP is certainly the eventual way forward, I still prefer to keep fax on "legacy" systems :-) Cheers Arne From vetali100 at gmail.com Sun Nov 13 08:23:57 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Sat, 12 Nov 2011 21:23:57 -0800 Subject: [Freeswitch-users] new git - no more 'SIP auth challenge' messages in log In-Reply-To: <18097.1320948567@ccs.covici.com> References: <1320945234630-6982405.post@n2.nabble.com> <18097.1320948567@ccs.covici.com> Message-ID: So how this integration with fail2ban should be performed? Should we create a separate event_socket handler for listening SWITCH_EVENT_AUTH? And then log something into a file, which will be parsed by fail2ban? Or fail2ban is already able to listen such events? Could you please provide more details? Thank you, Vitalie 2011/11/10 > So what is the parameter to put those messages back to warning from > DEBUG10? > > Jeff Lenk wrote: > > > Please see http://jira.freeswitch.org/browse/FS-3094 > > > > for a discussion on this. > > > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/new-git-no-more-SIP-auth-challenge-messages-in-log-tp6980582p6982405.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 > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.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/20111112/88fe19e8/attachment.html From admin at blindi.net Sun Nov 13 10:41:29 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Sun, 13 Nov 2011 08:41:29 +0100 (CET) Subject: [Freeswitch-users] Anser confirmation via script? Message-ID: Hi all, I am a newbi, what lua perl and python are concerned. Has someone already managed to write a answerconfirmation? My approach: I like to setup a greate callscreening. For example: press 1 to accept the caller. press 2 to forward to voicemail. Press 3 to transfer the call to my cellphone. Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From hynek.cihlar at gmail.com Sun Nov 13 15:09:31 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sun, 13 Nov 2011 13:09:31 +0100 Subject: [Freeswitch-users] Expires ignored Message-ID: Freeswitch ignores the Expires header, instead registers SIP clients with expiration timeout of 90 seconds. I know there are few variables/parameters which control the Expiration. I use pretty default SIP profile configuration but the directory is built with with a lua script. If the lua script defines sip-force-expires, the registrations is ok forced to this value. If the variable sip-force-expires is missing, 90 seconds is used regardless what the client sends. Any ideas? Thanks for any help. Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111113/5b34beb6/attachment.html From craigesmith at gmail.com Sun Nov 13 19:46:23 2011 From: craigesmith at gmail.com (Craig Smith) Date: Sun, 13 Nov 2011 11:46:23 -0500 Subject: [Freeswitch-users] FreeSWITCH Message-ID: Some recent frustrations with Asterisk have me looking at FreeSWITCH, but it?s lack? of adoption leaves me wondering. Here are some questions I have as a long time Asterisk user, dCAP, and Astricon attendee. Why hasn?t FreeSWITCH reached a critical mass? Is the XML configuration that scary? What does Asterisk have that FreeSWITCH doesn?t? Does FreeSWITCH need better documentation and/or tutorials? Is it git? Would a FreeSWITCH certification help? I want to help the FreeSWITCH project, but I?m not developer or philanthropist , what can I do? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111113/03eb137b/attachment.html From b_ball_henry at hotmail.com Sun Nov 13 20:04:25 2011 From: b_ball_henry at hotmail.com (Henry Huang) Date: Mon, 14 Nov 2011 01:04:25 +0800 Subject: [Freeswitch-users] FreeSWITCH In-Reply-To: References: Message-ID: Nothing. It's just hard to change the way people used to do things. I'd say if we can use existing FreePBX (version 2) to control underlying FreeSWITCH, then people will be jumping ship. But, that's just for IP PBX side of business. Wholesale or dialer application developer has been adopting the FreeSWITCH solution instead of Asterisk in general. Which side of critical mass are you referring to? (IP PBX or?) Henry On Mon, Nov 14, 2011 at 12:46 AM, Craig Smith wrote: > Some recent frustrations with Asterisk have me looking at FreeSWITCH, but > it?s lack? of adoption leaves me wondering. > > Here are some questions I have as a long time Asterisk user, dCAP, and > Astricon attendee. > > > Why hasn?t FreeSWITCH reached a critical mass? > > Is the XML configuration that scary? > > What does Asterisk have that FreeSWITCH doesn?t? > > Does FreeSWITCH need better documentation and/or tutorials? > > Is it git? > > Would a FreeSWITCH certification help? > > > I want to help the FreeSWITCH project, but I?m not developer or > philanthropist , what can I do? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/c51b596b/attachment-0001.html From avi at avimarcus.net Sun Nov 13 20:44:55 2011 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 13 Nov 2011 19:44:55 +0200 Subject: [Freeswitch-users] FreeSWITCH In-Reply-To: References: Message-ID: a) How do you know? On what basis are you judging that? It does seem true that the average personal tinkerer is using Asterisk, but it seems many, many, many (most?) larger scale when choosing between * and FS choose FS. Some have legacy systems that they don't want to bother migrating, but new stuff goes on FS. b) That said, the do-it-for-me/hold-my-hand-the-entire-way group of people - there's more tutorials, free code, distributions with decent guis for Asterisk. Just take a look at pbx in a flash, it sounds like there's lots of people who use that. However.. if that's all they are, it's not really much of a help to the project to try to reach them. Unless you expect some of them to grow beyond the basic need and offer back. Other than "less" FOSS gui kind of stuff.. no, it seems FS pretty much has it all. Basically, for both a & b: there's lots of tintkerers out there using asterisk. I doubt that's a worthwhile metric to use. -Avi Marcus On Sun, Nov 13, 2011 at 7:04 PM, Henry Huang wrote: > Nothing. It's just hard to change the way people used to do things. I'd > say if we can use existing FreePBX (version 2) to control underlying > FreeSWITCH, then people will be jumping ship. > > But, that's just for IP PBX side of business. Wholesale or dialer > application developer has been adopting the FreeSWITCH solution instead of > Asterisk in general. > > Which side of critical mass are you referring to? (IP PBX or?) > > Henry > > On Mon, Nov 14, 2011 at 12:46 AM, Craig Smith wrote: > >> Some recent frustrations with Asterisk have me looking at FreeSWITCH, but >> it?s lack? of adoption leaves me wondering. >> >> Here are some questions I have as a long time Asterisk user, dCAP, and >> Astricon attendee. >> >> >> Why hasn?t FreeSWITCH reached a critical mass? >> >> Is the XML configuration that scary? >> >> What does Asterisk have that FreeSWITCH doesn?t? >> >> Does FreeSWITCH need better documentation and/or tutorials? >> >> Is it git? >> >> Would a FreeSWITCH certification help? >> >> >> I want to help the FreeSWITCH project, but I?m not developer or >> philanthropist , what can I do? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111113/937d7198/attachment.html From curriegrad2004 at gmail.com Sun Nov 13 20:48:12 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 13 Nov 2011 09:48:12 -0800 Subject: [Freeswitch-users] FreeSWITCH In-Reply-To: References: Message-ID: Really, FreeSWITCH isn't rocket science. It's usually the user itself makes it look like rocket science. On a serious side of things, if you can freaking code a webpage, you can use FreeSWITCH. XML and HTML aren't too distant from each other. If you can code in JS or .Net, you can even build custom modules for FreeSWITCH using the API. Heck, with the ODBC-SQL interfaces, you can even build an application revolving around FreeSWITCH by having your application to look at that database that FreeSWITCH uses for calls and what not. There's even a great book written for FreeSWITCH by the authors. On Sun, Nov 13, 2011 at 9:04 AM, Henry Huang wrote: > Nothing. It's just hard to change the way people used to do things. I'd say > if we can use existing FreePBX (version 2) to control underlying FreeSWITCH, > then people will be jumping ship. > > But, that's just for IP PBX side of business. Wholesale or dialer > application developer has been adopting the FreeSWITCH solution instead of > Asterisk in general. > > Which side of critical mass are you referring to? (IP PBX or?) > > Henry > > On Mon, Nov 14, 2011 at 12:46 AM, Craig Smith wrote: >> >> Some recent frustrations with Asterisk have me looking at FreeSWITCH, but >> it?s lack? of adoption leaves me wondering. >> >> Here are some questions I have as a long time Asterisk user, dCAP, and >> Astricon attendee. >> >> >> Why hasn?t FreeSWITCH reached a critical mass? >> >> Is the XML configuration that scary? >> >> What does Asterisk have that FreeSWITCH doesn?t? >> >> Does FreeSWITCH need better documentation and/or tutorials? >> >> Is it git? >> >> Would a FreeSWITCH certification help? >> >> >> I want to help the FreeSWITCH project, but I?m not developer or >> philanthropist , what can I do? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 spencer at 5ninesolutions.com Sun Nov 13 23:05:23 2011 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Sun, 13 Nov 2011 12:05:23 -0800 Subject: [Freeswitch-users] Adjust nat-options-ping interval Message-ID: Hello all, Is there any way to adjust the interval of the nat-options-ping parameter? All of our endpoints are behind NAT using TCP and 30 seconds is a little short as pretty much every router I can think of has a TCP timeout greater than that. I realize that this is a very specific change so if someone could point me to the right place to look I can patch this for our use. Thanks! Spencer From spencer at 5ninesolutions.com Sun Nov 13 23:45:24 2011 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Sun, 13 Nov 2011 12:45:24 -0800 Subject: [Freeswitch-users] Expires ignored In-Reply-To: References: Message-ID: <40983149-72A8-4E32-8EBF-5062FDDDD741@5ninesolutions.com> Hynek, It looks like the Expires value is hard coded at 360 seconds for non NAT endpoints and 90 seconds for endpoints behind NAT. It might be nice to have a parameter to define this on a per profile basis. You could have a parameter to define the default and also NAT expiration times. I'll look into this. Thanks, Spencer On Nov 13, 2011, at 4:09 AM, Hynek Cihlar wrote: > Freeswitch ignores the Expires header, instead registers SIP clients with expiration timeout of 90 seconds. > > I know there are few variables/parameters which control the Expiration. I use pretty default SIP profile configuration but the directory is built with with a lua script. If the lua script defines sip-force-expires, the registrations is ok forced to this value. If the variable sip-force-expires is missing, 90 seconds is used regardless what the client sends. Any ideas? > > Thanks for any help. > > Hynek > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 henrikaagaardsorensen at gmail.com Mon Nov 14 00:48:26 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Sun, 13 Nov 2011 22:48:26 +0100 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. Message-ID: I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc. Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer? I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111113/282e111f/attachment-0001.html From moy at sangoma.com Mon Nov 14 02:06:50 2011 From: moy at sangoma.com (Moises Silva) Date: Sun, 13 Nov 2011 18:06:50 -0500 Subject: [Freeswitch-users] FreeTDM PRI tapping issue In-Reply-To: References: Message-ID: On Wed, Nov 9, 2011 at 2:28 AM, kiruthika sri wrote: > Resolved SEGFAULT by changing the following line, > > crv = tap_pri_get_crv(pritap->pri, e->ring.call) > > into > > crv = tap_pri_get_crv(pritap->pri, e->proceeding.call) > > under the case PRI_EVENT_PROGRESS from ftmod_pritap.c source. > > Fix committed. Thanks! *Moises Silva* *Software Engineer, Development Manager*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube VegaStream is now part of Sangoma! Ask us about both Gateway Appliances and Internal Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111113/62069fc9/attachment.html From ga at steadfasttelecom.com Mon Nov 14 03:18:20 2011 From: ga at steadfasttelecom.com (Gilad Abada) Date: Sun, 13 Nov 2011 19:18:20 -0500 Subject: [Freeswitch-users] Applying a patch to freeswitch Message-ID: Hi All Can anyone tell me how to apply a patch to freeswitch? Thanks Gill From freeswitch-list at puzzled.xs4all.nl Mon Nov 14 04:03:09 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Mon, 14 Nov 2011 02:03:09 +0100 Subject: [Freeswitch-users] FreeSWITCH In-Reply-To: References: Message-ID: <4EC068CD.2040304@puzzled.xs4all.nl> Craig, Welcome to the FreeSWITCH community. Comments inline. On 11/13/2011 05:46 PM, Craig Smith wrote: > Some recent frustrations with Asterisk have me looking at FreeSWITCH, > but it?s lack? of adoption leaves me wondering. Lack of adoption? How do you come to that conclusion? FreeSWITCH may not be as visible as Asterisk but it would not surprise me if in the ITSP world FreeSWITCH handled vastly more minutes than Asterisk. Admittedly FreeSWITCH is an excellent solution for an ITSP. > Here are some questions I have as a long time Asterisk user, dCAP, and > Astricon attendee. To meet the FreeSWITCH developers and community visit ClueCon. For more info check out: http://www.cluecon.com/ > Why hasn?t FreeSWITCH reached a critical mass? Define critical mass. There are quite a few ITSPs offering phone services which are powered by FreeSWITCH (amongst other things). You may even be using one without knowing it. > Is the XML configuration that scary? XML is about as scary as HTML 0.1. > What does Asterisk have that FreeSWITCH doesn?t? Deadlocks :-) > Does FreeSWITCH need better documentation and/or tutorials? There is always room for improvement of the Wiki. The same goes for writing tutorials. > Is it git? Yes, the SCM system used is git. http://fisheye.freeswitch.org/ > Would a FreeSWITCH certification help? Help what? > I want to help the FreeSWITCH project, but I?m not developer or > philanthropist , what can I do? Write tutorials, update/improve the wiki. If you succeed in building a succesfull business based on FreeSWITCH then you could give back e.g. by funding bounties to be developed by the FreeSWITCH core developers. Have fun setting up your first FreeSWITCH box. Afaik the FreeSWITCH core developers use CentOS so you can safely use that but ubuntu or debian should work fine too. It helps if you use git master because if you think you have found a bug and report it, the developers will ask if you were able to recreate it with git master. So you might as well be using that in the first place. Here is how to get git master: $ git clone git://git.freeswitch.org/freeswitch.git To get started go here: http://wiki.freeswitch.org/wiki/Main_Page Asterisk to FreeSWITCH Rosetta Stone: http://wiki.freeswitch.org/wiki/Rosetta_stone Highly recommended: buy the FreeSWITCH book https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book If you are interested in a GUI to manage a FreeSWITCH box check out: http://www.fusionpbx.com/ http://www.2600hz.org/ (Blue.Box) For a quick question the #freeswitch irc channel on irc.freenode.net is an excellent place. There is also a weekly conference where various things are discussed. The conference and topic(s) are announced on the mailing list. Enjoy! Regards, Patrick From henrikaagaardsorensen at gmail.com Mon Nov 14 04:14:03 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Mon, 14 Nov 2011 02:14:03 +0100 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: References: Message-ID: Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up. 2011/11/13 Henrik Aagaard S?rensen > I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. > > I've setup Kamailio via this: > http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html > > I've installed FreeSwitch from scratch on Ubuntu via: > http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start > > Now, when registering extensions via Kamailio Dispatcher I'm able to call > to FreeSwitch and listen to hold music. But that's it. I'm not able to call > between extensions etc. > > Can anyone help me setting up FreeSwitch to accept registration, calls > etc. from Kamailio and everything else that is needed to use FreeSwitch > behind a load balancer? > > I'm very new to FreeSwitch, but I'm trying to use the terminal (without > any GUI etc.) as I want the installation to be as clean as possible. So I > would prefer very precise help, as I'm still getting hold of FreeSwitch. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/033a61b7/attachment-0001.html -------------- next part -------------- interface: any filter: (ip or ip6) and ( port 5060 ) U 2011/11/13 17:08:25.538159 76.33.21.131:5060 -> 96.224.14.164:41264 NOTIFY sip:1001 at 96.224.14.164:41264;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKvm2Nyj3QSX0QH. Max-Forwards: 70. From: ;tag=9SQgDBSa41QaK. To: . Call-ID: f12a354d-88ff-122f-3d82-61e75d58280c. CSeq: 20280377 NOTIFY. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: message-summary. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=timeout. Content-Type: application/simple-message-summary. Content-Length: 63. . Messages-Waiting: no. Message-Account: sip:1001 at 76.33.21.131. . U 2011/11/13 17:08:27.258159 76.33.21.131:5060 -> 96.224.14.164:43975 NOTIFY sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKXXUe0DmUp6paD. Max-Forwards: 70. From: ;tag=cN3tjvBNUvt2N. To: . Call-ID: f489cf67-88ff-122f-3d82-61e75d58280c. CSeq: 20280380 NOTIFY. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: message-summary. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=timeout. Content-Type: application/simple-message-summary. Content-Length: 63. . Messages-Waiting: no. Message-Account: sip:1002 at 76.33.21.131. . U 2011/11/13 17:08:28.612571 216.231.104.54:5060 -> 76.33.21.131:5060 INVITE sip:1002 at sip.my-domain.com SIP/2.0. Record-Route: . Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKffcd.39c460a6.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjz2o9W9ZIszjxkGRP5zEUPD3srEHz8bXN. Max-Forwards: 69. From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: . Contact: . Call-ID: QgPGCRQNy6s01p9.zXA2udA2rGuPF89p. CSeq: 22751 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: Bria iPhone 1.3.4. Content-Type: application/sdp. Content-Length: 319. . v=0. o=- 3530221708 3530221708 IN IP4 10.0.1.3. s=cpc_med. c=IN IP4 10.0.1.3. t=0 0. m=audio 4002 RTP/AVP 120 0 8 104 3 96. a=rtpmap:120 SILK/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:104 iLBC/8000. a=fmtp:104 mode=30. a=rtpmap:3 GSM/8000. a=sendrecv. a=rtpmap:96 telephone-event/8000. a=fmtp:96 0-15. U 2011/11/13 17:08:28.613047 76.33.21.131:5060 -> 216.231.104.54:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKffcd.39c460a6.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjz2o9W9ZIszjxkGRP5zEUPD3srEHz8bXN. Record-Route: . From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: . Call-ID: QgPGCRQNy6s01p9.zXA2udA2rGuPF89p. CSeq: 22751 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Content-Length: 0. . U 2011/11/13 17:08:28.613798 76.33.21.131:5060 -> 216.231.104.54:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKffcd.39c460a6.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjz2o9W9ZIszjxkGRP5zEUPD3srEHz8bXN. From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: ;tag=DyvKmQvrr5gNH. Call-ID: QgPGCRQNy6s01p9.zXA2udA2rGuPF89p. CSeq: 22751 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Proxy-Authenticate: Digest realm="sip.my-domain.com", nonce="2956e2ae-0e5d-11e1-ae5f-ef58a163b248", algorithm=MD5, qop="auth". Content-Length: 0. . U 2011/11/13 17:08:28.614147 216.231.104.54:5060 -> 76.33.21.131:5060 ACK sip:1002 at sip.my-domain.com SIP/2.0. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKffcd.39c460a6.0. Max-Forwards: 69. From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: ;tag=DyvKmQvrr5gNH. Call-ID: QgPGCRQNy6s01p9.zXA2udA2rGuPF89p. CSeq: 22751 ACK. Content-Length: 0. . U 2011/11/13 17:08:28.788249 216.231.104.54:5060 -> 76.33.21.131:5060 INVITE sip:1002 at sip.my-domain.com SIP/2.0. Record-Route: . Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKcfcd.3c288a3.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjXqQE4ca7HrI0tOo8umTYyui8DbpSPjDQ. Max-Forwards: 69. From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: . Contact: . Call-ID: QgPGCRQNy6s01p9.zXA2udA2rGuPF89p. CSeq: 22752 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: Bria iPhone 1.3.4. Proxy-Authorization: Digest username="1001", realm="sip.my-domain.com", nonce="2956e2ae-0e5d-11e1-ae5f-ef58a163b248", uri="sip:1002 at sip.my-domain.com", response="f45d4fc87316361d7fcf0d262fdbf9d2", algorithm=MD5, cnonce="NeU9IElhK8q.1bNseRW2lM0vLHbHle3i", qop=auth, nc=00000001. Content-Type: application/sdp. Content-Length: 319. . v=0. o=- 3530221708 3530221708 IN IP4 10.0.1.3. s=cpc_med. c=IN IP4 10.0.1.3. t=0 0. m=audio 4002 RTP/AVP 120 0 8 104 3 96. a=rtpmap:120 SILK/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:104 iLBC/8000. a=fmtp:104 mode=30. a=rtpmap:3 GSM/8000. a=sendrecv. a=rtpmap:96 telephone-event/8000. a=fmtp:96 0-15. U 2011/11/13 17:08:28.788575 76.33.21.131:5060 -> 216.231.104.54:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKcfcd.3c288a3.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjXqQE4ca7HrI0tOo8umTYyui8DbpSPjDQ. Record-Route: . From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: . Call-ID: QgPGCRQNy6s01p9.zXA2udA2rGuPF89p. CSeq: 22752 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Content-Length: 0. . U 2011/11/13 17:08:28.803526 76.33.21.131:5060 -> 96.224.14.164:43975 INVITE sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKy6m7184yKFDXr. Max-Forwards: 68. From: "Extension 1001" ;tag=FgF5QDyZjQXtr. To: . Call-ID: 00d2223d-8900-122f-3d82-61e75d58280c. CSeq: 20280390 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 311. X-FS-Support: update_display. Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1321206960 1321206961 IN IP4 76.33.21.131. s=FreeSWITCH. c=IN IP4 76.33.21.131. t=0 0. m=audio 25948 RTP/AVP 0 98 99 9 8 3 101 13. a=rtpmap:98 G7221/32000. a=fmtp:98 bitrate=48000. a=rtpmap:99 G7221/16000. a=fmtp:99 bitrate=32000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/11/13 17:08:29.538255 76.33.21.131:5060 -> 96.224.14.164:41264 NOTIFY sip:1001 at 96.224.14.164:41264;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKvm2Nyj3QSX0QH. Max-Forwards: 70. From: ;tag=9SQgDBSa41QaK. To: . Call-ID: f12a354d-88ff-122f-3d82-61e75d58280c. CSeq: 20280377 NOTIFY. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: message-summary. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=timeout. Content-Type: application/simple-message-summary. Content-Length: 63. . Messages-Waiting: no. Message-Account: sip:1001 at 76.33.21.131. . U 2011/11/13 17:08:29.805308 76.33.21.131:5060 -> 96.224.14.164:43975 INVITE sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKy6m7184yKFDXr. Max-Forwards: 68. From: "Extension 1001" ;tag=FgF5QDyZjQXtr. To: . Call-ID: 00d2223d-8900-122f-3d82-61e75d58280c. CSeq: 20280390 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 311. X-FS-Support: update_display. Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1321206960 1321206961 IN IP4 76.33.21.131. s=FreeSWITCH. c=IN IP4 76.33.21.131. t=0 0. m=audio 25948 RTP/AVP 0 98 99 9 8 3 101 13. a=rtpmap:98 G7221/32000. a=fmtp:98 bitrate=48000. a=rtpmap:99 G7221/16000. a=fmtp:99 bitrate=32000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/11/13 17:08:31.258314 76.33.21.131:5060 -> 96.224.14.164:43975 NOTIFY sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKXXUe0DmUp6paD. Max-Forwards: 70. From: ;tag=cN3tjvBNUvt2N. To: . Call-ID: f489cf67-88ff-122f-3d82-61e75d58280c. CSeq: 20280380 NOTIFY. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: message-summary. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=timeout. Content-Type: application/simple-message-summary. Content-Length: 63. . Messages-Waiting: no. Message-Account: sip:1002 at 76.33.21.131. . U 2011/11/13 17:08:31.806231 76.33.21.131:5060 -> 96.224.14.164:43975 INVITE sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKy6m7184yKFDXr. Max-Forwards: 68. From: "Extension 1001" ;tag=FgF5QDyZjQXtr. To: . Call-ID: 00d2223d-8900-122f-3d82-61e75d58280c. CSeq: 20280390 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 311. X-FS-Support: update_display. Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1321206960 1321206961 IN IP4 76.33.21.131. s=FreeSWITCH. c=IN IP4 76.33.21.131. t=0 0. m=audio 25948 RTP/AVP 0 98 99 9 8 3 101 13. a=rtpmap:98 G7221/32000. a=fmtp:98 bitrate=48000. a=rtpmap:99 G7221/16000. a=fmtp:99 bitrate=32000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/11/13 17:08:33.538239 76.33.21.131:5060 -> 96.224.14.164:41264 NOTIFY sip:1001 at 96.224.14.164:41264;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKvm2Nyj3QSX0QH. Max-Forwards: 70. From: ;tag=9SQgDBSa41QaK. To: . Call-ID: f12a354d-88ff-122f-3d82-61e75d58280c. CSeq: 20280377 NOTIFY. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: message-summary. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=timeout. Content-Type: application/simple-message-summary. Content-Length: 63. . Messages-Waiting: no. Message-Account: sip:1001 at 76.33.21.131. . U 2011/11/13 17:08:35.258167 76.33.21.131:5060 -> 96.224.14.164:43975 NOTIFY sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKXXUe0DmUp6paD. Max-Forwards: 70. From: ;tag=cN3tjvBNUvt2N. To: . Call-ID: f489cf67-88ff-122f-3d82-61e75d58280c. CSeq: 20280380 NOTIFY. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: message-summary. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=timeout. Content-Type: application/simple-message-summary. Content-Length: 63. . Messages-Waiting: no. Message-Account: sip:1002 at 76.33.21.131. . U 2011/11/13 17:08:35.806188 76.33.21.131:5060 -> 96.224.14.164:43975 INVITE sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKy6m7184yKFDXr. Max-Forwards: 68. From: "Extension 1001" ;tag=FgF5QDyZjQXtr. To: . Call-ID: 00d2223d-8900-122f-3d82-61e75d58280c. CSeq: 20280390 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 311. X-FS-Support: update_display. Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1321206960 1321206961 IN IP4 76.33.21.131. s=FreeSWITCH. c=IN IP4 76.33.21.131. t=0 0. m=audio 25948 RTP/AVP 0 98 99 9 8 3 101 13. a=rtpmap:98 G7221/32000. a=fmtp:98 bitrate=48000. a=rtpmap:99 G7221/16000. a=fmtp:99 bitrate=32000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/11/13 17:08:39.258171 76.33.21.131:5060 -> 96.224.14.164:43975 NOTIFY sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKXXUe0DmUp6paD. Max-Forwards: 70. From: ;tag=cN3tjvBNUvt2N. To: . Call-ID: f489cf67-88ff-122f-3d82-61e75d58280c. CSeq: 20280380 NOTIFY. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: message-summary. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=timeout. Content-Type: application/simple-message-summary. Content-Length: 63. . Messages-Waiting: no. Message-Account: sip:1002 at 76.33.21.131. . U 2011/11/13 17:08:43.806190 76.33.21.131:5060 -> 96.224.14.164:43975 INVITE sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKy6m7184yKFDXr. Max-Forwards: 68. From: "Extension 1001" ;tag=FgF5QDyZjQXtr. To: . Call-ID: 00d2223d-8900-122f-3d82-61e75d58280c. CSeq: 20280390 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 311. X-FS-Support: update_display. Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1321206960 1321206961 IN IP4 76.33.21.131. s=FreeSWITCH. c=IN IP4 76.33.21.131. t=0 0. m=audio 25948 RTP/AVP 0 98 99 9 8 3 101 13. a=rtpmap:98 G7221/32000. a=fmtp:98 bitrate=48000. a=rtpmap:99 G7221/16000. a=fmtp:99 bitrate=32000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/11/13 17:08:58.024383 76.33.21.131:5060 -> 216.231.104.54:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKcfcd.3c288a3.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjXqQE4ca7HrI0tOo8umTYyui8DbpSPjDQ. Record-Route: . From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: ;tag=e7NcpjDvNe77c. Call-ID: QgPGCRQNy6s01p9.zXA2udA2rGuPF89p. CSeq: 22752 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Require: timer. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Session-Expires: 1800;refresher=uac. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 242. Remote-Party-ID: "1002" ;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1321202252 1321202253 IN IP4 76.33.21.131. s=FreeSWITCH. c=IN IP4 76.33.21.131. t=0 0. m=audio 30686 RTP/AVP 0 96. a=rtpmap:0 PCMU/8000. a=rtpmap:96 telephone-event/8000. a=fmtp:96 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 2011/11/13 17:08:58.097468 216.231.104.54:5060 -> 76.33.21.131:5060 ACK sip:1002 at 76.33.21.131:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjBMYVzyIOOsbglBEU8pTgcN5S1e1.bfou. Max-Forwards: 69. From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: ;tag=e7NcpjDvNe77c. Call-ID: QgPGCRQNy6s01p9.zXA2udA2rGuPF89p. CSeq: 22752 ACK. Content-Length: 0. . U 2011/11/13 17:08:59.806501 76.33.21.131:5060 -> 96.224.14.164:43975 INVITE sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKy6m7184yKFDXr. Max-Forwards: 68. From: "Extension 1001" ;tag=FgF5QDyZjQXtr. To: . Call-ID: 00d2223d-8900-122f-3d82-61e75d58280c. CSeq: 20280390 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 311. X-FS-Support: update_display. Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1321206960 1321206961 IN IP4 76.33.21.131. s=FreeSWITCH. c=IN IP4 76.33.21.131. t=0 0. m=audio 25948 RTP/AVP 0 98 99 9 8 3 101 13. a=rtpmap:98 G7221/32000. a=fmtp:98 bitrate=48000. a=rtpmap:99 G7221/16000. a=fmtp:99 bitrate=32000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/11/13 17:09:01.838355 216.231.104.54:5060 -> 76.33.21.131:5060 BYE sip:1002 at 76.33.21.131:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKdfcd.059bee14.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPj2sCKtKo7Sqy1bvngdk9RA1SFb4q5CZZc. Max-Forwards: 69. From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: ;tag=e7NcpjDvNe77c. Call-ID: QgPGCRQNy6s01p9.zXA2udA2rGuPF89p. CSeq: 22753 BYE. User-Agent: Bria iPhone 1.3.4. Content-Length: 0. . U 2011/11/13 17:09:01.846248 76.33.21.131:5060 -> 216.231.104.54:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKdfcd.059bee14.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPj2sCKtKo7Sqy1bvngdk9RA1SFb4q5CZZc. From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: ;tag=e7NcpjDvNe77c. Call-ID: QgPGCRQNy6s01p9.zXA2udA2rGuPF89p. CSeq: 22753 BYE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Content-Length: 0. . From govoiper at gmail.com Mon Nov 14 07:56:33 2011 From: govoiper at gmail.com (Sammy Govind) Date: Mon, 14 Nov 2011 09:56:33 +0500 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: References: Message-ID: Hi, If everything is setup as you expected it then don't read this > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy 2011/11/14 Henrik Aagaard S?rensen > Hi everyone. In regards to my earlier question regarding with FreeSwitch > behind Kamailio Dispatcher, I've attached a call from extension 1001 to > 1002, which fails. It just hangs for some time and then says that 1002 > cannot be found, and then the voicemail for it comes up. > > 2011/11/13 Henrik Aagaard S?rensen > >> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. >> >> I've setup Kamailio via this: >> http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >> >> I've installed FreeSwitch from scratch on Ubuntu via: >> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >> >> Now, when registering extensions via Kamailio Dispatcher I'm able to call >> to FreeSwitch and listen to hold music. But that's it. I'm not able to call >> between extensions etc. >> >> Can anyone help me setting up FreeSwitch to accept registration, calls >> etc. from Kamailio and everything else that is needed to use FreeSwitch >> behind a load balancer? >> >> I'm very new to FreeSwitch, but I'm trying to use the terminal (without >> any GUI etc.) as I want the installation to be as clean as possible. So I >> would prefer very precise help, as I'm still getting hold of FreeSwitch. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/a4a7dc7c/attachment.html From henrikaagaardsorensen at gmail.com Mon Nov 14 08:08:11 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Mon, 14 Nov 2011 06:08:11 +0100 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: References: Message-ID: Hi Sammy. I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch. Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later. My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch. But when going through Kamailio Dispatcher it fails between the extensions. So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it. My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything. On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind wrote: > Hi, > > If everything is setup as you expected it then don't read this > > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb > , > else follow the article from start till end and implement it. Don't worry > about the name asterisk, just replace that all with FreeSWICTH. It will > work like that too. > > Once that is implemented then apply dispatcher module on the route which > says [REGFWD] or [TOASTERISK]. > > Also explain your topology abit more and features required as well that > may help in telling which extra module you will require in order to make > things work. > > In your current implementation are you sure SIP phones are registering on > FS and not on Kamailio?and that both end points making call to each other > are on same FS? > -- > Regards, > Sammy > > > 2011/11/14 Henrik Aagaard S?rensen > >> Hi everyone. In regards to my earlier question regarding with FreeSwitch >> behind Kamailio Dispatcher, I've attached a call from extension 1001 to >> 1002, which fails. It just hangs for some time and then says that 1002 >> cannot be found, and then the voicemail for it comes up. >> >> 2011/11/13 Henrik Aagaard S?rensen >> >>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. >>> >>> I've setup Kamailio via this: >>> http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >>> >>> I've installed FreeSwitch from scratch on Ubuntu via: >>> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >>> >>> Now, when registering extensions via Kamailio Dispatcher I'm able to >>> call to FreeSwitch and listen to hold music. But that's it. I'm not able to >>> call between extensions etc. >>> >>> Can anyone help me setting up FreeSwitch to accept registration, calls >>> etc. from Kamailio and everything else that is needed to use FreeSwitch >>> behind a load balancer? >>> >>> I'm very new to FreeSwitch, but I'm trying to use the terminal (without >>> any GUI etc.) as I want the installation to be as clean as possible. So I >>> would prefer very precise help, as I'm still getting hold of FreeSwitch. >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/1be1a6dd/attachment.html From govoiper at gmail.com Mon Nov 14 08:26:35 2011 From: govoiper at gmail.com (Sammy Govind) Date: Mon, 14 Nov 2011 10:26:35 +0500 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: References: Message-ID: Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy 2011/11/14 Henrik Aagaard S?rensen > Hi Sammy. > > I've actually removed registration and presence from Kamailio, so all it > does is dispatch everything to FreeSwitch. > > Currently I only have 1 FreeSwitch, for testing this basic setup. > Next move would be 2 FreeSwitch with 1 common database etc. But that's > later. > > My FreeSwitch is the basic setup, without anything else. Just as the > installation manuel is written. So I have my extension 1000 - 1019 etc. And > calls between them works when connected directly to FreeSwitch. > > But when going through Kamailio Dispatcher it fails between the extensions. > > So I guess there should be some more setup in FreeSwitch when using a load > balancer (dispatcher) in front of it. > > My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle > everything. > > > On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind wrote: > >> Hi, >> >> If everything is setup as you expected it then don't read this > >> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb >> , >> else follow the article from start till end and implement it. Don't worry >> about the name asterisk, just replace that all with FreeSWICTH. It will >> work like that too. >> >> Once that is implemented then apply dispatcher module on the route which >> says [REGFWD] or [TOASTERISK]. >> >> Also explain your topology abit more and features required as well that >> may help in telling which extra module you will require in order to make >> things work. >> >> In your current implementation are you sure SIP phones are registering on >> FS and not on Kamailio?and that both end points making call to each other >> are on same FS? >> -- >> Regards, >> Sammy >> >> >> 2011/11/14 Henrik Aagaard S?rensen >> >>> Hi everyone. In regards to my earlier question regarding with FreeSwitch >>> behind Kamailio Dispatcher, I've attached a call from extension 1001 to >>> 1002, which fails. It just hangs for some time and then says that 1002 >>> cannot be found, and then the voicemail for it comes up. >>> >>> 2011/11/13 Henrik Aagaard S?rensen >>> >>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. >>>> >>>> I've setup Kamailio via this: >>>> http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >>>> >>>> I've installed FreeSwitch from scratch on Ubuntu via: >>>> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >>>> >>>> Now, when registering extensions via Kamailio Dispatcher I'm able to >>>> call to FreeSwitch and listen to hold music. But that's it. I'm not able to >>>> call between extensions etc. >>>> >>>> Can anyone help me setting up FreeSwitch to accept registration, calls >>>> etc. from Kamailio and everything else that is needed to use FreeSwitch >>>> behind a load balancer? >>>> >>>> I'm very new to FreeSwitch, but I'm trying to use the terminal (without >>>> any GUI etc.) as I want the installation to be as clean as possible. So I >>>> would prefer very precise help, as I'm still getting hold of FreeSwitch. >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/e5db23e9/attachment-0001.html From henrikaagaardsorensen at gmail.com Mon Nov 14 08:40:34 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Mon, 14 Nov 2011 00:40:34 -0500 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: References: Message-ID: <-7236397912894270049@unknownmsgid> The handling of several FS are not a issue, actually easy enough. I would like to have as little load on Kamailio as possible, as it just should load balance. Also, having to handle users on both Kamailio and FS makes unecessary work loads. On 14/11/2011, at 00.32, Sammy Govind wrote: Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy 2011/11/14 Henrik Aagaard S?rensen > Hi Sammy. > > I've actually removed registration and presence from Kamailio, so all it > does is dispatch everything to FreeSwitch. > > Currently I only have 1 FreeSwitch, for testing this basic setup. > Next move would be 2 FreeSwitch with 1 common database etc. But that's > later. > > My FreeSwitch is the basic setup, without anything else. Just as the > installation manuel is written. So I have my extension 1000 - 1019 etc. And > calls between them works when connected directly to FreeSwitch. > > But when going through Kamailio Dispatcher it fails between the extensions. > > So I guess there should be some more setup in FreeSwitch when using a load > balancer (dispatcher) in front of it. > > My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle > everything. > > > On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind wrote: > >> Hi, >> >> If everything is setup as you expected it then don't read this > >> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb >> , >> else follow the article from start till end and implement it. Don't worry >> about the name asterisk, just replace that all with FreeSWICTH. It will >> work like that too. >> >> Once that is implemented then apply dispatcher module on the route which >> says [REGFWD] or [TOASTERISK]. >> >> Also explain your topology abit more and features required as well that >> may help in telling which extra module you will require in order to make >> things work. >> >> In your current implementation are you sure SIP phones are registering on >> FS and not on Kamailio?and that both end points making call to each other >> are on same FS? >> -- >> Regards, >> Sammy >> >> >> 2011/11/14 Henrik Aagaard S?rensen >> >>> Hi everyone. In regards to my earlier question regarding with FreeSwitch >>> behind Kamailio Dispatcher, I've attached a call from extension 1001 to >>> 1002, which fails. It just hangs for some time and then says that 1002 >>> cannot be found, and then the voicemail for it comes up. >>> >>> 2011/11/13 Henrik Aagaard S?rensen >>> >>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. >>>> >>>> I've setup Kamailio via this: >>>> http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >>>> >>>> I've installed FreeSwitch from scratch on Ubuntu via: >>>> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >>>> >>>> Now, when registering extensions via Kamailio Dispatcher I'm able to >>>> call to FreeSwitch and listen to hold music. But that's it. I'm not able to >>>> call between extensions etc. >>>> >>>> Can anyone help me setting up FreeSwitch to accept registration, calls >>>> etc. from Kamailio and everything else that is needed to use FreeSwitch >>>> behind a load balancer? >>>> >>>> I'm very new to FreeSwitch, but I'm trying to use the terminal (without >>>> any GUI etc.) as I want the installation to be as clean as possible. So I >>>> would prefer very precise help, as I'm still getting hold of FreeSwitch. >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111114/cb19850b/attachment.html From lakindia89 at gmail.com Mon Nov 14 09:18:18 2011 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 14 Nov 2011 11:48:18 +0530 Subject: [Freeswitch-users] Anser confirmation via script? In-Reply-To: References: Message-ID: Try group_confirm variables On Sun, Nov 13, 2011 at 1:11 PM, Thomas Hoellriegel wrote: > Hi all, > I am a newbi, what lua perl and python are concerned. > Has someone already managed to write a answerconfirmation? > My approach: > I like to setup a greate callscreening. > For example: press 1 to accept the caller. > press 2 to forward to voicemail. > Press 3 to transfer the call to my cellphone. > > Thanks. > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/24f4e3bc/attachment.html From miha at softnet.si Mon Nov 14 10:24:06 2011 From: miha at softnet.si (Miha Zoubek) Date: Mon, 14 Nov 2011 08:24:06 +0100 Subject: [Freeswitch-users] freeswitch radiusclinet (mod_radius_cdr) problems In-Reply-To: <4EBCD686.9040305@softnet.si> References: <4EBA8B5E.2050900@softnet.si> <4EBCD686.9040305@softnet.si> Message-ID: <4EC0C216.3040806@softnet.si> Hi, can anyone can help me with this issue? BR, Miha On 11/11/2011 9:02 AM, Miha Zoubek wrote: > Hi @ Ognjen, > > bellow you can see that my Accounting-Request(4), > Accounting-Response(5) are twice send. > > Any idea? > > BR, > Miha > > > > 71.449050 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS > Accounting-Request(4) (id=235, l=265) > 71.517347 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS > Accounting-Response(5) (id=235, l=20) > 73.536126 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Request(1) > (id=236, l=210) > 73.567412 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Accept(2) > (id=236, l=20) > 73.572794 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS > Accounting-Request(4) (id=237, l=321) > 73.574156 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS > Accounting-Response(5) (id=237, l=20) > 83.482760 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS > Accounting-Request(4) (id=238, l=401) > 83.485670 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS > Accounting-Response(5) (id=238, l=20) > 83.514594 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS > Accounting-Request(4) (id=239, l=402) > 83.516404 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS > Accounting-Response(5) (id=239, l=20) > > On 11/10/2011 6:24 PM, Ognjen Seslija wrote: >> Radius clients generally send Start and Stop records, I guess this is >> why you get two records. >> >> On Wed, Nov 9, 2011 at 3:17 PM, Miha Zoubek > > wrote: >> >> Hi, >> >> radiusclient on freeswitch in sending more that once same >> response. So I >> am getting in my sql tables for every call two inputs which is wrong. >> How can I deal whit this issue? >> >> I have paste a log from freeradius server in pastebin that you >> can see >> what is freeswitch sending. >> >> http://pastebin.freeswitch.org/17730 >> >> >> Thank you! >> >> BR, >> Miha >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/1eff23ec/attachment-0001.html From tculjaga at gmail.com Mon Nov 14 14:24:54 2011 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 14 Nov 2011 12:24:54 +0100 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EBD4012.7040802@coppice.org> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> Message-ID: On Fri, Nov 11, 2011 at 4:32 PM, Steve Underwood wrote: > Turning off ECM is equivalent to saying "I use FAX, but I couldn't give > a damn if they are received". > and it always work if the network is stable (you might loose a pixel here and there...)! If you are running it via open internet, sorry but try something else (T.38 or switch to e-mail). You cannot expect pass-through to work reliably in such scenarios. T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/a5606c2f/attachment.html From hynek.cihlar at gmail.com Mon Nov 14 15:07:28 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Mon, 14 Nov 2011 13:07:28 +0100 Subject: [Freeswitch-users] Expires ignored In-Reply-To: <40983149-72A8-4E32-8EBF-5062FDDDD741@5ninesolutions.com> References: <40983149-72A8-4E32-8EBF-5062FDDDD741@5ninesolutions.com> Message-ID: Spencer, thanks for the effort. I will simply use a reasonable default as a workaround. Hynek On Sun, Nov 13, 2011 at 9:45 PM, Spencer Thomason < spencer at 5ninesolutions.com> wrote: > Hynek, > It looks like the Expires value is hard coded at 360 seconds for non NAT > endpoints and 90 seconds for endpoints behind NAT. It might be nice to > have a parameter to define this on a per profile basis. You could have a > parameter to define the default and also NAT expiration times. I'll look > into this. > > Thanks, > Spencer > > > On Nov 13, 2011, at 4:09 AM, Hynek Cihlar wrote: > > > Freeswitch ignores the Expires header, instead registers SIP clients > with expiration timeout of 90 seconds. > > > > I know there are few variables/parameters which control the Expiration. > I use pretty default SIP profile configuration but the directory is built > with with a lua script. If the lua script defines sip-force-expires, the > registrations is ok forced to this value. If the variable sip-force-expires > is missing, 90 seconds is used regardless what the client sends. Any ideas? > > > > Thanks for any help. > > > > Hynek > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/33f5f5f2/attachment.html From admin at blindi.net Mon Nov 14 15:54:56 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Mon, 14 Nov 2011 13:54:56 +0100 (CET) Subject: [Freeswitch-users] http://latest.freeswitch.org/ not found Message-ID: Hi all, i read the install-instuction from: http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Source_Tarball Why is this link http://latest.freeswitch.org/ not working? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From x.liu at hw.ac.uk Mon Nov 14 16:07:11 2011 From: x.liu at hw.ac.uk (xl127) Date: Mon, 14 Nov 2011 13:07:11 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <4EBD70A9.6090505@hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> Message-ID: <4EC1127F.2000504@hw.ac.uk> Hi, The first problem I need to solve is that FS stucks when I speak during the prompt is playing. It looks like the recognizer is not listening any more. It works fine if I speak after the playing prompt finishes. I tried Nuance and PocketSphinx recognizers and got same problem. Any idea about what the possible causes are? Thanks! Xing On 11/11/11 18:59, xl127 wrote: > Hello, > > I found this is a question that was asked before by others, but I didn't > find the answer. > > Anyway, I am using FS ESL outbound mode connecting to my IVR app, using > FS's "speak" > and "detect_speech" to access Nuance MRCP V1 server. > > I want the user to be able to barge-in during the system's speaking. > How could I implement it? > > I tried to specify "kill-on-barge-in=true" in the mrcp config profile. > The barge-in doesn't work. > > With or without setting kill-on-barge-in, FS stops responding to my > phone call and > eventually it hangs up my call if I speak somthing (do the barge-in) > during the system's speaking. > > I made a turn-by-turn loop in my app, the ASR/TTS works fine if I do not > do barge-in ( I wait until the TTS finishes then I start to speak) > > Any advices please? > > Thanks! > Xing > > > -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 From cmrienzo at gmail.com Mon Nov 14 16:23:24 2011 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Mon, 14 Nov 2011 08:23:24 -0500 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <4EC1127F.2000504@hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> Message-ID: I'll write up a dialplan app this week to deal with this. Getting tired of being asked the same question over and over since it's too complicated as is currently designed :) On Mon, Nov 14, 2011 at 8:07 AM, xl127 wrote: > Hi, > > The first problem I need to solve is that FS stucks when I speak during > the prompt is playing. > It looks like the recognizer is not listening any more. It works fine if > I speak after the playing prompt finishes. > > I tried Nuance and PocketSphinx recognizers and got same problem. > > Any idea about what the possible causes are? > > Thanks! > Xing > > > On 11/11/11 18:59, xl127 wrote: > > Hello, > > > > I found this is a question that was asked before by others, but I didn't > > find the answer. > > > > Anyway, I am using FS ESL outbound mode connecting to my IVR app, using > > FS's "speak" > > and "detect_speech" to access Nuance MRCP V1 server. > > > > I want the user to be able to barge-in during the system's speaking. > > How could I implement it? > > > > I tried to specify "kill-on-barge-in=true" in the mrcp config profile. > > The barge-in doesn't work. > > > > With or without setting kill-on-barge-in, FS stops responding to my > > phone call and > > eventually it hangs up my call if I speak somthing (do the barge-in) > > during the system's speaking. > > > > I made a turn-by-turn loop in my app, the ASR/TTS works fine if I do not > > do barge-in ( I wait until the TTS finishes then I start to speak) > > > > Any advices please? > > > > Thanks! > > Xing > > > > > > > > > > -- > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > Heriot-Watt University is the Sunday Times > Scottish University of the Year 2011-2012 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/d912557d/attachment.html From freeswitch-list at puzzled.xs4all.nl Mon Nov 14 16:29:12 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Mon, 14 Nov 2011 14:29:12 +0100 Subject: [Freeswitch-users] http://latest.freeswitch.org/ not found In-Reply-To: References: Message-ID: <4EC117A8.7050609@puzzled.xs4all.nl> On 11/14/2011 01:54 PM, Thomas Hoellriegel wrote: > Hi all, i read the install-instuction from: > http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Source_Tarball > Why is this link > http://latest.freeswitch.org/ > not working? Not sure but just use git master instead: $ git clone git://git.freeswitch.org/freeswitch.git Regards, Patrick From x.liu at hw.ac.uk Mon Nov 14 17:19:26 2011 From: x.liu at hw.ac.uk (xl127) Date: Mon, 14 Nov 2011 14:19:26 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> Message-ID: <4EC1236E.3090402@hw.ac.uk> Hi Christopher, That will be great! Thanks and looking forward to your app! By the way, do you have a quick thoughts about the reasons the FS stucks when garge-in occurs? Otherwise if it is not straightforward to explain it, just leave it alone until your new app comes out. Cheers, Xing On 14/11/11 13:23, Christopher Rienzo wrote: > I'll write up a dialplan app this week to deal with this. Getting > tired of being asked the same question over and over since it's too > complicated as is currently designed :) > > > > > On Mon, Nov 14, 2011 at 8:07 AM, xl127 > wrote: > > Hi, > > The first problem I need to solve is that FS stucks when I speak > during > the prompt is playing. > It looks like the recognizer is not listening any more. It works > fine if > I speak after the playing prompt finishes. > > I tried Nuance and PocketSphinx recognizers and got same problem. > > Any idea about what the possible causes are? > > Thanks! > Xing > > > On 11/11/11 18:59, xl127 wrote: > > Hello, > > > > I found this is a question that was asked before by others, but > I didn't > > find the answer. > > > > Anyway, I am using FS ESL outbound mode connecting to my IVR > app, using > > FS's "speak" > > and "detect_speech" to access Nuance MRCP V1 server. > > > > I want the user to be able to barge-in during the system's speaking. > > How could I implement it? > > > > I tried to specify "kill-on-barge-in=true" in the mrcp config > profile. > > The barge-in doesn't work. > > > > With or without setting kill-on-barge-in, FS stops responding to my > > phone call and > > eventually it hangs up my call if I speak somthing (do the barge-in) > > during the system's speaking. > > > > I made a turn-by-turn loop in my app, the ASR/TTS works fine if > I do not > > do barge-in ( I wait until the TTS finishes then I start to speak) > > > > Any advices please? > > > > Thanks! > > Xing > > > > > > > > > > -- > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > Heriot-Watt University is the Sunday Times > Scottish University of the Year 2011-2012 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/12754b0b/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/12754b0b/attachment-0001.jpg From cmrienzo at gmail.com Mon Nov 14 17:48:01 2011 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Mon, 14 Nov 2011 09:48:01 -0500 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <4EC1236E.3090402@hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> Message-ID: I'm not sure why it locks up for you. It works for me and I know of at least one other developer that got it all to work. The way I do it in my custom APP is to set {start-input-timers=false} in the speech recognition request and register an input callback function to deal with the recognition and DTMF events. The input callback reacts to the start of speech and recognition complete events by stopping the prompt. At start of speech, the input timers are started and silence is played until the final result arrives from the recognizer (no match, match, etc). An alternative way to do this is to watch for the recognition events over ESL and react to them. On Mon, Nov 14, 2011 at 9:19 AM, xl127 wrote: > Hi Christopher, > > That will be great! Thanks and looking forward to your app! > > By the way, do you have a quick thoughts about the reasons the FS stucks > when garge-in occurs? > Otherwise if it is not straightforward to explain it, just leave it alone > until your new app comes out. > > Cheers, > Xing > > > > > On 14/11/11 13:23, Christopher Rienzo wrote: > > I'll write up a dialplan app this week to deal with this. Getting tired > of being asked the same question over and over since it's too complicated > as is currently designed :) > > > > > On Mon, Nov 14, 2011 at 8:07 AM, xl127 wrote: > >> Hi, >> >> The first problem I need to solve is that FS stucks when I speak during >> the prompt is playing. >> It looks like the recognizer is not listening any more. It works fine if >> I speak after the playing prompt finishes. >> >> I tried Nuance and PocketSphinx recognizers and got same problem. >> >> Any idea about what the possible causes are? >> >> Thanks! >> Xing >> >> >> On 11/11/11 18:59, xl127 wrote: >> > Hello, >> > >> > I found this is a question that was asked before by others, but I didn't >> > find the answer. >> > >> > Anyway, I am using FS ESL outbound mode connecting to my IVR app, using >> > FS's "speak" >> > and "detect_speech" to access Nuance MRCP V1 server. >> > >> > I want the user to be able to barge-in during the system's speaking. >> > How could I implement it? >> > >> > I tried to specify "kill-on-barge-in=true" in the mrcp config profile. >> > The barge-in doesn't work. >> > >> > With or without setting kill-on-barge-in, FS stops responding to my >> > phone call and >> > eventually it hangs up my call if I speak somthing (do the barge-in) >> > during the system's speaking. >> > >> > I made a turn-by-turn loop in my app, the ASR/TTS works fine if I do not >> > do barge-in ( I wait until the TTS finishes then I start to speak) >> > >> > Any advices please? >> > >> > Thanks! >> > Xing >> > >> > >> > >> >> >> >> -- >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> Heriot-Watt University is the Sunday Times >> Scottish University of the Year 2011-2012 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > ------------------------------ > > [image: Scottish University of the Year 2011-12] *Heriot-Watt > University is the Sunday Times > > Scottish University of the Year 2011-2012 > * > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/35283242/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/35283242/attachment.jpe From leon at scarlet-internet.nl Mon Nov 14 18:18:29 2011 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 14 Nov 2011 16:18:29 +0100 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: <-7236397912894270049@unknownmsgid> References: <-7236397912894270049@unknownmsgid> Message-ID: <71430C91-82A1-4130-94FC-E404F1C6AC57@scarlet-internet.nl> Hi, I'm actually trying to configure the same thing right now - with opensips though, but (afaik) it uses the same dispatcher module. First I used a simple route script in opensips with using dispatcher, but after the first message (from ua through proxy to fs), the proxy would get out of the signaling path, while I want it to stay in. To fix that, I added record routing in the proxy configuration so it stays in the path. Registrations are also balanced towards the fs servers - but some clients are nat'ed so the contact header was wrong - fix_nated_contact() in route and in onreply_route fixed that. To get calls originating from fs to go through the proxy before going to the client, I tried: originate sofia/some_profile/sip:user at mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 &park which worked. Then I found out about the sip_route_uri variable that can be set in the bridge string: originate {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user at mydomain.com &park which seemed to do the same thing (does anyone know if there's any difference between the two variants ?) Then I tried putting the sip_route_uri in the dial-string param of the domain (or user), for example: so now it's possible to just call the user as: originate user/some_user at mydomain.com &park nice ! And then I found out that you can just set an outbound-proxy param in the sip profile that's used for originating: so now you can leave out the sip_route_uri or fs_path variables, it just works. My proxy config still needs a lot of work, but I got the basic functionality working, still need to find out about using acls on fs side based on a sip header added by the proxy - I'm already adding X-AUTH-IP headers on messages from clients towards fs which should do the trick, but I didn't test it yet. Anyway, would love to read some configs of ppl who successfuly setup opensips/kamailio/openser/... proxies in front of fs, what their experience is with using record-routing or not, etc. regards, Leon On Nov 14, 2011, at 6:40 AM, Henrik Aagaard S?rensen wrote: > The handling of several FS are not a issue, actually easy enough. > > I would like to have as little load on Kamailio as possible, as it just should load balance. > > Also, having to handle users on both Kamailio and FS makes unecessary work loads. > > On 14/11/2011, at 00.32, Sammy Govind wrote: > >> Hi again, >> >> Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. >> This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) >> >> Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. >> >> By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! >> >> -- >> Regards, >> Sammy >> >> 2011/11/14 Henrik Aagaard S?rensen >> Hi Sammy. >> >> I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch. >> >> Currently I only have 1 FreeSwitch, for testing this basic setup. >> Next move would be 2 FreeSwitch with 1 common database etc. But that's later. >> >> My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch. >> >> But when going through Kamailio Dispatcher it fails between the extensions. >> >> So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it. >> >> My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything. >> >> >> On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind wrote: >> Hi, >> >> If everything is setup as you expected it then don't read this > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, >> else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. >> >> Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. >> >> Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. >> >> In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? >> -- >> Regards, >> Sammy >> >> >> 2011/11/14 Henrik Aagaard S?rensen >> Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up. >> >> 2011/11/13 Henrik Aagaard S?rensen >> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. >> >> I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >> >> I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >> >> Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc. >> >> Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer? >> >> I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/ee287702/attachment-0001.html From jeff at jefflenk.com Mon Nov 14 19:01:03 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 14 Nov 2011 08:01:03 -0800 (PST) Subject: [Freeswitch-users] Applying a patch to freeswitch In-Reply-To: References: Message-ID: <1321286463479-6992971.post@n2.nabble.com> If you wish to contribute a change back to FS open an account at jira.freeswitch.org and attach your patch for review(with all needed commentary). -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Applying-a-patch-to-freeswitch-tp6991064p6992971.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vipkilla at gmail.com Mon Nov 14 19:08:17 2011 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 14 Nov 2011 11:08:17 -0500 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: <71430C91-82A1-4130-94FC-E404F1C6AC57@scarlet-internet.nl> References: <-7236397912894270049@unknownmsgid> <71430C91-82A1-4130-94FC-E404F1C6AC57@scarlet-internet.nl> Message-ID: Any chance you can post your opensips config? I've been struggling with mixed success trying same setup. On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij wrote: > Hi, > I'm actually trying to configure the same thing right now - with opensips > though, but (afaik) it uses the same dispatcher module. > First I used a simple route script in opensips with using dispatcher, but > after the first message (from ua through proxy to fs), the proxy would get > out of the signaling path, while I want it to stay in. > To fix that, I added record routing in the proxy configuration so it stays > in the path. > Registrations are also balanced towards the fs servers - but some clients > are nat'ed so the contact header was wrong - fix_nated_contact() in route > and in ?onreply_route fixed that. > To get calls originating from fs to go through the proxy before going to the > client, I tried: > originate > sofia/some_profile/sip:user at mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 > &park > which worked.?Then I found out about the sip_route_uri variable that can be > set in the bridge string: > originate > {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user at mydomain.com > &park > which seemed to do the same thing (does anyone know if there's any > difference between the two variants ?) > Then I tried putting the sip_route_uri in the dial-string param of the > domain (or user), for example: > value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> > so now it's possible to just call the user as: > originate user/some_user at mydomain.com &park > nice ! > And then I found out that you can just set an outbound-proxy param in the > sip profile that's used for originating: > > so now you can leave out the sip_route_uri or fs_path variables, it just > works. > My proxy config still needs a lot of work, but I got the basic functionality > working, still need to find out about using acls on fs side based on a sip > header added by the proxy - I'm already adding X-AUTH-IP headers on messages > from clients towards fs which should do the trick, but I didn't test it yet. > Anyway, would love to read some configs of ppl who successfuly setup > opensips/kamailio/openser/... proxies in front of fs, what their experience > is with using record-routing or not, etc. > regards, > Leon > > > > On Nov 14, 2011, at 6:40 AM, Henrik Aagaard S?rensen wrote: > > The handling of several FS are not a issue, actually easy enough. > I would like to have as little load on Kamailio as possible, as it just > should load balance. > Also, having to handle users on both Kamailio and FS makes unecessary work > loads. > > On 14/11/2011, at 00.32, Sammy Govind wrote: > > Hi again, > Why don't you just let Kamailio handle?registrations. Anyway I was thinking > about that LBing the registration would result in such a scenario that > calling one extension to another would not make a?successful?call because > the other?endpoint?maybe?registered?on some other FS. > This may further lead you to making a dial-plan which would work somewhat > like DUNDI but it'd just have to search all the FS servers before joining a > call.(Obv there are other intelligent approaches to minimize the headache) > Try following the link I sent you and implement that in?front?of your FS, I > think that Kamailio configuration is so well written that anyone can start > understanding kamailio and implement such setups with little effort. > By explaining your topology I meant how do you plan to use Kamailio in front > of FS? i.e Kamailio on Public IP and all FS on private IPs and etc ?as in a > topo-hiding or SBC like setup! > -- > Regards, > Sammy > > 2011/11/14 Henrik Aagaard S?rensen >> >> Hi Sammy. >> I've actually removed registration and presence from Kamailio, so all it >> does is dispatch everything to FreeSwitch. >> Currently I only have 1 FreeSwitch, for testing this basic setup. >> Next move would be 2 FreeSwitch with 1 common database etc. But that's >> later. >> My FreeSwitch is the basic setup, without anything else. Just as the >> installation manuel is written. So I have my extension 1000 - 1019 etc. And >> calls between them works when connected directly to FreeSwitch. >> But when going through Kamailio Dispatcher it fails between the >> extensions. >> So I guess there should be some more setup in FreeSwitch when using a load >> balancer (dispatcher) in front of it. >> My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle >> everything. >> >> On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind wrote: >>> >>> Hi, >>> If everything is setup as you expected it then don't read this >>> >?http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, >>> else follow the article from start till end and implement it. Don't worry >>> about the name asterisk, just replace that all with FreeSWICTH. It will work >>> like that too. >>> Once that is implemented then apply dispatcher module on the route which >>> says [REGFWD] or [TOASTERISK]. >>> Also explain your topology abit more ?and features required as well that >>> may help in telling which extra module you will require in order to make >>> things work. >>> In your current implementation are you sure SIP phones are registering on >>> FS and not on Kamailio?and that both end points making call to each other >>> are on same FS? >>> -- >>> Regards, >>> Sammy >>> >>> 2011/11/14 Henrik Aagaard S?rensen >>>> >>>> Hi everyone. In regards to my earlier question regarding with FreeSwitch >>>> behind Kamailio Dispatcher, I've attached a call from extension 1001 to >>>> 1002, which fails. It just hangs for some time and then says that 1002 >>>> cannot be found, and then the voicemail for it comes up. >>>> 2011/11/13 Henrik Aagaard S?rensen >>>>> >>>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. >>>>> I've setup Kamailio via >>>>> this:?http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >>>>> I've installed FreeSwitch from scratch on Ubuntu >>>>> via:?http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >>>>> Now, when registering extensions via Kamailio Dispatcher I'm able to >>>>> call to FreeSwitch and listen to hold music. But that's it. I'm not able to >>>>> call between extensions etc. >>>>> Can anyone help me setting up FreeSwitch to accept registration, calls >>>>> etc. from Kamailio and everything else that is needed to use FreeSwitch >>>>> behind a load balancer? >>>>> I'm very new to FreeSwitch, but I'm trying to use the terminal (without >>>>> any GUI etc.) as I want the installation to be as clean as possible. So I >>>>> would prefer very precise help, as I'm still getting hold of FreeSwitch. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 aasishp at gmail.com Mon Nov 14 02:38:28 2011 From: aasishp at gmail.com (Aasish Pappu) Date: Sun, 13 Nov 2011 18:38:28 -0500 Subject: [Freeswitch-users] Cannot receive CUSTOM event in Python Message-ID: Hi All I am writing a python app using it with the flex rtmp client. The javascript function testevent( ) does fire an event that shows up on fs_cli as mod_rtmp receives it. How do I catch this event from python? I am using a input_callback method to handle events. The testevent never gets there, however DTMF events fired using the flex keypad do reach. What should I do? I also tried adding this line to my dialplan I found out the Subclass of this event by calling DUMP_EVENT from the sendevent function inside the mod/endpoints/mod_rtmp/rtmp_sig.c Thanks -- Aasish Pappu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111113/791edd5c/attachment.html From hugh at irvine.com.au Mon Nov 14 12:30:34 2011 From: hugh at irvine.com.au (Hugh Irvine) Date: Mon, 14 Nov 2011 20:30:34 +1100 Subject: [Freeswitch-users] freeswitch radiusclinet (mod_radius_cdr) problems In-Reply-To: <4EC0C216.3040806@softnet.si> References: <4EBA8B5E.2050900@softnet.si> <4EBCD686.9040305@softnet.si> <4EC0C216.3040806@softnet.si> Message-ID: Hello Miha - You are probably getting a RADIUS accounting request for each call leg. The id's in the requests are different, as are the lengths, but you really need to check the contents of the requests to see exactly what they contain. regards Hugh On 14 Nov 2011, at 18:24, Miha Zoubek wrote: > Hi, > > can anyone can help me with this issue? > > > BR, > Miha > > On 11/11/2011 9:02 AM, Miha Zoubek wrote: >> Hi @ Ognjen, >> >> bellow you can see that my Accounting-Request(4), Accounting-Response(5) are twice send. >> >> Any idea? >> >> BR, >> Miha >> >> >> >> 71.449050 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=235, l=265) >> 71.517347 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=235, l=20) >> 73.536126 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Request(1) (id=236, l=210) >> 73.567412 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Accept(2) (id=236, l=20) >> 73.572794 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=237, l=321) >> 73.574156 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=237, l=20) >> 83.482760 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=238, l=401) >> 83.485670 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=238, l=20) >> 83.514594 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=239, l=402) >> 83.516404 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=239, l=20) >> >> On 11/10/2011 6:24 PM, Ognjen Seslija wrote: >>> Radius clients generally send Start and Stop records, I guess this is why you get two records. >>> >>> On Wed, Nov 9, 2011 at 3:17 PM, Miha Zoubek wrote: >>> Hi, >>> >>> radiusclient on freeswitch in sending more that once same response. So I >>> am getting in my sql tables for every call two inputs which is wrong. >>> How can I deal whit this issue? >>> >>> I have paste a log from freeradius server in pastebin that you can see >>> what is freeswitch sending. >>> >>> http://pastebin.freeswitch.org/17730 >>> >>> >>> Thank you! >>> >>> BR, >>> Miha >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 mstockton at harqen.com Mon Nov 14 19:18:04 2011 From: mstockton at harqen.com (Matt Stockton) Date: Mon, 14 Nov 2011 10:18:04 -0600 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: References: <-7236397912894270049@unknownmsgid> <71430C91-82A1-4130-94FC-E404F1C6AC57@scarlet-internet.nl> Message-ID: I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa wrote: > Any chance you can post your opensips config? I've been struggling > with mixed success trying same setup. > > On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij > wrote: > > Hi, > > I'm actually trying to configure the same thing right now - with opensips > > though, but (afaik) it uses the same dispatcher module. > > First I used a simple route script in opensips with using dispatcher, but > > after the first message (from ua through proxy to fs), the proxy would > get > > out of the signaling path, while I want it to stay in. > > To fix that, I added record routing in the proxy configuration so it > stays > > in the path. > > Registrations are also balanced towards the fs servers - but some clients > > are nat'ed so the contact header was wrong - fix_nated_contact() in route > > and in onreply_route fixed that. > > To get calls originating from fs to go through the proxy before going to > the > > client, I tried: > > originate > > sofia/some_profile/sip:user at mydomain.com > ;fs_path=sip:ip_of_sip_proxy:5060 > > &park > > which worked. Then I found out about the sip_route_uri variable that can > be > > set in the bridge string: > > originate > > {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/ > sip:some_user at mydomain.com > > &park > > which seemed to do the same thing (does anyone know if there's any > > difference between the two variants ?) > > Then I tried putting the sip_route_uri in the dial-string param of the > > domain (or user), for example: > > > > value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@ > ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> > > so now it's possible to just call the user as: > > originate user/some_user at mydomain.com &park > > nice ! > > And then I found out that you can just set an outbound-proxy param in the > > sip profile that's used for originating: > > > > so now you can leave out the sip_route_uri or fs_path variables, it just > > works. > > My proxy config still needs a lot of work, but I got the basic > functionality > > working, still need to find out about using acls on fs side based on a > sip > > header added by the proxy - I'm already adding X-AUTH-IP headers on > messages > > from clients towards fs which should do the trick, but I didn't test it > yet. > > Anyway, would love to read some configs of ppl who successfuly setup > > opensips/kamailio/openser/... proxies in front of fs, what their > experience > > is with using record-routing or not, etc. > > regards, > > Leon > > > > > > > > On Nov 14, 2011, at 6:40 AM, Henrik Aagaard S?rensen wrote: > > > > The handling of several FS are not a issue, actually easy enough. > > I would like to have as little load on Kamailio as possible, as it just > > should load balance. > > Also, having to handle users on both Kamailio and FS makes unecessary > work > > loads. > > > > On 14/11/2011, at 00.32, Sammy Govind wrote: > > > > Hi again, > > Why don't you just let Kamailio handle registrations. Anyway I was > thinking > > about that LBing the registration would result in such a scenario that > > calling one extension to another would not make a successful call because > > the other endpoint maybe registered on some other FS. > > This may further lead you to making a dial-plan which would work somewhat > > like DUNDI but it'd just have to search all the FS servers before > joining a > > call.(Obv there are other intelligent approaches to minimize the > headache) > > Try following the link I sent you and implement that in front of your > FS, I > > think that Kamailio configuration is so well written that anyone can > start > > understanding kamailio and implement such setups with little effort. > > By explaining your topology I meant how do you plan to use Kamailio in > front > > of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as > in a > > topo-hiding or SBC like setup! > > -- > > Regards, > > Sammy > > > > 2011/11/14 Henrik Aagaard S?rensen > >> > >> Hi Sammy. > >> I've actually removed registration and presence from Kamailio, so all it > >> does is dispatch everything to FreeSwitch. > >> Currently I only have 1 FreeSwitch, for testing this basic setup. > >> Next move would be 2 FreeSwitch with 1 common database etc. But that's > >> later. > >> My FreeSwitch is the basic setup, without anything else. Just as the > >> installation manuel is written. So I have my extension 1000 - 1019 etc. > And > >> calls between them works when connected directly to FreeSwitch. > >> But when going through Kamailio Dispatcher it fails between the > >> extensions. > >> So I guess there should be some more setup in FreeSwitch when using a > load > >> balancer (dispatcher) in front of it. > >> My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle > >> everything. > >> > >> On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind > wrote: > >>> > >>> Hi, > >>> If everything is setup as you expected it then don't read this > >>> > > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb > , > >>> else follow the article from start till end and implement it. Don't > worry > >>> about the name asterisk, just replace that all with FreeSWICTH. It > will work > >>> like that too. > >>> Once that is implemented then apply dispatcher module on the route > which > >>> says [REGFWD] or [TOASTERISK]. > >>> Also explain your topology abit more and features required as well > that > >>> may help in telling which extra module you will require in order to > make > >>> things work. > >>> In your current implementation are you sure SIP phones are registering > on > >>> FS and not on Kamailio?and that both end points making call to each > other > >>> are on same FS? > >>> -- > >>> Regards, > >>> Sammy > >>> > >>> 2011/11/14 Henrik Aagaard S?rensen > >>>> > >>>> Hi everyone. In regards to my earlier question regarding with > FreeSwitch > >>>> behind Kamailio Dispatcher, I've attached a call from extension 1001 > to > >>>> 1002, which fails. It just hangs for some time and then says that 1002 > >>>> cannot be found, and then the voicemail for it comes up. > >>>> 2011/11/13 Henrik Aagaard S?rensen > >>>>> > >>>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to > work. > >>>>> I've setup Kamailio via > >>>>> this: > http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html > >>>>> I've installed FreeSwitch from scratch on Ubuntu > >>>>> via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start > >>>>> Now, when registering extensions via Kamailio Dispatcher I'm able to > >>>>> call to FreeSwitch and listen to hold music. But that's it. I'm not > able to > >>>>> call between extensions etc. > >>>>> Can anyone help me setting up FreeSwitch to accept registration, > calls > >>>>> etc. from Kamailio and everything else that is needed to use > FreeSwitch > >>>>> behind a load balancer? > >>>>> I'm very new to FreeSwitch, but I'm trying to use the terminal > (without > >>>>> any GUI etc.) as I want the installation to be as clean as possible. > So I > >>>>> would prefer very precise help, as I'm still getting hold of > FreeSwitch. > >>>> > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.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 > >>>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.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 > >>> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.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 > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/39f8fa8d/attachment-0001.html From x.liu at hw.ac.uk Mon Nov 14 20:06:26 2011 From: x.liu at hw.ac.uk (xl127) Date: Mon, 14 Nov 2011 17:06:26 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> Message-ID: <4EC14A92.1040704@hw.ac.uk> The way you did and described here is similar to what I wanted. I want to set start-input-timers=false in the initial setup, then start the recognition timer once the TTS finishes playing the prompt. I do not use input callback as I am using ESL event (Java ESL Client 0.9.2). I've not yet figured out how to know the prompt playing has finished, how to stop the TTS and how to start the input timer. I didn't get many events when I did barge-in: only got two CHANNEL_EXECUTE and CAHNNEL_EXECUTE_COMPLETE events. so didn't get the DETECTED_SPEECH at this point. In principle it shouldn't stop FS generating events when barging in, so there must be somethings wrong in my setup. From Examples_directory_lua_asr_tts in the wiki, it mentions the session needs to sleep for a few seconds before getting the results. I tried but it didn't help. On 14/11/11 14:48, Christopher Rienzo wrote: > I'm not sure why it locks up for you. It works for me and I know of > at least one other developer that got it all to work. > > The way I do it in my custom APP is to set {start-input-timers=false} > in the speech recognition request and register an input callback > function to deal with the recognition and DTMF events. The input > callback reacts to the start of speech and recognition complete events > by stopping the prompt. At start of speech, the input timers are > started and silence is played until the final result arrives from the > recognizer (no match, match, etc). > > An alternative way to do this is to watch for the recognition events > over ESL and react to them. > > > > On Mon, Nov 14, 2011 at 9:19 AM, xl127 > wrote: > > Hi Christopher, > > That will be great! Thanks and looking forward to your app! > > By the way, do you have a quick thoughts about the reasons the FS > stucks when garge-in occurs? > Otherwise if it is not straightforward to explain it, just leave > it alone until your new app comes out. > > Cheers, > Xing > > > > > On 14/11/11 13:23, Christopher Rienzo wrote: >> I'll write up a dialplan app this week to deal with this. >> Getting tired of being asked the same question over and over >> since it's too complicated as is currently designed :) >> >> >> >> >> On Mon, Nov 14, 2011 at 8:07 AM, xl127 > > wrote: >> >> Hi, >> >> The first problem I need to solve is that FS stucks when I >> speak during >> the prompt is playing. >> It looks like the recognizer is not listening any more. It >> works fine if >> I speak after the playing prompt finishes. >> >> I tried Nuance and PocketSphinx recognizers and got same problem. >> >> Any idea about what the possible causes are? >> >> Thanks! >> Xing >> >> >> On 11/11/11 18:59, xl127 wrote: >> > Hello, >> > >> > I found this is a question that was asked before by others, >> but I didn't >> > find the answer. >> > >> > Anyway, I am using FS ESL outbound mode connecting to my >> IVR app, using >> > FS's "speak" >> > and "detect_speech" to access Nuance MRCP V1 server. >> > >> > I want the user to be able to barge-in during the system's >> speaking. >> > How could I implement it? >> > >> > I tried to specify "kill-on-barge-in=true" in the mrcp >> config profile. >> > The barge-in doesn't work. >> > >> > With or without setting kill-on-barge-in, FS stops >> responding to my >> > phone call and >> > eventually it hangs up my call if I speak somthing (do the >> barge-in) >> > during the system's speaking. >> > >> > I made a turn-by-turn loop in my app, the ASR/TTS works >> fine if I do not >> > do barge-in ( I wait until the TTS finishes then I start to >> speak) >> > >> > Any advices please? >> > >> > Thanks! >> > Xing >> > >> > >> > >> >> >> >> -- >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> Heriot-Watt University is the Sunday Times >> Scottish University of the Year 2011-2012 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > ------------------------------------------------------------------------ > > Scottish University of the Year 2011-12 *Heriot-Watt University is > the Sunday Times > > Scottish University of the Year 2011-2012 > * > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/017947b6/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/017947b6/attachment-0001.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/017947b6/attachment-0001.jpg From msc at freeswitch.org Mon Nov 14 20:07:29 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Nov 2011 09:07:29 -0800 Subject: [Freeswitch-users] FreeSWITCH In-Reply-To: References: Message-ID: Hi Craig, I see that some others have chimed in as well, but as the official FreeSWITCH "community guy" I thought it might be good for me to say a few things. First off, welcome! We are glad you are investigating FreeSWITCH as a potential use candidate. We highly recommend that any prospective FS user do his or her due diligence. This is one area where we can all agree that OSS is awesome - you get to download it for free and take it for a test drive with no limitations. As far as adoption rates go, please understand that the FreeSWITCH devs are not at all concerned with how many people are using it. We don't have Digium's marketing department, nor do we need it. I think our marketing can be summed up by looking at the first result from this google query: http://www.google.com/#h&q=asterisk+alternative The FS devs have a mantra: use what works for you. Specifically, use the right tool for your application/scenario/problem. Many people download trixbox or piaf and color it done. We have no problem with that. If those tools work for you then great. However, it has been our experience that a steady stream of Asterisk "refugees" have come over to FS because of situations similar to yours: "frustrations with Asterisk." (If you read Appendix B of the FreeSWITCH book you'll see that Anthony Minessale started the FreeSWITCH project in large part because of "Asterisk frustrations" - so don't think that you're alone.) Perhaps the best thing to do to find out if FreeSWITCH is for you is to talk to some of our community members. We have a number of people who've made the jump from Asterisk to FreeSWITCH. I think they'll all agree that the pain is short-term but the benefits are long term. I would recommend two things: #1 - Join #freeswitch on irc.freenode.net. You'll find nearly 200 people there, many of whom can relate to your Asterisk frustrations and who are willing to talk to you. #2 - Call in to the FreeSWITCH weekly conference call on Wednesday. It's at 1PM eastern, 10AM pacific. I'm the moderator of the call. More information can be found here . We think you will find our community is quite energetic and engaging. Oh, and we have a few people who are quite opinionated as well. :) You are welcome to come join us. -Michael Collins (IRC: mercutioviz) On Sun, Nov 13, 2011 at 8:46 AM, Craig Smith wrote: > Some recent frustrations with Asterisk have me looking at FreeSWITCH, but > it?s lack? of adoption leaves me wondering. > > Here are some questions I have as a long time Asterisk user, dCAP, and > Astricon attendee. > > > Why hasn?t FreeSWITCH reached a critical mass? > > Is the XML configuration that scary? > > What does Asterisk have that FreeSWITCH doesn?t? > > Does FreeSWITCH need better documentation and/or tutorials? > > Is it git? > > Would a FreeSWITCH certification help? > > > I want to help the FreeSWITCH project, but I?m not developer or > philanthropist , what can I do? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/7ba071b5/attachment.html From vidan at centrum.cz Mon Nov 14 19:50:38 2011 From: vidan at centrum.cz (Vidan) Date: Mon, 14 Nov 2011 17:50:38 +0100 Subject: [Freeswitch-users] Howto set only one console debug level messages in fs_cli Message-ID: <20111114175038.7DBABD6C@centrum.cz> Hi, I am newbie on FreeSwitch, I would like show only the INFO(6) messages in fs_cli console, not any others (err, warning etc.). I am tried to modify switch.conf.xml, var.xml and console.conf.xml and switch.conf.xml. I see alway all type of messages until defined console loglevel value. works fine only in logfile.conf.xml. The same line does not work in my case in console.conf.xml for fs_cli. For example I will set "ERR" (3) and I will see on Fs_cli "CONSOLE" (0)+"ALERT" (1)+"CRIT" (2) messages. Is it possible to set on screen only one messages layer in fsi_cli? How is it possible to do that? I am doing this because alway I had full screen with this: 011-11-14 16:39:22.230114 [WARNING] ftmod_wanpipe.c:917 [s1c1][1:1] Rx Queue length exceeded 80% hreshold (9/10) 2011-11-14 16:39:22.230114 [WARNING] ftmod_wanpipe.c:917 [s1c1][1:1] Rx Queue length exceeded 80% hreshold (8/10) 2011-11-14 16:39:24.610115 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx idle changed from 250 to 294 2011-11-14 16:39:24.630115 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx idle changed from 38 to 294 2011-11-14 16:39:24.650130 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx idle changed from 38 to 294 2011-11-14 16:39:24.670115 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx idle changed from 38 to 294 2011-11-14 16:39:24.690126 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx idle changed from 38 to 294 2011-11-14 16:39:24.710144 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx idle changed from 38 to 294 2011-11-14 16:39:24.730125 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx idle changed from 38 to 294 and I need only notice and info messages from isdn. Thank you for answer. Best Regard, Vit From msc at freeswitch.org Mon Nov 14 20:20:02 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Nov 2011 09:20:02 -0800 Subject: [Freeswitch-users] Howto set only one console debug level messages in fs_cli In-Reply-To: <20111114175038.7DBABD6C@centrum.cz> References: <20111114175038.7DBABD6C@centrum.cz> Message-ID: Hi Vidan, I recommend you look in conf/autoload_configs/console.conf.xml. You can look at the sample mappings that are commented out. About the best you can do is to specify the log levels for the various files that you need. I know it's hard, but it's all the control that is available at the moment. -MC On Mon, Nov 14, 2011 at 8:50 AM, Vidan wrote: > > Hi, > I am newbie on FreeSwitch, > I would like show only the INFO(6) messages in fs_cli console, not any > others (err, warning etc.). I am tried to modify switch.conf.xml, var.xml > and console.conf.xml and switch.conf.xml. I see alway all type of messages > until defined console loglevel value. works > fine only in logfile.conf.xml. The same line does not work in my case in > console.conf.xml for fs_cli. > For example I will set "ERR" (3) and I will see on Fs_cli "CONSOLE" > (0)+"ALERT" (1)+"CRIT" (2) messages. Is it possible to set on screen only > one messages layer in fsi_cli? How is it possible to do that? > > I am doing this because alway I had full screen with this: > > 011-11-14 16:39:22.230114 [WARNING] ftmod_wanpipe.c:917 [s1c1][1:1] Rx > Queue length exceeded 80% hreshold (9/10) > 2011-11-14 16:39:22.230114 [WARNING] ftmod_wanpipe.c:917 [s1c1][1:1] Rx > Queue length exceeded 80% hreshold (8/10) > 2011-11-14 16:39:24.610115 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx > idle changed from 250 to 294 > 2011-11-14 16:39:24.630115 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx > idle changed from 38 to 294 > 2011-11-14 16:39:24.650130 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx > idle changed from 38 to 294 > 2011-11-14 16:39:24.670115 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx > idle changed from 38 to 294 > 2011-11-14 16:39:24.690126 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx > idle changed from 38 to 294 > 2011-11-14 16:39:24.710144 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx > idle changed from 38 to 294 > 2011-11-14 16:39:24.730125 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx > idle changed from 38 to 294 > > and I need only notice and info messages from isdn. > > Thank you for answer. > Best Regard, > Vit > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111114/f06f2ee3/attachment.html From victor.chukalovskiy at utoronto.ca Mon Nov 14 20:41:19 2011 From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy) Date: Mon, 14 Nov 2011 12:41:19 -0500 Subject: [Freeswitch-users] Issue with: answer confirmation + custom ringback + mod_freetdm Message-ID: <4EC152BF.9090201@utoronto.ca> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/28a5679b/attachment-0001.html From jmoran at secureachsystems.com Mon Nov 14 20:47:22 2011 From: jmoran at secureachsystems.com (Jason Moran) Date: Mon, 14 Nov 2011 12:47:22 -0500 Subject: [Freeswitch-users] File playback delay Message-ID: <361E98F99D3CC3439EED59BC1924ED6958A158@SERVER2003.SecuReachSystems.local> I have found that the playback of a file can be delayed to the point that the callee will either hangup (thinking it MUST be over now since it has been quiet for 7-8 seconds) or a Voicemail's silence detection will trigger even though I hadn't finished playing back all of the audio files I had intended to play. Ideas: Does session.stream block (spidermonkey script)? Since I am doing several simultaneous calls if session.stream blocked file access that could account for it (although that would be a terrible calamity if true). Does anybody else have ideas? I have been using pcapsipdump to "record" the calls and I can confirm that the reports are true - in some cases the playback of the next audio file can delay for a long time. I have tested these calls to my mobile phone and office phone without any delays present, but it seems to happen to some people some of the time. Thanks, Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/ae8b1d41/attachment.html From avi at avimarcus.net Mon Nov 14 21:59:01 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 14 Nov 2011 20:59:01 +0200 Subject: [Freeswitch-users] UUID logging - help test patch? Message-ID: If you use uuid prepending on your logging to help you grep out a call, Math wrote a patch to apply that to the multi-line SDP, too. (See http://wiki.freeswitch.org/wiki/Mod_logfile#Logging_UUIDs for how to turn on) If you can test it out and comment, the patch is here: http://jira.freeswitch.org/browse/FS-3572 Also, there's still at least 4 log types that don't get the UUID, I unfortunately still don't know enough of C & FS to check for myself why: 2011-11-03 08:41:12.696695 [INFO] switch_nat.c:589 NAT port mapping disabled 2011-11-03 08:41:12.876687 [DEBUG] switch_rtp.c:3181 Correct ip/port confirmed. 2011-11-03 08:41:12.736698 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz 2011-09-16 15:44:06.776690 [DEBUG] switch_core_file.c:180 File .....v3f.wav sample rate 44100 doesn't match requested rate 8000 -Avi Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/1fde9d89/attachment.html From brian.wiese.freeswitch at gmail.com Tue Nov 15 04:37:13 2011 From: brian.wiese.freeswitch at gmail.com (Brian Wiese) Date: Mon, 14 Nov 2011 19:37:13 -0600 Subject: [Freeswitch-users] SIP/2.0 500 No session set by user In-Reply-To: References: Message-ID: I'm still struggling to figure out what my problem is and how to resolve it. If anyone has a moment to check this out I'd really appreciate it. I've also posted another set of logs, this time one from each server, in which I'm trying to call a conference extension on 10.15.100.10 from a Polycom endpoint registered to 10.15.100.20: Polycom IP650 >>> FreeSWITCH Server "A" >>> FreeSWITCH Server "B" Log from FreeSWITCH server "A": http://pastebin.freeswitch.org/17776 Log from FreeSWITCH server "B": http://pastebin.freeswitch.org/17775 What's worse is I'm able to reproduce this problem on nearly every call. Thanks again for your help. ~Brian On Wed, Oct 26, 2011 at 8:03 PM, Brian Wiese < brian.wiese.freeswitch at gmail.com> wrote: > Hello again everyone! > > I'm getting a strange result when calling from a Polycom endpoint on FS > server "A" to FS server "B": About one out of ten calls won't complete > with the error "SIP/2.0 500 No session set by user". If I immediately > redial the number again it works. > > Here's my log: http://pastebin.freeswitch.org/17622 > Note the response to the endpoint from server "A" on line 452. > > Important IP addresses: > Polycom endpoint: 10.15.100.250 > Server "A": 10.15.100.20 > Server "B": 10.15.100.10 > > I'm on the latest revision (git-2f786a0 2011-10-26 12-57-59 -0500) as well. > > I appreciate your help. > > ~Brian > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111114/38efb62e/attachment.html From miha at softnet.si Tue Nov 15 12:09:36 2011 From: miha at softnet.si (Miha Zoubek) Date: Tue, 15 Nov 2011 10:09:36 +0100 Subject: [Freeswitch-users] freeswitch radiusclinet (mod_radius_cdr) problems In-Reply-To: References: <4EBA8B5E.2050900@softnet.si> <4EBCD686.9040305@softnet.si> <4EC0C216.3040806@softnet.si> Message-ID: <4EC22C50.4090400@softnet.si> Hi Hugh, I have put log and my dialplan in pastebin (http://pastebin.freeswitch.org/17778). Could you please take a look due to my luck of skills with freeswith and inform me about your opinion. Regards, Miha On 11/14/2011 10:30 AM, Hugh Irvine wrote: > Hello Miha - > > You are probably getting a RADIUS accounting request for each call leg. > > The id's in the requests are different, as are the lengths, but you really need to check the contents of the requests to see exactly what they contain. > > regards > > Hugh > > > On 14 Nov 2011, at 18:24, Miha Zoubek wrote: > >> Hi, >> >> can anyone can help me with this issue? >> >> >> BR, >> Miha >> >> On 11/11/2011 9:02 AM, Miha Zoubek wrote: >>> Hi @ Ognjen, >>> >>> bellow you can see that my Accounting-Request(4), Accounting-Response(5) are twice send. >>> >>> Any idea? >>> >>> BR, >>> Miha >>> >>> >>> >>> 71.449050 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=235, l=265) >>> 71.517347 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=235, l=20) >>> 73.536126 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Request(1) (id=236, l=210) >>> 73.567412 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Accept(2) (id=236, l=20) >>> 73.572794 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=237, l=321) >>> 73.574156 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=237, l=20) >>> 83.482760 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=238, l=401) >>> 83.485670 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=238, l=20) >>> 83.514594 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=239, l=402) >>> 83.516404 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=239, l=20) >>> >>> On 11/10/2011 6:24 PM, Ognjen Seslija wrote: >>>> Radius clients generally send Start and Stop records, I guess this is why you get two records. >>>> >>>> On Wed, Nov 9, 2011 at 3:17 PM, Miha Zoubek wrote: >>>> Hi, >>>> >>>> radiusclient on freeswitch in sending more that once same response. So I >>>> am getting in my sql tables for every call two inputs which is wrong. >>>> How can I deal whit this issue? >>>> >>>> I have paste a log from freeradius server in pastebin that you can see >>>> what is freeswitch sending. >>>> >>>> http://pastebin.freeswitch.org/17730 >>>> >>>> >>>> Thank you! >>>> >>>> BR, >>>> Miha >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 miha at softnet.si Tue Nov 15 13:37:54 2011 From: miha at softnet.si (Miha Zoubek) Date: Tue, 15 Nov 2011 11:37:54 +0100 Subject: [Freeswitch-users] freeswitch radiusclinet (mod_radius_cdr) problems In-Reply-To: <4EC22C50.4090400@softnet.si> References: <4EBA8B5E.2050900@softnet.si> <4EBCD686.9040305@softnet.si> <4EC0C216.3040806@softnet.si> <4EC22C50.4090400@softnet.si> Message-ID: <4EC24102.90605@softnet.si> Hi Hugh, and other question. For incoming calls... Why there is also AAA request to radius server if I do not have it in my public/dialplan? My freeradius sever is set for AAA (outgoing calls). Why NAS is sending also accounting request for incoming calls? If the radius is set for AAA, that I will also need o authenticate incoming calls which is stupid. All CRD for incoming call we have on SBC. I would like to have only for outgoing calls. Any idea? Thank you very much! BR, Miha On 11/15/2011 10:09 AM, Miha Zoubek wrote: > Hi Hugh, > > I have put log and my dialplan in pastebin > (http://pastebin.freeswitch.org/17778). > Could you please take a look due to my luck of skills with freeswith and > inform me about your opinion. > > > Regards, > Miha > > On 11/14/2011 10:30 AM, Hugh Irvine wrote: >> Hello Miha - >> >> You are probably getting a RADIUS accounting request for each call leg. >> >> The id's in the requests are different, as are the lengths, but you really need to check the contents of the requests to see exactly what they contain. >> >> regards >> >> Hugh >> >> >> On 14 Nov 2011, at 18:24, Miha Zoubek wrote: >> >>> Hi, >>> >>> can anyone can help me with this issue? >>> >>> >>> BR, >>> Miha >>> >>> On 11/11/2011 9:02 AM, Miha Zoubek wrote: >>>> Hi @ Ognjen, >>>> >>>> bellow you can see that my Accounting-Request(4), Accounting-Response(5) are twice send. >>>> >>>> Any idea? >>>> >>>> BR, >>>> Miha >>>> >>>> >>>> >>>> 71.449050 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=235, l=265) >>>> 71.517347 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=235, l=20) >>>> 73.536126 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Request(1) (id=236, l=210) >>>> 73.567412 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Accept(2) (id=236, l=20) >>>> 73.572794 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=237, l=321) >>>> 73.574156 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=237, l=20) >>>> 83.482760 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=238, l=401) >>>> 83.485670 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=238, l=20) >>>> 83.514594 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=239, l=402) >>>> 83.516404 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=239, l=20) >>>> >>>> On 11/10/2011 6:24 PM, Ognjen Seslija wrote: >>>>> Radius clients generally send Start and Stop records, I guess this is why you get two records. >>>>> >>>>> On Wed, Nov 9, 2011 at 3:17 PM, Miha Zoubek wrote: >>>>> Hi, >>>>> >>>>> radiusclient on freeswitch in sending more that once same response. So I >>>>> am getting in my sql tables for every call two inputs which is wrong. >>>>> How can I deal whit this issue? >>>>> >>>>> I have paste a log from freeradius server in pastebin that you can see >>>>> what is freeswitch sending. >>>>> >>>>> http://pastebin.freeswitch.org/17730 >>>>> >>>>> >>>>> Thank you! >>>>> >>>>> BR, >>>>> Miha >>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 leon at scarlet-internet.nl Tue Nov 15 13:50:36 2011 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 15 Nov 2011 11:50:36 +0100 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: References: <-7236397912894270049@unknownmsgid> <71430C91-82A1-4130-94FC-E404F1C6AC57@scarlet-internet.nl> Message-ID: <7444F199-DC2A-494B-899C-B84A88108282@scarlet-internet.nl> Hello Vik and Matt, Of course, here's my current opensips.cfg, but note that it contains bugs and is far from ready, just call setup/teardown and registrations work. Also, I have a dispatcher.list containing one set ("1") pointing to two sip profiles (at different udp ports) on one fs instance. Both sip profiles have the outbound-proxy param set to the sip uri of the proxy. kind regards, Leon On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote: > I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well > > On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa wrote: > Any chance you can post your opensips config? I've been struggling > with mixed success trying same setup. > > On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij > wrote: > > Hi, > > I'm actually trying to configure the same thing right now - with opensips > > though, but (afaik) it uses the same dispatcher module. > > First I used a simple route script in opensips with using dispatcher, but > > after the first message (from ua through proxy to fs), the proxy would get > > out of the signaling path, while I want it to stay in. > > To fix that, I added record routing in the proxy configuration so it stays > > in the path. > > Registrations are also balanced towards the fs servers - but some clients > > are nat'ed so the contact header was wrong - fix_nated_contact() in route > > and in onreply_route fixed that. > > To get calls originating from fs to go through the proxy before going to the > > client, I tried: > > originate > > sofia/some_profile/sip:user at mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 > > &park > > which worked. Then I found out about the sip_route_uri variable that can be > > set in the bridge string: > > originate > > {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user at mydomain.com > > &park > > which seemed to do the same thing (does anyone know if there's any > > difference between the two variants ?) > > Then I tried putting the sip_route_uri in the dial-string param of the > > domain (or user), for example: > > > value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> > > so now it's possible to just call the user as: > > originate user/some_user at mydomain.com &park > > nice ! > > And then I found out that you can just set an outbound-proxy param in the > > sip profile that's used for originating: > > > > so now you can leave out the sip_route_uri or fs_path variables, it just > > works. > > My proxy config still needs a lot of work, but I got the basic functionality > > working, still need to find out about using acls on fs side based on a sip > > header added by the proxy - I'm already adding X-AUTH-IP headers on messages > > from clients towards fs which should do the trick, but I didn't test it yet. > > Anyway, would love to read some configs of ppl who successfuly setup > > opensips/kamailio/openser/... proxies in front of fs, what their experience > > is with using record-routing or not, etc. > > regards, > > Leon > > > > > > > > On Nov 14, 2011, at 6:40 AM, Henrik Aagaard S?rensen wrote: > > > > The handling of several FS are not a issue, actually easy enough. > > I would like to have as little load on Kamailio as possible, as it just > > should load balance. > > Also, having to handle users on both Kamailio and FS makes unecessary work > > loads. > > > > On 14/11/2011, at 00.32, Sammy Govind wrote: > > > > Hi again, > > Why don't you just let Kamailio handle registrations. Anyway I was thinking > > about that LBing the registration would result in such a scenario that > > calling one extension to another would not make a successful call because > > the other endpoint maybe registered on some other FS. > > This may further lead you to making a dial-plan which would work somewhat > > like DUNDI but it'd just have to search all the FS servers before joining a > > call.(Obv there are other intelligent approaches to minimize the headache) > > Try following the link I sent you and implement that in front of your FS, I > > think that Kamailio configuration is so well written that anyone can start > > understanding kamailio and implement such setups with little effort. > > By explaining your topology I meant how do you plan to use Kamailio in front > > of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a > > topo-hiding or SBC like setup! > > -- > > Regards, > > Sammy > > > > 2011/11/14 Henrik Aagaard S?rensen > >> > >> Hi Sammy. > >> I've actually removed registration and presence from Kamailio, so all it > >> does is dispatch everything to FreeSwitch. > >> Currently I only have 1 FreeSwitch, for testing this basic setup. > >> Next move would be 2 FreeSwitch with 1 common database etc. But that's > >> later. > >> My FreeSwitch is the basic setup, without anything else. Just as the > >> installation manuel is written. So I have my extension 1000 - 1019 etc. And > >> calls between them works when connected directly to FreeSwitch. > >> But when going through Kamailio Dispatcher it fails between the > >> extensions. > >> So I guess there should be some more setup in FreeSwitch when using a load > >> balancer (dispatcher) in front of it. > >> My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle > >> everything. > >> > >> On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind wrote: > >>> > >>> Hi, > >>> If everything is setup as you expected it then don't read this > >>> > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, > >>> else follow the article from start till end and implement it. Don't worry > >>> about the name asterisk, just replace that all with FreeSWICTH. It will work > >>> like that too. > >>> Once that is implemented then apply dispatcher module on the route which > >>> says [REGFWD] or [TOASTERISK]. > >>> Also explain your topology abit more and features required as well that > >>> may help in telling which extra module you will require in order to make > >>> things work. > >>> In your current implementation are you sure SIP phones are registering on > >>> FS and not on Kamailio?and that both end points making call to each other > >>> are on same FS? > >>> -- > >>> Regards, > >>> Sammy > >>> > >>> 2011/11/14 Henrik Aagaard S?rensen > >>>> > >>>> Hi everyone. In regards to my earlier question regarding with FreeSwitch > >>>> behind Kamailio Dispatcher, I've attached a call from extension 1001 to > >>>> 1002, which fails. It just hangs for some time and then says that 1002 > >>>> cannot be found, and then the voicemail for it comes up. > >>>> 2011/11/13 Henrik Aagaard S?rensen > >>>>> > >>>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. > >>>>> I've setup Kamailio via > >>>>> this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html > >>>>> I've installed FreeSwitch from scratch on Ubuntu > >>>>> via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start > >>>>> Now, when registering extensions via Kamailio Dispatcher I'm able to > >>>>> call to FreeSwitch and listen to hold music. But that's it. I'm not able to > >>>>> call between extensions etc. > >>>>> Can anyone help me setting up FreeSwitch to accept registration, calls > >>>>> etc. from Kamailio and everything else that is needed to use FreeSwitch > >>>>> behind a load balancer? > >>>>> I'm very new to FreeSwitch, but I'm trying to use the terminal (without > >>>>> any GUI etc.) as I want the installation to be as clean as possible. So I > >>>>> would prefer very precise help, as I'm still getting hold of FreeSwitch. > >>>> > >>>> > >>>> _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.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 > >>>> > >>> > >>> > >>> _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.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 > >>> > >> > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.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 > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/bbb63a86/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: opensips.cfg Type: application/octet-stream Size: 6819 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/bbb63a86/attachment-0001.obj -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/bbb63a86/attachment-0003.html From govoiper at gmail.com Tue Nov 15 14:33:16 2011 From: govoiper at gmail.com (Sammy Govind) Date: Tue, 15 Nov 2011 16:33:16 +0500 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: <7444F199-DC2A-494B-899C-B84A88108282@scarlet-internet.nl> References: <-7236397912894270049@unknownmsgid> <71430C91-82A1-4130-94FC-E404F1C6AC57@scarlet-internet.nl> <7444F199-DC2A-494B-899C-B84A88108282@scarlet-internet.nl> Message-ID: Looking at the original topic and the opensips.cfg really confused me that we started from kamailio and ended up in openSIPS ! On Tue, Nov 15, 2011 at 3:50 PM, Leon de Rooij wrote: > Hello Vik and Matt, > > Of course, here's my current opensips.cfg, but note that it contains bugs > and is far from ready, just call setup/teardown and registrations work. > > Also, I have a dispatcher.list containing one set ("1") pointing to two > sip profiles (at different udp ports) on one fs instance. Both sip profiles > have the outbound-proxy param set to the sip uri of the proxy. > > kind regards, > > Leon > > > > > > On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote: > > I would love to see the config too. I'm actively trying to use OpenSips as > an inbound / outbound proxy SBC and am having mixed success as well > > On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa wrote: > >> Any chance you can post your opensips config? I've been struggling >> with mixed success trying same setup. >> >> On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij >> wrote: >> > Hi, >> > I'm actually trying to configure the same thing right now - with >> opensips >> > though, but (afaik) it uses the same dispatcher module. >> > First I used a simple route script in opensips with using dispatcher, >> but >> > after the first message (from ua through proxy to fs), the proxy would >> get >> > out of the signaling path, while I want it to stay in. >> > To fix that, I added record routing in the proxy configuration so it >> stays >> > in the path. >> > Registrations are also balanced towards the fs servers - but some >> clients >> > are nat'ed so the contact header was wrong - fix_nated_contact() in >> route >> > and in onreply_route fixed that. >> > To get calls originating from fs to go through the proxy before going >> to the >> > client, I tried: >> > originate >> > sofia/some_profile/sip:user at mydomain.com >> ;fs_path=sip:ip_of_sip_proxy:5060 >> > &park >> > which worked. Then I found out about the sip_route_uri variable that >> can be >> > set in the bridge string: >> > originate >> > {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/ >> sip:some_user at mydomain.com >> > &park >> > which seemed to do the same thing (does anyone know if there's any >> > difference between the two variants ?) >> > Then I tried putting the sip_route_uri in the dial-string param of the >> > domain (or user), for example: >> > > > >> value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@ >> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> >> > so now it's possible to just call the user as: >> > originate user/some_user at mydomain.com &park >> > nice ! >> > And then I found out that you can just set an outbound-proxy param in >> the >> > sip profile that's used for originating: >> > >> > so now you can leave out the sip_route_uri or fs_path variables, it just >> > works. >> > My proxy config still needs a lot of work, but I got the basic >> functionality >> > working, still need to find out about using acls on fs side based on a >> sip >> > header added by the proxy - I'm already adding X-AUTH-IP headers on >> messages >> > from clients towards fs which should do the trick, but I didn't test it >> yet. >> > Anyway, would love to read some configs of ppl who successfuly setup >> > opensips/kamailio/openser/... proxies in front of fs, what their >> experience >> > is with using record-routing or not, etc. >> > regards, >> > Leon >> > >> > >> > >> > On Nov 14, 2011, at 6:40 AM, Henrik Aagaard S?rensen wrote: >> > >> > The handling of several FS are not a issue, actually easy enough. >> > I would like to have as little load on Kamailio as possible, as it just >> > should load balance. >> > Also, having to handle users on both Kamailio and FS makes unecessary >> work >> > loads. >> > >> > On 14/11/2011, at 00.32, Sammy Govind wrote: >> > >> > Hi again, >> > Why don't you just let Kamailio handle registrations. Anyway I was >> thinking >> > about that LBing the registration would result in such a scenario that >> > calling one extension to another would not make a successful call >> because >> > the other endpoint maybe registered on some other FS. >> > This may further lead you to making a dial-plan which would work >> somewhat >> > like DUNDI but it'd just have to search all the FS servers before >> joining a >> > call.(Obv there are other intelligent approaches to minimize the >> headache) >> > Try following the link I sent you and implement that in front of your >> FS, I >> > think that Kamailio configuration is so well written that anyone can >> start >> > understanding kamailio and implement such setups with little effort. >> > By explaining your topology I meant how do you plan to use Kamailio in >> front >> > of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as >> in a >> > topo-hiding or SBC like setup! >> > -- >> > Regards, >> > Sammy >> > >> > 2011/11/14 Henrik Aagaard S?rensen >> >> >> >> Hi Sammy. >> >> I've actually removed registration and presence from Kamailio, so all >> it >> >> does is dispatch everything to FreeSwitch. >> >> Currently I only have 1 FreeSwitch, for testing this basic setup. >> >> Next move would be 2 FreeSwitch with 1 common database etc. But that's >> >> later. >> >> My FreeSwitch is the basic setup, without anything else. Just as the >> >> installation manuel is written. So I have my extension 1000 - 1019 >> etc. And >> >> calls between them works when connected directly to FreeSwitch. >> >> But when going through Kamailio Dispatcher it fails between the >> >> extensions. >> >> So I guess there should be some more setup in FreeSwitch when using a >> load >> >> balancer (dispatcher) in front of it. >> >> My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle >> >> everything. >> >> >> >> On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind >> wrote: >> >>> >> >>> Hi, >> >>> If everything is setup as you expected it then don't read this >> >>> > >> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb >> , >> >>> else follow the article from start till end and implement it. Don't >> worry >> >>> about the name asterisk, just replace that all with FreeSWICTH. It >> will work >> >>> like that too. >> >>> Once that is implemented then apply dispatcher module on the route >> which >> >>> says [REGFWD] or [TOASTERISK]. >> >>> Also explain your topology abit more and features required as well >> that >> >>> may help in telling which extra module you will require in order to >> make >> >>> things work. >> >>> In your current implementation are you sure SIP phones are >> registering on >> >>> FS and not on Kamailio?and that both end points making call to each >> other >> >>> are on same FS? >> >>> -- >> >>> Regards, >> >>> Sammy >> >>> >> >>> 2011/11/14 Henrik Aagaard S?rensen >> >>>> >> >>>> Hi everyone. In regards to my earlier question regarding with >> FreeSwitch >> >>>> behind Kamailio Dispatcher, I've attached a call from extension 1001 >> to >> >>>> 1002, which fails. It just hangs for some time and then says that >> 1002 >> >>>> cannot be found, and then the voicemail for it comes up. >> >>>> 2011/11/13 Henrik Aagaard S?rensen >> >>>>> >> >>>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to >> work. >> >>>>> I've setup Kamailio via >> >>>>> this: >> http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >> >>>>> I've installed FreeSwitch from scratch on Ubuntu >> >>>>> via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >> >>>>> Now, when registering extensions via Kamailio Dispatcher I'm able to >> >>>>> call to FreeSwitch and listen to hold music. But that's it. I'm not >> able to >> >>>>> call between extensions etc. >> >>>>> Can anyone help me setting up FreeSwitch to accept registration, >> calls >> >>>>> etc. from Kamailio and everything else that is needed to use >> FreeSwitch >> >>>>> behind a load balancer? >> >>>>> I'm very new to FreeSwitch, but I'm trying to use the terminal >> (without >> >>>>> any GUI etc.) as I want the installation to be as clean as >> possible. So I >> >>>>> would prefer very precise help, as I'm still getting hold of >> FreeSwitch. >> >>>> >> >>>> >> >>>> >> _________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://wiki.freeswitch.org >> >>>> http://www.cluecon.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 >> >>>> >> >>> >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.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 >> >>> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.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 >> >> >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.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 >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.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 >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.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 >> > >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/5667bffd/attachment-0001.html From leon at scarlet-internet.nl Tue Nov 15 17:05:12 2011 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 15 Nov 2011 15:05:12 +0100 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: References: <-7236397912894270049@unknownmsgid> <71430C91-82A1-4130-94FC-E404F1C6AC57@scarlet-internet.nl> <7444F199-DC2A-494B-899C-B84A88108282@scarlet-internet.nl> Message-ID: Hello Sammy, Well, Kamailio and OpenSIPS are essentially the same, eh. They are both forks from the same codebase. Kind regards, Leon On Nov 15, 2011, at 12:33 PM, Sammy Govind wrote: > Looking at the original topic and the opensips.cfg really confused me that we started from kamailio and ended up in openSIPS ! > > On Tue, Nov 15, 2011 at 3:50 PM, Leon de Rooij wrote: > Hello Vik and Matt, > > Of course, here's my current opensips.cfg, but note that it contains bugs and is far from ready, just call setup/teardown and registrations work. > > Also, I have a dispatcher.list containing one set ("1") pointing to two sip profiles (at different udp ports) on one fs instance. Both sip profiles have the outbound-proxy param set to the sip uri of the proxy. > > kind regards, > > Leon > > > > > > On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote: > >> I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well >> >> On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa wrote: >> Any chance you can post your opensips config? I've been struggling >> with mixed success trying same setup. >> >> On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij >> wrote: >> > Hi, >> > I'm actually trying to configure the same thing right now - with opensips >> > though, but (afaik) it uses the same dispatcher module. >> > First I used a simple route script in opensips with using dispatcher, but >> > after the first message (from ua through proxy to fs), the proxy would get >> > out of the signaling path, while I want it to stay in. >> > To fix that, I added record routing in the proxy configuration so it stays >> > in the path. >> > Registrations are also balanced towards the fs servers - but some clients >> > are nat'ed so the contact header was wrong - fix_nated_contact() in route >> > and in onreply_route fixed that. >> > To get calls originating from fs to go through the proxy before going to the >> > client, I tried: >> > originate >> > sofia/some_profile/sip:user at mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 >> > &park >> > which worked. Then I found out about the sip_route_uri variable that can be >> > set in the bridge string: >> > originate >> > {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user at mydomain.com >> > &park >> > which seemed to do the same thing (does anyone know if there's any >> > difference between the two variants ?) >> > Then I tried putting the sip_route_uri in the dial-string param of the >> > domain (or user), for example: >> > > > value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> >> > so now it's possible to just call the user as: >> > originate user/some_user at mydomain.com &park >> > nice ! >> > And then I found out that you can just set an outbound-proxy param in the >> > sip profile that's used for originating: >> > >> > so now you can leave out the sip_route_uri or fs_path variables, it just >> > works. >> > My proxy config still needs a lot of work, but I got the basic functionality >> > working, still need to find out about using acls on fs side based on a sip >> > header added by the proxy - I'm already adding X-AUTH-IP headers on messages >> > from clients towards fs which should do the trick, but I didn't test it yet. >> > Anyway, would love to read some configs of ppl who successfuly setup >> > opensips/kamailio/openser/... proxies in front of fs, what their experience >> > is with using record-routing or not, etc. >> > regards, >> > Leon >> > >> > >> > >> > On Nov 14, 2011, at 6:40 AM, Henrik Aagaard S?rensen wrote: >> > >> > The handling of several FS are not a issue, actually easy enough. >> > I would like to have as little load on Kamailio as possible, as it just >> > should load balance. >> > Also, having to handle users on both Kamailio and FS makes unecessary work >> > loads. >> > >> > On 14/11/2011, at 00.32, Sammy Govind wrote: >> > >> > Hi again, >> > Why don't you just let Kamailio handle registrations. Anyway I was thinking >> > about that LBing the registration would result in such a scenario that >> > calling one extension to another would not make a successful call because >> > the other endpoint maybe registered on some other FS. >> > This may further lead you to making a dial-plan which would work somewhat >> > like DUNDI but it'd just have to search all the FS servers before joining a >> > call.(Obv there are other intelligent approaches to minimize the headache) >> > Try following the link I sent you and implement that in front of your FS, I >> > think that Kamailio configuration is so well written that anyone can start >> > understanding kamailio and implement such setups with little effort. >> > By explaining your topology I meant how do you plan to use Kamailio in front >> > of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a >> > topo-hiding or SBC like setup! >> > -- >> > Regards, >> > Sammy >> > >> > 2011/11/14 Henrik Aagaard S?rensen >> >> >> >> Hi Sammy. >> >> I've actually removed registration and presence from Kamailio, so all it >> >> does is dispatch everything to FreeSwitch. >> >> Currently I only have 1 FreeSwitch, for testing this basic setup. >> >> Next move would be 2 FreeSwitch with 1 common database etc. But that's >> >> later. >> >> My FreeSwitch is the basic setup, without anything else. Just as the >> >> installation manuel is written. So I have my extension 1000 - 1019 etc. And >> >> calls between them works when connected directly to FreeSwitch. >> >> But when going through Kamailio Dispatcher it fails between the >> >> extensions. >> >> So I guess there should be some more setup in FreeSwitch when using a load >> >> balancer (dispatcher) in front of it. >> >> My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle >> >> everything. >> >> >> >> On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind wrote: >> >>> >> >>> Hi, >> >>> If everything is setup as you expected it then don't read this >> >>> > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, >> >>> else follow the article from start till end and implement it. Don't worry >> >>> about the name asterisk, just replace that all with FreeSWICTH. It will work >> >>> like that too. >> >>> Once that is implemented then apply dispatcher module on the route which >> >>> says [REGFWD] or [TOASTERISK]. >> >>> Also explain your topology abit more and features required as well that >> >>> may help in telling which extra module you will require in order to make >> >>> things work. >> >>> In your current implementation are you sure SIP phones are registering on >> >>> FS and not on Kamailio?and that both end points making call to each other >> >>> are on same FS? >> >>> -- >> >>> Regards, >> >>> Sammy >> >>> >> >>> 2011/11/14 Henrik Aagaard S?rensen >> >>>> >> >>>> Hi everyone. In regards to my earlier question regarding with FreeSwitch >> >>>> behind Kamailio Dispatcher, I've attached a call from extension 1001 to >> >>>> 1002, which fails. It just hangs for some time and then says that 1002 >> >>>> cannot be found, and then the voicemail for it comes up. >> >>>> 2011/11/13 Henrik Aagaard S?rensen >> >>>>> >> >>>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. >> >>>>> I've setup Kamailio via >> >>>>> this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >> >>>>> I've installed FreeSwitch from scratch on Ubuntu >> >>>>> via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >> >>>>> Now, when registering extensions via Kamailio Dispatcher I'm able to >> >>>>> call to FreeSwitch and listen to hold music. But that's it. I'm not able to >> >>>>> call between extensions etc. >> >>>>> Can anyone help me setting up FreeSwitch to accept registration, calls >> >>>>> etc. from Kamailio and everything else that is needed to use FreeSwitch >> >>>>> behind a load balancer? >> >>>>> I'm very new to FreeSwitch, but I'm trying to use the terminal (without >> >>>>> any GUI etc.) as I want the installation to be as clean as possible. So I >> >>>>> would prefer very precise help, as I'm still getting hold of FreeSwitch. >> >>>> >> >>>> >> >>>> _________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://wiki.freeswitch.org >> >>>> http://www.cluecon.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 >> >>>> >> >>> >> >>> >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.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 >> >>> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.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 >> >> >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.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 >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.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 >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.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 >> > >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/581972b5/attachment-0001.html From bsevier at inetwork.com Tue Nov 15 17:21:19 2011 From: bsevier at inetwork.com (Blake Sevier) Date: Tue, 15 Nov 2011 07:21:19 -0700 Subject: [Freeswitch-users] Not seeing sip_call_id in b-leg XML CDR for cancelled calls Message-ID: <7f201b60894d031ebed132ca359a8f6b@mail.gmail.com> Hi all, Quick question - For my bridged calls, if my b-leg call is cancelled during call setup (ex. hangup_cause=ORIGINATOR_CANCEL), I?m not seeing a ?sip_call_id? for the b-leg of the call within the XML CDRs. I probably have something setup incorrectly, but does anyone else see this behavior? Thanks, Blake -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/0dd2519a/attachment.html From turqmr2 at gmail.com Tue Nov 15 17:38:38 2011 From: turqmr2 at gmail.com (Jacob Smith) Date: Tue, 15 Nov 2011 09:38:38 -0500 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: <7f201b60894d031ebed132ca359a8f6b@mail.gmail.com> References: <7f201b60894d031ebed132ca359a8f6b@mail.gmail.com> Message-ID: <4EC2796E.40405@gmail.com> I am trying to get FS to work in the simplest scenario possible: one phone, one path: google voice. On a brand new install, I followed the instructions on http://wiki.freeswitch.org/wiki/Google_Voice then I uncommented the line for in the modules.conf.xml I am getting closer, but need another nudge in the right direction. Can anyone tell me what I am missing? Here is what happens when I try to place a call: [effective_caller_id_name]=[Smith Family] EXECUTE sofia/internal/1000 at 192.168.0.5 bridge(sofia/gateway//15551234567) 2011-11-14 21:48:18.286750 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-11-14 21:48:18.286750 [ERR] mod_sofia.c:4238 Invalid Gateway 2011-11-14 21:48:18.286750 [NOTICE] mod_sofia.c:4611 Close Channel N/A [CS_NEW] 2011-11-14 21:48:18.286750 [DEBUG] switch_core_state_machine.c:494 () Running State Change CS_DESTROY 2011-11-14 21:48:18.286750 [DEBUG] switch_core_state_machine.c:504 (N/A) State DESTROY 2011-11-14 21:48:18.286750 [DEBUG] mod_sofia.c:370 N/A SOFIA DESTROY 2011-11-14 21:48:18.286750 [DEBUG] switch_core_state_machine.c:504 (N/A) State DESTROY going to sleep 2011-11-14 21:48:18.286750 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2011-11-14 21:48:18.286750 [DEBUG] switch_ivr_originate.c:3352 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2011-11-14 21:48:18.286750 [INFO] mod_dptools.c:2811 Originate Failed. Cause: INVALID_NUMBER_FORMAT 2011-11-14 21:48:18.286750 [DEBUG] switch_channel.c:2826 (sofia/internal/1000 at 192.168.0.5) Callstate Change RINGING -> HANGUP From valery.kalinin at gmail.com Tue Nov 15 17:52:42 2011 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Tue, 15 Nov 2011 20:52:42 +0600 Subject: [Freeswitch-users] My new web utility - FreeSWITCH panel Message-ID: Hi all! I present my new online web utility for FreeSWITCH - admin panel. This application allows you to view online: - registered subscribers (Sofia) - current calls - conferences - FreeTDM channels It is also possible some Freeswitch management. Screenshot with simple help: https://sites.google.com/site/freeswitched/home/downloads/fspanel_help.png Installation guide: Upload files to your web-server with PHP support. Check modules.conf and event_socket.conf for enable module mod_event_socket Change variables if needed in fspanel.php file: $FreeSWITCHserver = '127 .0.0.1 '; $FreeSWITCHport = 8021; $FreeSWITCHpassword = 'ClueCon'; $SofiaProfiles = array('internal', 'external'); // write empty array if not used: array(); $FreeTDMspans = array('pri'); // write empty array if not used: array(); Enter in browser http://you-server-name/somedir/fspanel.html Click on menu item "settings" for choice items Tested on IE 9, Firefox 4 & 8 Home: https://sites.google.com/site/freeswitched/home/freeswitch-panel Download: https://sites.google.com/site/freeswitched/home/downloads/fspanel.zip?attredirects=0&d=1 Also available at github. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/36b74672/attachment.html From govoiper at gmail.com Tue Nov 15 18:06:03 2011 From: govoiper at gmail.com (Sammy Govind) Date: Tue, 15 Nov 2011 20:06:03 +0500 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: References: <-7236397912894270049@unknownmsgid> <71430C91-82A1-4130-94FC-E404F1C6AC57@scarlet-internet.nl> <7444F199-DC2A-494B-899C-B84A88108282@scarlet-internet.nl> Message-ID: :) :) :) LOL-- right- Sorry, I'm from Narnia didn't know. Original Poster had issue with Kamailio and FS setup, even if both the projects share common parents but I don't believe that sharing opensips.cfg will help the issue here. Anyways I appreciate the spirit to help others. -- BR, Sammy On Tue, Nov 15, 2011 at 7:05 PM, Leon de Rooij wrote: > Hello Sammy, > > Well, Kamailio and OpenSIPS are essentially the same, eh. They are both > forks from the same codebase. > > Kind regards, > > Leon > > > On Nov 15, 2011, at 12:33 PM, Sammy Govind wrote: > > Looking at the original topic and the opensips.cfg really confused me that > we started from kamailio and ended up in openSIPS ! > > On Tue, Nov 15, 2011 at 3:50 PM, Leon de Rooij wrote: > >> Hello Vik and Matt, >> >> Of course, here's my current opensips.cfg, but note that it contains bugs >> and is far from ready, just call setup/teardown and registrations work. >> >> Also, I have a dispatcher.list containing one set ("1") pointing to two >> sip profiles (at different udp ports) on one fs instance. Both sip profiles >> have the outbound-proxy param set to the sip uri of the proxy. >> >> kind regards, >> >> Leon >> >> >> >> >> >> On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote: >> >> I would love to see the config too. I'm actively trying to use OpenSips >> as an inbound / outbound proxy SBC and am having mixed success as well >> >> On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa wrote: >> >>> Any chance you can post your opensips config? I've been struggling >>> with mixed success trying same setup. >>> >>> On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij >>> wrote: >>> > Hi, >>> > I'm actually trying to configure the same thing right now - with >>> opensips >>> > though, but (afaik) it uses the same dispatcher module. >>> > First I used a simple route script in opensips with using dispatcher, >>> but >>> > after the first message (from ua through proxy to fs), the proxy would >>> get >>> > out of the signaling path, while I want it to stay in. >>> > To fix that, I added record routing in the proxy configuration so it >>> stays >>> > in the path. >>> > Registrations are also balanced towards the fs servers - but some >>> clients >>> > are nat'ed so the contact header was wrong - fix_nated_contact() in >>> route >>> > and in onreply_route fixed that. >>> > To get calls originating from fs to go through the proxy before going >>> to the >>> > client, I tried: >>> > originate >>> > sofia/some_profile/sip:user at mydomain.com >>> ;fs_path=sip:ip_of_sip_proxy:5060 >>> > &park >>> > which worked. Then I found out about the sip_route_uri variable that >>> can be >>> > set in the bridge string: >>> > originate >>> > {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/ >>> sip:some_user at mydomain.com >>> > &park >>> > which seemed to do the same thing (does anyone know if there's any >>> > difference between the two variants ?) >>> > Then I tried putting the sip_route_uri in the dial-string param of the >>> > domain (or user), for example: >>> > >> > >>> value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@ >>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> >>> > so now it's possible to just call the user as: >>> > originate user/some_user at mydomain.com &park >>> > nice ! >>> > And then I found out that you can just set an outbound-proxy param in >>> the >>> > sip profile that's used for originating: >>> > >>> > so now you can leave out the sip_route_uri or fs_path variables, it >>> just >>> > works. >>> > My proxy config still needs a lot of work, but I got the basic >>> functionality >>> > working, still need to find out about using acls on fs side based on a >>> sip >>> > header added by the proxy - I'm already adding X-AUTH-IP headers on >>> messages >>> > from clients towards fs which should do the trick, but I didn't test >>> it yet. >>> > Anyway, would love to read some configs of ppl who successfuly setup >>> > opensips/kamailio/openser/... proxies in front of fs, what their >>> experience >>> > is with using record-routing or not, etc. >>> > regards, >>> > Leon >>> > >>> > >>> > >>> > On Nov 14, 2011, at 6:40 AM, Henrik Aagaard S?rensen wrote: >>> > >>> > The handling of several FS are not a issue, actually easy enough. >>> > I would like to have as little load on Kamailio as possible, as it just >>> > should load balance. >>> > Also, having to handle users on both Kamailio and FS makes unecessary >>> work >>> > loads. >>> > >>> > On 14/11/2011, at 00.32, Sammy Govind wrote: >>> > >>> > Hi again, >>> > Why don't you just let Kamailio handle registrations. Anyway I was >>> thinking >>> > about that LBing the registration would result in such a scenario that >>> > calling one extension to another would not make a successful call >>> because >>> > the other endpoint maybe registered on some other FS. >>> > This may further lead you to making a dial-plan which would work >>> somewhat >>> > like DUNDI but it'd just have to search all the FS servers before >>> joining a >>> > call.(Obv there are other intelligent approaches to minimize the >>> headache) >>> > Try following the link I sent you and implement that in front of your >>> FS, I >>> > think that Kamailio configuration is so well written that anyone can >>> start >>> > understanding kamailio and implement such setups with little effort. >>> > By explaining your topology I meant how do you plan to use Kamailio in >>> front >>> > of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as >>> in a >>> > topo-hiding or SBC like setup! >>> > -- >>> > Regards, >>> > Sammy >>> > >>> > 2011/11/14 Henrik Aagaard S?rensen >>> >> >>> >> Hi Sammy. >>> >> I've actually removed registration and presence from Kamailio, so all >>> it >>> >> does is dispatch everything to FreeSwitch. >>> >> Currently I only have 1 FreeSwitch, for testing this basic setup. >>> >> Next move would be 2 FreeSwitch with 1 common database etc. But that's >>> >> later. >>> >> My FreeSwitch is the basic setup, without anything else. Just as the >>> >> installation manuel is written. So I have my extension 1000 - 1019 >>> etc. And >>> >> calls between them works when connected directly to FreeSwitch. >>> >> But when going through Kamailio Dispatcher it fails between the >>> >> extensions. >>> >> So I guess there should be some more setup in FreeSwitch when using a >>> load >>> >> balancer (dispatcher) in front of it. >>> >> My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle >>> >> everything. >>> >> >>> >> On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind >>> wrote: >>> >>> >>> >>> Hi, >>> >>> If everything is setup as you expected it then don't read this >>> >>> > >>> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb >>> , >>> >>> else follow the article from start till end and implement it. Don't >>> worry >>> >>> about the name asterisk, just replace that all with FreeSWICTH. It >>> will work >>> >>> like that too. >>> >>> Once that is implemented then apply dispatcher module on the route >>> which >>> >>> says [REGFWD] or [TOASTERISK]. >>> >>> Also explain your topology abit more and features required as well >>> that >>> >>> may help in telling which extra module you will require in order to >>> make >>> >>> things work. >>> >>> In your current implementation are you sure SIP phones are >>> registering on >>> >>> FS and not on Kamailio?and that both end points making call to each >>> other >>> >>> are on same FS? >>> >>> -- >>> >>> Regards, >>> >>> Sammy >>> >>> >>> >>> 2011/11/14 Henrik Aagaard S?rensen >>> >>>> >>> >>>> Hi everyone. In regards to my earlier question regarding with >>> FreeSwitch >>> >>>> behind Kamailio Dispatcher, I've attached a call from extension >>> 1001 to >>> >>>> 1002, which fails. It just hangs for some time and then says that >>> 1002 >>> >>>> cannot be found, and then the voicemail for it comes up. >>> >>>> 2011/11/13 Henrik Aagaard S?rensen >> > >>> >>>>> >>> >>>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to >>> work. >>> >>>>> I've setup Kamailio via >>> >>>>> this: >>> http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >>> >>>>> I've installed FreeSwitch from scratch on Ubuntu >>> >>>>> via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >>> >>>>> Now, when registering extensions via Kamailio Dispatcher I'm able >>> to >>> >>>>> call to FreeSwitch and listen to hold music. But that's it. I'm >>> not able to >>> >>>>> call between extensions etc. >>> >>>>> Can anyone help me setting up FreeSwitch to accept registration, >>> calls >>> >>>>> etc. from Kamailio and everything else that is needed to use >>> FreeSwitch >>> >>>>> behind a load balancer? >>> >>>>> I'm very new to FreeSwitch, but I'm trying to use the terminal >>> (without >>> >>>>> any GUI etc.) as I want the installation to be as clean as >>> possible. So I >>> >>>>> would prefer very precise help, as I'm still getting hold of >>> FreeSwitch. >>> >>>> >>> >>>> >>> >>>> >>> _________________________________________________________________________ >>> >>>> Professional FreeSWITCH Consulting Services: >>> >>>> consulting at freeswitch.org >>> >>>> http://www.freeswitchsolutions.com >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> Official FreeSWITCH Sites >>> >>>> http://www.freeswitch.org >>> >>>> http://wiki.freeswitch.org >>> >>>> http://www.cluecon.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 >>> >>>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.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 >>> >>> >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.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 >>> >> >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.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 >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.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 >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.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 >>> > >>> > >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/daa45543/attachment-0001.html From adam.kelloway at newpace.ca Tue Nov 15 18:26:35 2011 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Tue, 15 Nov 2011 11:26:35 -0400 Subject: [Freeswitch-users] sendevent issues Message-ID: <4EC284AB.7070509@newpace.ca> Hi there, I have client connection to a freeswitch instance, and send the following message to freeswitch from it: sendevent CUSTOM\n Event-Name: CUSTOM\n Event-Subclass: my::event\n my-header: my-value\n \n I receive the following reply: Content-Type: command/reply Reply-Text: +OK I subscribe to all events from fs_cli, using: /events all However, I never see the event that was sent. Am I misinterpreting how sendevent is used? I would expect that the console application would see this event. Am I perhaps missing some mandatory headers? Thanks, Adam From valery.kalinin at gmail.com Tue Nov 15 08:44:34 2011 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Tue, 15 Nov 2011 11:44:34 +0600 Subject: [Freeswitch-users] My new web utility - FreeSWITCH panel Message-ID: Hi all! I present my new online web utility for FreeSWITCH - admin panel. This application allows you to view online: - registered subscribers (Sofia) - current calls - conferences - FreeTDM channels It is also possible Freeswitch management. Screenshot with simple help: https://sites.google.com/site/freeswitched/home/downloads/fspanel_help.png Installation guide: Upload files to your web-server with PHP support. Check modules.conf and event_socket.conf for enable module mod_event_socket Change variables if needed in fspanel.php file: $FreeSWITCHserver = '127 .0.0.1 '; $FreeSWITCHport = 8021; $FreeSWITCHpassword = 'ClueCon'; $SofiaProfiles = array('internal', 'external'); // write empty array if not used: array(); $FreeTDMspans = array('pri'); // write empty array if not used: array(); Enter in browser http://you-server-name/somedir/fspanel.html Click on menu item "settings" for choice items Tested on IE 9, Firefox 4 & 8 Home: https://sites.google.com/site/freeswitched/home/freeswitch-panel Download: https://sites.google.com/site/freeswitched/home/downloads/fspanel.zip?attredirects=0&d=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/48fd8fb2/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: fspanel_help.png Type: image/png Size: 314458 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/48fd8fb2/attachment-0001.png From msc at freeswitch.org Tue Nov 15 19:21:28 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 08:21:28 -0800 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: <4EC2796E.40405@gmail.com> References: <7f201b60894d031ebed132ca359a8f6b@mail.gmail.com> <4EC2796E.40405@gmail.com> Message-ID: I would guess that you have an issue in your dialplan somewhere. Notice that there is no gateway name specified in this bridge call: EXECUTE sofia/internal/1000 at 192.168.0.5 bridge(sofia/gateway//15551234567) > If you need help looking at your dialplan then throw it on pastebin.freeswitch.org and we'll take a peek at it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/6ed4d671/attachment.html From leon at scarlet-internet.nl Tue Nov 15 19:30:00 2011 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 15 Nov 2011 17:30:00 +0100 Subject: [Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher. In-Reply-To: References: <-7236397912894270049@unknownmsgid> <71430C91-82A1-4130-94FC-E404F1C6AC57@scarlet-internet.nl> <7444F199-DC2A-494B-899C-B84A88108282@scarlet-internet.nl> Message-ID: Just gave the config I'm working on because two ppl asked for it. But from what I understand from the original poster, is that he's having trouble with calling from one ua to another where both ua's are registered on fs and all signaling originating from the ua's towards fs is balanced by a proxy - which is the same as what I'm trying to achieve and have basically working now. I agree that just pasting a config is not the solution for the OP as there are so many combinations / possibilities to implement this. So indeed as you asked: * what is the topology * do you want the proxy to _always_ be in the signaling path between fs and ua The trace that was attached shows: U 2011/11/13 17:08:28.788249 216.231.104.54:5060 -> 76.33.21.131:5060 INVITE sip:1002 at sip.my-domain.com SIP/2.0. Record-Route: . Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKcfcd.3c288a3.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjXqQE4ca7HrI0tOo8umTYyui8DbpSPjDQ. From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: . U 2011/11/13 17:08:28.788575 76.33.21.131:5060 -> 216.231.104.54:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKcfcd.3c288a3.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjXqQE4ca7HrI0tOo8umTYyui8DbpSPjDQ. Record-Route: . From: ;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: . U 2011/11/13 17:08:28.803526 76.33.21.131:5060 -> 96.224.14.164:43975 INVITE sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKy6m7184yKFDXr. From: "Extension 1001" ;tag=FgF5QDyZjQXtr. To: . Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. U 2011/11/13 17:08:29.805308 76.33.21.131:5060 -> 96.224.14.164:43975 INVITE sip:1002 at 96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKy6m7184yKFDXr. From: "Extension 1001" ;tag=FgF5QDyZjQXtr. To: . Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. etc I can assume that: * 76.33.21.131 is fs * SIP packets (for 1002) are sent from FS directly to 96.224.14.164:43975 Since the invite towards the ua is such a high port, can I assume that the ua is nat'ed ? The INVITE is sent multiple times without an answer, so I think that the UA (1002) behind 96.224.14.164 has registered through the proxy on the fs server, so a nat entry (triple internal ip/port - local ip/port - external ip/port) in the state table of the firewall in front of the ua exists towards the proxy where it registered to, so the fs server can never directly reach the UA, so the INVITE from FS towards the UA must go through the proxy as well which isn't happening right now - that can be fixed by setting the outbound-proxy param on fs in the sip_profile. Anyway, am looking forward to updates on this as I really want to get this working as well ! Kind regards, Leon On Nov 15, 2011, at 4:06 PM, Sammy Govind wrote: > :) :) :) > LOL-- right- Sorry, I'm from Narnia didn't know. > > Original Poster had issue with Kamailio and FS setup, even if both the projects share common parents but I don't believe that sharing opensips.cfg will help the issue here. Anyways I appreciate the spirit to help others. > > -- > BR, > Sammy > > > On Tue, Nov 15, 2011 at 7:05 PM, Leon de Rooij wrote: > Hello Sammy, > > Well, Kamailio and OpenSIPS are essentially the same, eh. They are both forks from the same codebase. > > Kind regards, > > Leon > > > On Nov 15, 2011, at 12:33 PM, Sammy Govind wrote: > >> Looking at the original topic and the opensips.cfg really confused me that we started from kamailio and ended up in openSIPS ! >> >> On Tue, Nov 15, 2011 at 3:50 PM, Leon de Rooij wrote: >> Hello Vik and Matt, >> >> Of course, here's my current opensips.cfg, but note that it contains bugs and is far from ready, just call setup/teardown and registrations work. >> >> Also, I have a dispatcher.list containing one set ("1") pointing to two sip profiles (at different udp ports) on one fs instance. Both sip profiles have the outbound-proxy param set to the sip uri of the proxy. >> >> kind regards, >> >> Leon >> >> >> >> >> >> On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote: >> >>> I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well >>> >>> On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa wrote: >>> Any chance you can post your opensips config? I've been struggling >>> with mixed success trying same setup. >>> >>> On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij >>> wrote: >>> > Hi, >>> > I'm actually trying to configure the same thing right now - with opensips >>> > though, but (afaik) it uses the same dispatcher module. >>> > First I used a simple route script in opensips with using dispatcher, but >>> > after the first message (from ua through proxy to fs), the proxy would get >>> > out of the signaling path, while I want it to stay in. >>> > To fix that, I added record routing in the proxy configuration so it stays >>> > in the path. >>> > Registrations are also balanced towards the fs servers - but some clients >>> > are nat'ed so the contact header was wrong - fix_nated_contact() in route >>> > and in onreply_route fixed that. >>> > To get calls originating from fs to go through the proxy before going to the >>> > client, I tried: >>> > originate >>> > sofia/some_profile/sip:user at mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 >>> > &park >>> > which worked. Then I found out about the sip_route_uri variable that can be >>> > set in the bridge string: >>> > originate >>> > {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user at mydomain.com >>> > &park >>> > which seemed to do the same thing (does anyone know if there's any >>> > difference between the two variants ?) >>> > Then I tried putting the sip_route_uri in the dial-string param of the >>> > domain (or user), for example: >>> > >> > value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> >>> > so now it's possible to just call the user as: >>> > originate user/some_user at mydomain.com &park >>> > nice ! >>> > And then I found out that you can just set an outbound-proxy param in the >>> > sip profile that's used for originating: >>> > >>> > so now you can leave out the sip_route_uri or fs_path variables, it just >>> > works. >>> > My proxy config still needs a lot of work, but I got the basic functionality >>> > working, still need to find out about using acls on fs side based on a sip >>> > header added by the proxy - I'm already adding X-AUTH-IP headers on messages >>> > from clients towards fs which should do the trick, but I didn't test it yet. >>> > Anyway, would love to read some configs of ppl who successfuly setup >>> > opensips/kamailio/openser/... proxies in front of fs, what their experience >>> > is with using record-routing or not, etc. >>> > regards, >>> > Leon >>> > >>> > >>> > >>> > On Nov 14, 2011, at 6:40 AM, Henrik Aagaard S?rensen wrote: >>> > >>> > The handling of several FS are not a issue, actually easy enough. >>> > I would like to have as little load on Kamailio as possible, as it just >>> > should load balance. >>> > Also, having to handle users on both Kamailio and FS makes unecessary work >>> > loads. >>> > >>> > On 14/11/2011, at 00.32, Sammy Govind wrote: >>> > >>> > Hi again, >>> > Why don't you just let Kamailio handle registrations. Anyway I was thinking >>> > about that LBing the registration would result in such a scenario that >>> > calling one extension to another would not make a successful call because >>> > the other endpoint maybe registered on some other FS. >>> > This may further lead you to making a dial-plan which would work somewhat >>> > like DUNDI but it'd just have to search all the FS servers before joining a >>> > call.(Obv there are other intelligent approaches to minimize the headache) >>> > Try following the link I sent you and implement that in front of your FS, I >>> > think that Kamailio configuration is so well written that anyone can start >>> > understanding kamailio and implement such setups with little effort. >>> > By explaining your topology I meant how do you plan to use Kamailio in front >>> > of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a >>> > topo-hiding or SBC like setup! >>> > -- >>> > Regards, >>> > Sammy >>> > >>> > 2011/11/14 Henrik Aagaard S?rensen >>> >> >>> >> Hi Sammy. >>> >> I've actually removed registration and presence from Kamailio, so all it >>> >> does is dispatch everything to FreeSwitch. >>> >> Currently I only have 1 FreeSwitch, for testing this basic setup. >>> >> Next move would be 2 FreeSwitch with 1 common database etc. But that's >>> >> later. >>> >> My FreeSwitch is the basic setup, without anything else. Just as the >>> >> installation manuel is written. So I have my extension 1000 - 1019 etc. And >>> >> calls between them works when connected directly to FreeSwitch. >>> >> But when going through Kamailio Dispatcher it fails between the >>> >> extensions. >>> >> So I guess there should be some more setup in FreeSwitch when using a load >>> >> balancer (dispatcher) in front of it. >>> >> My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle >>> >> everything. >>> >> >>> >> On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind wrote: >>> >>> >>> >>> Hi, >>> >>> If everything is setup as you expected it then don't read this >>> >>> > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, >>> >>> else follow the article from start till end and implement it. Don't worry >>> >>> about the name asterisk, just replace that all with FreeSWICTH. It will work >>> >>> like that too. >>> >>> Once that is implemented then apply dispatcher module on the route which >>> >>> says [REGFWD] or [TOASTERISK]. >>> >>> Also explain your topology abit more and features required as well that >>> >>> may help in telling which extra module you will require in order to make >>> >>> things work. >>> >>> In your current implementation are you sure SIP phones are registering on >>> >>> FS and not on Kamailio?and that both end points making call to each other >>> >>> are on same FS? >>> >>> -- >>> >>> Regards, >>> >>> Sammy >>> >>> >>> >>> 2011/11/14 Henrik Aagaard S?rensen >>> >>>> >>> >>>> Hi everyone. In regards to my earlier question regarding with FreeSwitch >>> >>>> behind Kamailio Dispatcher, I've attached a call from extension 1001 to >>> >>>> 1002, which fails. It just hangs for some time and then says that 1002 >>> >>>> cannot be found, and then the voicemail for it comes up. >>> >>>> 2011/11/13 Henrik Aagaard S?rensen >>> >>>>> >>> >>>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. >>> >>>>> I've setup Kamailio via >>> >>>>> this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >>> >>>>> I've installed FreeSwitch from scratch on Ubuntu >>> >>>>> via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >>> >>>>> Now, when registering extensions via Kamailio Dispatcher I'm able to >>> >>>>> call to FreeSwitch and listen to hold music. But that's it. I'm not able to >>> >>>>> call between extensions etc. >>> >>>>> Can anyone help me setting up FreeSwitch to accept registration, calls >>> >>>>> etc. from Kamailio and everything else that is needed to use FreeSwitch >>> >>>>> behind a load balancer? >>> >>>>> I'm very new to FreeSwitch, but I'm trying to use the terminal (without >>> >>>>> any GUI etc.) as I want the installation to be as clean as possible. So I >>> >>>>> would prefer very precise help, as I'm still getting hold of FreeSwitch. >>> >>>> >>> >>>> >>> >>>> _________________________________________________________________________ >>> >>>> Professional FreeSWITCH Consulting Services: >>> >>>> consulting at freeswitch.org >>> >>>> http://www.freeswitchsolutions.com >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> Official FreeSWITCH Sites >>> >>>> http://www.freeswitch.org >>> >>>> http://wiki.freeswitch.org >>> >>>> http://www.cluecon.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 >>> >>>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.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 >>> >>> >>> >> >>> >> >>> >> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.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 >>> >> >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.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 >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.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 >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.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 >>> > >>> > >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/102b219d/attachment-0001.html From msc at freeswitch.org Tue Nov 15 19:32:55 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 08:32:55 -0800 Subject: [Freeswitch-users] sendevent issues In-Reply-To: <4EC284AB.7070509@newpace.ca> References: <4EC284AB.7070509@newpace.ca> Message-ID: I just tested this and it worked for me. However, instead of doing "/events all" i did "/events plain all". I see all the events on the system, including the ones injected. -MC On Tue, Nov 15, 2011 at 7:26 AM, Adam Kelloway wrote: > Hi there, > > I have client connection to a freeswitch instance, and send the > following message to freeswitch from it: > > sendevent CUSTOM\n > Event-Name: CUSTOM\n > Event-Subclass: my::event\n > my-header: my-value\n > \n > > I receive the following reply: > > Content-Type: command/reply > Reply-Text: +OK > > I subscribe to all events from fs_cli, using: > > /events all > > However, I never see the event that was sent. Am I misinterpreting how > sendevent is used? I would expect that the console application would see > this event. Am I perhaps missing some mandatory headers? > > Thanks, > > Adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/fe6174fd/attachment.html From anthony.minessale at gmail.com Tue Nov 15 19:33:47 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Nov 2011 10:33:47 -0600 Subject: [Freeswitch-users] Howto set only one console debug level messages in fs_cli In-Reply-To: References: <20111114175038.7DBABD6C@centrum.cz> Message-ID: When you set the default level it will show everything above that level. Try these commands: >console loglevel info will only show info and console levels >console loglevel err will only show error and warning and console On Mon, Nov 14, 2011 at 11:20 AM, Michael Collins wrote: > Hi Vidan, > > I recommend you look in conf/autoload_configs/console.conf.xml. You can > look at the sample mappings that are commented out. About the best you can > do is to specify the log levels for the various files that you need. I know > it's hard, but it's all the control that is available at the moment. > > -MC > > > On Mon, Nov 14, 2011 at 8:50 AM, Vidan wrote: > >> >> Hi, >> I am newbie on FreeSwitch, >> I would like show only the INFO(6) messages in fs_cli console, not any >> others (err, warning etc.). I am tried to modify switch.conf.xml, var.xml >> and console.conf.xml and switch.conf.xml. I see alway all type of messages >> until defined console loglevel value. works >> fine only in logfile.conf.xml. The same line does not work in my case in >> console.conf.xml for fs_cli. >> For example I will set "ERR" (3) and I will see on Fs_cli "CONSOLE" >> (0)+"ALERT" (1)+"CRIT" (2) messages. Is it possible to set on screen only >> one messages layer in fsi_cli? How is it possible to do that? >> >> I am doing this because alway I had full screen with this: >> >> 011-11-14 16:39:22.230114 [WARNING] ftmod_wanpipe.c:917 [s1c1][1:1] Rx >> Queue length exceeded 80% hreshold (9/10) >> 2011-11-14 16:39:22.230114 [WARNING] ftmod_wanpipe.c:917 [s1c1][1:1] Rx >> Queue length exceeded 80% hreshold (8/10) >> 2011-11-14 16:39:24.610115 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx >> idle changed from 250 to 294 >> 2011-11-14 16:39:24.630115 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx >> idle changed from 38 to 294 >> 2011-11-14 16:39:24.650130 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx >> idle changed from 38 to 294 >> 2011-11-14 16:39:24.670115 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx >> idle changed from 38 to 294 >> 2011-11-14 16:39:24.690126 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx >> idle changed from 38 to 294 >> 2011-11-14 16:39:24.710144 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx >> idle changed from 38 to 294 >> 2011-11-14 16:39:24.730125 [WARNING] ftmod_wanpipe.c:864 [s3c1][3:1] Tx >> idle changed from 38 to 294 >> >> and I need only notice and info messages from isdn. >> >> Thank you for answer. >> Best Regard, >> Vit >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/f0fcb229/attachment.html From turqmr2 at gmail.com Tue Nov 15 20:21:54 2011 From: turqmr2 at gmail.com (Jacob Smith) Date: Tue, 15 Nov 2011 12:21:54 -0500 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: <7f201b60894d031ebed132ca359a8f6b@mail.gmail.com> <4EC2796E.40405@gmail.com> Message-ID: <4EC29FB2.6020303@gmail.com> I skipped the step of providing the mod_dingaling.dll. I found this line in the log: 2011-11-14 21:21:15.974250 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\FreeSWITCH\mod\mod_dingaling.dll **dll open error [126l] Thanks for responding, I will have an intelligent question soon enough. On 11/15/2011 11:21 AM, Michael Collins wrote: > I would guess that you have an issue in your dialplan somewhere. > Notice that there is no gateway name specified in this bridge call: > > EXECUTE sofia/internal/1000 at 192.168.0.5 > bridge(sofia/gateway//15551234567) > > > If you need help looking at your dialplan then throw it on > pastebin.freeswitch.org and we'll > take a peek at it. > -MC > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/a376b761/attachment.html From cmrienzo at gmail.com Tue Nov 15 23:05:22 2011 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 15 Nov 2011 15:05:22 -0500 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <4EC14A92.1040704@hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> Message-ID: Latest version of FreeSWITCH now has http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_detect_speech Give it a try and let us know if there is anything that can be improved. On Mon, Nov 14, 2011 at 12:06 PM, xl127 wrote: > The way you did and described here is similar to what I wanted. > I want to set start-input-timers=false in the initial setup, then start > the recognition timer once the TTS finishes playing the prompt. > I do not use input callback as I am using ESL event (Java ESL Client > 0.9.2). > > I've not yet figured out how to know the prompt playing has finished, how > to stop the TTS and how to start the input timer. > I didn't get many events when I did barge-in: only got two CHANNEL_EXECUTE > and CAHNNEL_EXECUTE_COMPLETE events. > so didn't get the DETECTED_SPEECH at this point. > > In principle it shouldn't stop FS generating events when barging in, so > there must be somethings wrong in my setup. > > From Examples_directory_lua_asr_tts in the wiki, it mentions the session > needs to sleep for a few seconds before getting the results. > I tried but it didn't help. > > > > > On 14/11/11 14:48, Christopher Rienzo wrote: > > I'm not sure why it locks up for you. It works for me and I know of at > least one other developer that got it all to work. > > The way I do it in my custom APP is to set {start-input-timers=false} in > the speech recognition request and register an input callback function to > deal with the recognition and DTMF events. The input callback reacts to > the start of speech and recognition complete events by stopping the > prompt. At start of speech, the input timers are started and silence is > played until the final result arrives from the recognizer (no match, match, > etc). > > An alternative way to do this is to watch for the recognition events over > ESL and react to them. > > > > On Mon, Nov 14, 2011 at 9:19 AM, xl127 wrote: > >> Hi Christopher, >> >> That will be great! Thanks and looking forward to your app! >> >> By the way, do you have a quick thoughts about the reasons the FS stucks >> when garge-in occurs? >> Otherwise if it is not straightforward to explain it, just leave it >> alone until your new app comes out. >> >> Cheers, >> Xing >> >> >> >> >> On 14/11/11 13:23, Christopher Rienzo wrote: >> >> I'll write up a dialplan app this week to deal with this. Getting tired >> of being asked the same question over and over since it's too complicated >> as is currently designed :) >> >> >> >> >> On Mon, Nov 14, 2011 at 8:07 AM, xl127 wrote: >> >>> Hi, >>> >>> The first problem I need to solve is that FS stucks when I speak during >>> the prompt is playing. >>> It looks like the recognizer is not listening any more. It works fine if >>> I speak after the playing prompt finishes. >>> >>> I tried Nuance and PocketSphinx recognizers and got same problem. >>> >>> Any idea about what the possible causes are? >>> >>> Thanks! >>> Xing >>> >>> >>> On 11/11/11 18:59, xl127 wrote: >>> > Hello, >>> > >>> > I found this is a question that was asked before by others, but I >>> didn't >>> > find the answer. >>> > >>> > Anyway, I am using FS ESL outbound mode connecting to my IVR app, >>> using >>> > FS's "speak" >>> > and "detect_speech" to access Nuance MRCP V1 server. >>> > >>> > I want the user to be able to barge-in during the system's speaking. >>> > How could I implement it? >>> > >>> > I tried to specify "kill-on-barge-in=true" in the mrcp config profile. >>> > The barge-in doesn't work. >>> > >>> > With or without setting kill-on-barge-in, FS stops responding to my >>> > phone call and >>> > eventually it hangs up my call if I speak somthing (do the barge-in) >>> > during the system's speaking. >>> > >>> > I made a turn-by-turn loop in my app, the ASR/TTS works fine if I do >>> not >>> > do barge-in ( I wait until the TTS finishes then I start to speak) >>> > >>> > Any advices please? >>> > >>> > Thanks! >>> > Xing >>> > >>> > >>> > >>> >>> >>> >>> -- >>> Heriot-Watt University is a Scottish charity >>> registered under charity number SC000278. >>> >>> Heriot-Watt University is the Sunday Times >>> Scottish University of the Year 2011-2012 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> ------------------------------ >> >> [image: Scottish University of the Year 2011-12] *Heriot-Watt >> University is the Sunday Times >> >> Scottish University of the Year 2011-2012 >> * >> >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > ------------------------------ > > [image: Scottish University of the Year 2011-12] *Heriot-Watt > University is the Sunday Times > Scottish University of the Year 2011-2012* > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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... 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Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/0d568c24/attachment-0003.jpe From notlikeme75 at yahoo.com Wed Nov 16 01:15:03 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 15 Nov 2011 14:15:03 -0800 (PST) Subject: [Freeswitch-users] update freeswitch on windows Message-ID: <1321395303.55295.YahooMailNeo@web65314.mail.ac2.yahoo.com> what is the proper way to update freeswitch on windows. i would like to update to the latest msi every week while in development and maybe on a schedule after deployment. i am using 32 bit windows server 2008 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/ca1c836f/attachment.html From notlikeme75 at yahoo.com Wed Nov 16 01:19:17 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 15 Nov 2011 14:19:17 -0800 (PST) Subject: [Freeswitch-users] tts on windows Message-ID: <1321395557.94056.YahooMailNeo@web65301.mail.ac2.yahoo.com> i am using the nov 4th msi of fs. none of my dialplan that require tts work. how do i get flite to work on windows. and does anyone know the reason why this is not set up out of the box. it seems a very important part of the software. error opening mod_flite.dll as its not in the package and i do not know where to get it. windows server 2008 32bit -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/0b17bc1d/attachment.html From msc at freeswitch.org Wed Nov 16 02:20:05 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 15:20:05 -0800 Subject: [Freeswitch-users] tts on windows In-Reply-To: <1321395557.94056.YahooMailNeo@web65301.mail.ac2.yahoo.com> References: <1321395557.94056.YahooMailNeo@web65301.mail.ac2.yahoo.com> Message-ID: Rodney, Hello! I see you've sent like 6 messages to the mailing list. New members are moderated on their first posts and I've let your first two questions go through. Let's slow down a bit and take them one at a time. Keep in mind that we have several Windows gurus, but the vast majority of our active community members are using Linux. Therefore, it may take a bit longer to get your Windows-specific questions answered. That being said, mod_flite *should* be included with the download and it should also build automatically. If it isn't then there's definitely something amiss. I'll have to defer to Jeff Lenk, our resident Windows build system guru. -MC On Tue, Nov 15, 2011 at 2:19 PM, Rodney wrote: > i am using the nov 4th msi of fs. none of my dialplan that require tts > work. how do i get flite to work on windows. and does anyone know the > reason why this is not set up out of the box. it seems a very important > part of the software. > > error opening mod_flite.dll as its not in the package and i do not know > where to get it. > > > windows server 2008 32bit > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/07bef1ac/attachment.html From msc at freeswitch.org Wed Nov 16 02:21:47 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 15:21:47 -0800 Subject: [Freeswitch-users] update freeswitch on windows In-Reply-To: <1321395303.55295.YahooMailNeo@web65314.mail.ac2.yahoo.com> References: <1321395303.55295.YahooMailNeo@web65314.mail.ac2.yahoo.com> Message-ID: The "proper" way is to use the latest git, although the Windows guys can probably tell you if using the weekly MSI files are doable. -MC On Tue, Nov 15, 2011 at 2:15 PM, Rodney wrote: > what is the proper way to update freeswitch on windows. i would like to > update to the latest msi every week while in development and maybe on a > schedule after deployment. > > i am using 32 bit windows server 2008 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/629b662d/attachment.html From notlikeme75 at yahoo.com Wed Nov 16 01:22:33 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 15 Nov 2011 14:22:33 -0800 (PST) Subject: [Freeswitch-users] block unknown callers Message-ID: <1321395753.10566.YahooMailNeo@web65306.mail.ac2.yahoo.com> I am using ipkall for one of my gateways and it does not pass flex ani. so they are allowed to block themselves . what can i add before or after the answer condition so it transfers the call to a toll free trap that forwards unblocked ball back to the dialplan or either transfers them to a local message that tells them to call back unblocked then hangs up. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/e51ece6c/attachment.html From msc at freeswitch.org Wed Nov 16 02:29:35 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 15:29:35 -0800 Subject: [Freeswitch-users] block unknown callers In-Reply-To: <1321395753.10566.YahooMailNeo@web65306.mail.ac2.yahoo.com> References: <1321395753.10566.YahooMailNeo@web65306.mail.ac2.yahoo.com> Message-ID: Here's a clue on how to react based on the caller id: http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_16:_Block_certain_codes It's just a matter of matching on what is (or isn't) being sent and then playing your "blocked caller ID not accepted" message before hanging up. -MC On Tue, Nov 15, 2011 at 2:22 PM, Rodney wrote: > I am using ipkall for one of my gateways and it does not pass flex ani. so > they are allowed to block themselves . what can i add before or after the > answer condition so it transfers the call to a toll free trap that forwards > unblocked ball back to the dialplan or either transfers them to a local > message that tells them to call back unblocked then hangs up. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/fc237805/attachment.html From notlikeme75 at yahoo.com Wed Nov 16 02:06:27 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 15 Nov 2011 15:06:27 -0800 (PST) Subject: [Freeswitch-users] method for creating public pins used for private conference 1 on 1 Message-ID: <1321398387.25593.YahooMailNeo@web65301.mail.ac2.yahoo.com> windows server 2008 32bit I would like to have each of my users who call my ivr get assigned a public pin random code 3 digits that they could exchange with other callers in a public chat rooms that will allow them to go to a private conference setup just for them. scenario: caller 1 calls , ivr automatically assigns pin? #123 caller 2 calls, ivr automatically assigns pin#124 caller 1 and 2 are talking with callers 3, 4, 5 in conference room 1 but caller 1 and 2 are interested in talking private so caller 1 presses a dtmf option that asks them what the other callers' (caller 2) pin # is? . the system parks him until caller 2 does the same, then the system puts them into a private conference. so what i think i need is : a. method for assigning pins to every caller to ivr b. method for capturing that pin when asked and matching with parked user who has done the same c. method for transferring them into the same 1 on 1 conference.? a demo of how this works on a similar line is 218-895-2080 any help would be greatly appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/04053ff0/attachment-0001.html From notlikeme75 at yahoo.com Wed Nov 16 01:32:10 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 15 Nov 2011 14:32:10 -0800 (PST) Subject: [Freeswitch-users] list of conference rooms . "next room command" Message-ID: <1321396330.62279.YahooMailNeo@web65314.mail.ac2.yahoo.com> scenario i have a conference / voice personals dating line that has a list of conferences . in this case they are labeled conference room 1 conference room 2 conference room 3 etc users select dtmf 1-19 from a conference menu and get put into the conference room. i want to setup options for them to move forward and backward through this list but since the conf module profile binds digits to every conference i currently would have to setup up a separate profile for each conference room in order for me to make bind digit 7 move forward in the list to the "next room" or 6 to go to "previous" room number . ie. is there a variable that would take the room im in say and make dtmf 6 mean transfer me to room 1 backwards 1 getting me to room 19 without going back to conference menu and pressing 7 moves me forward in the list numerically without having to go back to conference menu and choosing a new dtmf press 7 from room 1, you go to room 2 press? 7 from room 2, you go to room 3, etc? press 6 from room 1 , you go to 19 press 6 from room 9 , you go to 8 but if i only have 1 profile "default" i can only use that bind digit to move to a static conference in the ivr. please tell me that i am missing the understanding? and there is an easy way to do this with one conference profile. thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/9fd2444f/attachment.html From msc at freeswitch.org Wed Nov 16 02:38:47 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 15:38:47 -0800 Subject: [Freeswitch-users] method for creating public pins used for private conference 1 on 1 In-Reply-To: <1321398387.25593.YahooMailNeo@web65301.mail.ac2.yahoo.com> References: <1321398387.25593.YahooMailNeo@web65301.mail.ac2.yahoo.com> Message-ID: I highly recommend you do 3 things: #1 - Look up "bind_digit_action" on the wiki - it has lots of powerful features #2 - Look at the default.xml dialplan file as it has lots of samples #3 - Get our book. We spent a lot of time writing it and it will give you the foundation you need to build what you're building. -MC On Tue, Nov 15, 2011 at 3:06 PM, Rodney wrote: > windows server 2008 32bit > > I would like to have each of my users who call my ivr get assigned a > public pin random code 3 digits that they could exchange with other callers > in a public chat rooms that will allow them to go to a private conference > setup just for them. > > scenario: > > caller 1 calls , ivr automatically assigns pin #123 > caller 2 calls, ivr automatically assigns pin#124 > > > caller 1 and 2 are talking with callers 3, 4, 5 in conference room 1 but > caller 1 and 2 are interested in talking private so caller 1 presses a dtmf > option that asks them what the other callers' (caller 2) pin # is . the > system parks him until caller 2 does the same, then the system puts them > into a private conference. > > so what i think i need is : > > a. method for assigning pins to every caller to ivr > b. method for capturing that pin when asked and matching with parked user > who has done the same > c. method for transferring them into the same 1 on 1 conference. > > a demo of how this works on a similar line is 218-895-2080 > > any help would be greatly appreciated. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/32075c2b/attachment.html From notlikeme75 at yahoo.com Wed Nov 16 02:15:10 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 15 Nov 2011 15:15:10 -0800 (PST) Subject: [Freeswitch-users] voice bulletin board system Message-ID: <1321398910.24875.YahooMailNeo@web65316.mail.ac2.yahoo.com> system= windows server 2008 32bit I would like a method for a voice bulletin board system. This would allow my callers to system the ability to post a message of a stated length to a public "bulletin board" scenario caller dials ivr option for voice bbs . options are 1 to play the last message recorded through the first recorded, option 2 next message, and 3 to record so lifo? i would need to set a maximum of say "20" messages" so it deletes the oldest . i would also want to limit it to 3 or 5 posts per call from 1 individual. also would like a spam option for auto delete. 1. method to record? 2. method to review that recording and give options 2 to repeat or 3 to save and 0 to cancel and go back to ivr asking if they want to record or play 3. method to play LIFO messages up to 20, with auto delete 4. method for callers to hit dtmf option 9 to mark as spam. if 2 or 3 unique callers mark as spam then message gets auto deleted.? demo of what i need programmed is 218-895-2080 this system is propitiatory and i am trying to duplicate using freeswitch. any help would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/bb0bb800/attachment.html From msc at freeswitch.org Wed Nov 16 02:42:58 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 15:42:58 -0800 Subject: [Freeswitch-users] list of conference rooms . "next room command" In-Reply-To: <1321396330.62279.YahooMailNeo@web65314.mail.ac2.yahoo.com> References: <1321396330.62279.YahooMailNeo@web65314.mail.ac2.yahoo.com> Message-ID: See my other email, but basically you want to do research on "bind_digit_action" -MC On Tue, Nov 15, 2011 at 2:32 PM, Rodney wrote: > scenario > > i have a conference / voice personals dating line that has a list of > conferences . in this case they are labeled > > conference room 1 > conference room 2 > conference room 3 > etc > > users select dtmf 1-19 from a conference menu and get put into the > conference room. i want to setup options for them to move forward and > backward through this list but since the conf module profile binds digits > to every conference i currently would have to setup up a separate profile > for each conference room in order for me to make bind digit 7 move forward > in the list to the "next room" or 6 to go to "previous" room number . ie. > is there a variable that would take the room im in say and make dtmf 6 mean > transfer me to room 1 backwards 1 getting me to room 19 without going back > to conference menu and pressing 7 moves me forward in the list numerically > without having to go back to conference menu and choosing a new dtmf > > press 7 from room 1, you go to room 2 > press 7 from room 2, you go to room 3, etc > > press 6 from room 1 , you go to 19 > press 6 from room 9 , you go to 8 > > but if i only have 1 profile "default" i can only use that bind digit to > move to a static conference in the ivr. > > please tell me that i am missing the understanding and there is an easy > way to do this with one conference profile. thank you > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/24eab84d/attachment.html From notlikeme75 at yahoo.com Wed Nov 16 02:21:39 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 15 Nov 2011 15:21:39 -0800 (PST) Subject: [Freeswitch-users] mod conference? method to auto mute /kick Message-ID: <1321399299.38607.YahooMailNeo@web65303.mail.ac2.yahoo.com> I would like a method to set an max energy threshold? to help with auto moderating conference channels. the this would work is if someone wanted to be annoying in a conference room and be loud with music or three waying annoying phone company tones then the system would auto mute the caller until he realizes he was muted and hangs up or presses bind digit to unmute himself. is this method already available or do you have any thoughts on how to implement such an option. system= windows server 2008 32bit -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/9a46144e/attachment-0001.html From msc at freeswitch.org Wed Nov 16 02:53:40 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 15:53:40 -0800 Subject: [Freeswitch-users] voice bulletin board system In-Reply-To: <1321398910.24875.YahooMailNeo@web65316.mail.ac2.yahoo.com> References: <1321398910.24875.YahooMailNeo@web65316.mail.ac2.yahoo.com> Message-ID: > > > this system is propitiatory and i am trying to duplicate using freeswitch. > any help would be appreciated. > I'm probably the only person on this list who knows what "propitiatory" means and I'm pretty sure that the system you've described is not propitiatory at all. :P I'm guessing this is a proprietary system that you're trying to create in FreeSWITCH. You've got a fair amount of work ahead of you. I hope you are, or have access to, a skilled programmer. You might also want to check in with consulting at freeswitch.org for some paid professional assistance. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/cf12c6db/attachment.html From msc at freeswitch.org Wed Nov 16 02:58:19 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 15:58:19 -0800 Subject: [Freeswitch-users] mod conference? method to auto mute /kick In-Reply-To: <1321399299.38607.YahooMailNeo@web65303.mail.ac2.yahoo.com> References: <1321399299.38607.YahooMailNeo@web65303.mail.ac2.yahoo.com> Message-ID: This is actually an intriguing idea. I don't know how practical it would be in production, but it's definitely a curious thought. There's nothing like that at the moment, so your best bet there would be to open a bounty on jira.freeswitch.org. -MC On Tue, Nov 15, 2011 at 3:21 PM, Rodney wrote: > I would like a method to set an max energy threshold to help with auto > moderating conference channels. the this would work is if someone wanted to > be annoying in a conference room and be loud with music or three waying > annoying phone company tones then the system would auto mute the caller > until he realizes he was muted and hangs up or presses bind digit to unmute > himself. > > is this method already available or do you have any thoughts on how to > implement such an option. > > system= windows server 2008 32bit > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/32f69a6b/attachment.html From anthony.minessale at gmail.com Wed Nov 16 03:20:51 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Nov 2011 18:20:51 -0600 Subject: [Freeswitch-users] Help Wanted [Consultants and Developers] Message-ID: Anyone out there interested in getting some consulting leads and doing regular work regarding FS or anyone who is ambitious and interested in a larger more permanent position doing development frontend and backend for Barracuda Networks drop us a line. email jobs at freeswitch.org with relevant contact info and CV -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/50fcc0df/attachment.html From cmrienzo at gmail.com Wed Nov 16 03:40:15 2011 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Tue, 15 Nov 2011 19:40:15 -0500 Subject: [Freeswitch-users] mod conference? method to auto mute /kick In-Reply-To: References: <1321399299.38607.YahooMailNeo@web65303.mail.ac2.yahoo.com> Message-ID: A media bug on a conference member's channel could monitor energy to kick him if a threshold was exceeded for some time. On Nov 15, 2011, at 6:58 PM, Michael Collins wrote: > This is actually an intriguing idea. I don't know how practical it would be in production, but it's definitely a curious thought. There's nothing like that at the moment, so your best bet there would be to open a bounty on jira.freeswitch.org. > > -MC > > On Tue, Nov 15, 2011 at 3:21 PM, Rodney wrote: > I would like a method to set an max energy threshold to help with auto moderating conference channels. the this would work is if someone wanted to be annoying in a conference room and be loud with music or three waying annoying phone company tones then the system would auto mute the caller until he realizes he was muted and hangs up or presses bind digit to unmute himself. > > is this method already available or do you have any thoughts on how to implement such an option. > > system= windows server 2008 32bit > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/4ccd1bee/attachment.html From brad at tech21.com Wed Nov 16 03:40:38 2011 From: brad at tech21.com (Brad Mina) Date: Tue, 15 Nov 2011 16:40:38 -0800 Subject: [Freeswitch-users] method for creating public pins used for private conference 1 on 1 In-Reply-To: References: <1321398387.25593.YahooMailNeo@web65301.mail.ac2.yahoo.com> Message-ID: Rodney, Might I support Michael's recommendation in investing in the FreeSWITCH book? That is, if you don't already have it. Michael, You're as helpful as ever! On Tue, Nov 15, 2011 at 3:38 PM, Michael Collins wrote: > I highly recommend you do 3 things: > > #1 - Look up "bind_digit_action" on the wiki - it has lots of powerful > features > #2 - Look at the default.xml dialplan file as it has lots of samples > #3 - Get our book. We spent a lot of time writing it and it will give you > the foundation you need to build what you're building. > > -MC > > On Tue, Nov 15, 2011 at 3:06 PM, Rodney wrote: > >> windows server 2008 32bit >> >> I would like to have each of my users who call my ivr get assigned a >> public pin random code 3 digits that they could exchange with other callers >> in a public chat rooms that will allow them to go to a private conference >> setup just for them. >> >> scenario: >> >> caller 1 calls , ivr automatically assigns pin #123 >> caller 2 calls, ivr automatically assigns pin#124 >> >> >> caller 1 and 2 are talking with callers 3, 4, 5 in conference room 1 but >> caller 1 and 2 are interested in talking private so caller 1 presses a dtmf >> option that asks them what the other callers' (caller 2) pin # is . the >> system parks him until caller 2 does the same, then the system puts them >> into a private conference. >> >> so what i think i need is : >> >> a. method for assigning pins to every caller to ivr >> b. method for capturing that pin when asked and matching with parked user >> who has done the same >> c. method for transferring them into the same 1 on 1 conference. >> >> a demo of how this works on a similar line is 218-895-2080 >> >> any help would be greatly appreciated. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/7fbba8a9/attachment-0001.html From msc at freeswitch.org Wed Nov 16 04:21:20 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 17:21:20 -0800 Subject: [Freeswitch-users] mod conference? method to auto mute /kick In-Reply-To: References: <1321399299.38607.YahooMailNeo@web65303.mail.ac2.yahoo.com> Message-ID: Yep - FS makes things like this "easy." Of course, the question isn't whether it can be done, the question is whether or not it will work well in an environment where lots of different people from lots of different environments are all calling in to a single conference. What is loud to one person may not be to another. In any case, if someone is feeling frisky he (or she) is more than welcome to do the patch. -MC On Tue, Nov 15, 2011 at 4:40 PM, wrote: > A media bug on a conference member's channel could monitor energy to kick > him if a threshold was exceeded for some time. > > > On Nov 15, 2011, at 6:58 PM, Michael Collins wrote: > > This is actually an intriguing idea. I don't know how practical it would > be in production, but it's definitely a curious thought. There's nothing > like that at the moment, so your best bet there would be to open a bounty > on jira.freeswitch.org. > > -MC > > On Tue, Nov 15, 2011 at 3:21 PM, Rodney wrote: > >> I would like a method to set an max energy threshold to help with auto >> moderating conference channels. the this would work is if someone wanted to >> be annoying in a conference room and be loud with music or three waying >> annoying phone company tones then the system would auto mute the caller >> until he realizes he was muted and hangs up or presses bind digit to unmute >> himself. >> >> is this method already available or do you have any thoughts on how to >> implement such an option. >> >> system= windows server 2008 32bit >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111115/2c711e39/attachment.html From notlikeme75 at yahoo.com Wed Nov 16 04:49:18 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 15 Nov 2011 17:49:18 -0800 (PST) Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 65, Issue 104 In-Reply-To: References: Message-ID: <1321408158.2988.YahooMailNeo@web65316.mail.ac2.yahoo.com> this is the error i get on nov 4th msi install on windows server 2008 for flite.dll , none of my tts works 2011-11-2011 20:35:07.435832 {crit} switch_loadable_module.c:1281 error loading module c:\program files\files\freeswtich\mod\mod_flite.dll ***dll open error {1261} ** ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, November 15, 2011 6:32 PM Subject: FreeSWITCH-users Digest, Vol 65, Issue 104 ----- Forwarded Message ----- 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. update freeswitch on windows (Rodney) ? 2. tts on windows (Rodney) ? 3. Re: tts on windows (Michael Collins) ? 4. Re: update freeswitch on windows (Michael Collins) ? 5. block unknown callers (Rodney) ? 6. Re: block unknown callers (Michael Collins) ? 7. method for creating public pins used for private??? conference 1 ? ? ? on 1 (Rodney) what is the proper way to update freeswitch on windows. i would like to update to the latest msi every week while in development and maybe on a schedule after deployment. i am using 32 bit windows server 2008 i am using the nov 4th msi of fs. none of my dialplan that require tts work. how do i get flite to work on windows. and does anyone know the reason why this is not set up out of the box. it seems a very important part of the software. error opening mod_flite.dll as its not in the package and i do not know where to get it. windows server 2008 32bit Rodney, Hello! I see you've sent like 6 messages to the mailing list. New members are moderated on their first posts and I've let your first two questions go through. Let's slow down a bit and take them one at a time. Keep in mind that we have several Windows gurus, but the vast majority of our active community members are using Linux. Therefore, it may take a bit longer to get your Windows-specific questions answered. That being said, mod_flite *should* be included with the download and it should also build automatically. If it isn't then there's definitely something amiss. I'll have to defer to Jeff Lenk, our resident Windows build system guru. -MC On Tue, Nov 15, 2011 at 2:19 PM, Rodney wrote: i am using the nov 4th msi of fs. none of my dialplan that require tts work. how do i get flite to work on windows. and does anyone know the reason why this is not set up out of the box. it seems a very important part of the software. > > >error opening mod_flite.dll as its not in the package and i do not know where to get it. > > > > > >windows server 2008 32bit > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.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 > > The "proper" way is to use the latest git, although the Windows guys can probably tell you if using the weekly MSI files are doable. -MC On Tue, Nov 15, 2011 at 2:15 PM, Rodney wrote: what is the proper way to update freeswitch on windows. i would like to update to the latest msi every week while in development and maybe on a schedule after deployment. > > > >i am using 32 bit windows server 2008 > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.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 > > I am using ipkall for one of my gateways and it does not pass flex ani. so they are allowed to block themselves . what can i add before or after the answer condition so it transfers the call to a toll free trap that forwards unblocked ball back to the dialplan or either transfers them to a local message that tells them to call back unblocked then hangs up. Here's a clue on how to react based on the caller id: http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_16:_Block_certain_codes It's just a matter of matching on what is (or isn't) being sent and then playing your "blocked caller ID not accepted" message before hanging up. -MC On Tue, Nov 15, 2011 at 2:22 PM, Rodney wrote: I am using ipkall for one of my gateways and it does not pass flex ani. so they are allowed to block themselves . what can i add before or after the answer condition so it transfers the call to a toll free trap that forwards unblocked ball back to the dialplan or either transfers them to a local message that tells them to call back unblocked then hangs up. > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.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 > > windows server 2008 32bit I would like to have each of my users who call my ivr get assigned a public pin random code 3 digits that they could exchange with other callers in a public chat rooms that will allow them to go to a private conference setup just for them. scenario: caller 1 calls , ivr automatically assigns pin? #123 caller 2 calls, ivr automatically assigns pin#124 caller 1 and 2 are talking with callers 3, 4, 5 in conference room 1 but caller 1 and 2 are interested in talking private so caller 1 presses a dtmf option that asks them what the other callers' (caller 2) pin # is? . the system parks him until caller 2 does the same, then the system puts them into a private conference. so what i think i need is : a. method for assigning pins to every caller to ivr b. method for capturing that pin when asked and matching with parked user who has done the same c. method for transferring them into the same 1 on 1 conference.? a demo of how this works on a similar line is 218-895-2080 any help would be greatly appreciated. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111115/b962610e/attachment-0001.html From henrikaagaardsorensen at gmail.com Wed Nov 16 05:50:24 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 16 Nov 2011 03:50:24 +0100 Subject: [Freeswitch-users] Disable e-mail and forwarding in voicemail. Message-ID: I'm going to use FreeSwitch for mobile phones and should therefore disable any e-mail and forwarding features in the voicemail, as this is not supported in GSM etc. So the voicemail should only have the very basic features. Can anyone help me out on how to do this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/3ab0789e/attachment.html From fieldpeak at gmail.com Wed Nov 16 13:10:03 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Wed, 16 Nov 2011 18:10:03 +0800 Subject: [Freeswitch-users] ICE/TURN support on FS Message-ID: Dear Masters, Could anyone can advise how can FS support ICE/TURN, concerning video will cosume lots of bandwitdh, i'm considering use ICE/TURN to force the RTP to bypass FS dynamically, or any alternative solution? Any advise will be much appreciated! -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/e5e3ab3f/attachment.html From ocset at the800group.com Wed Nov 16 14:30:00 2011 From: ocset at the800group.com (ocset) Date: Wed, 16 Nov 2011 19:30:00 +0800 Subject: [Freeswitch-users] Best place to add a new group Message-ID: <4EC39EB8.4020101@the800group.com> Hi I am trying to add a new group and was hoping to refrain from putting it to the "/usr/local/freeswitch/conf/directory/default.xml" file to make it upgrade proof. I have added an xml file as follows but it never gets picked up so either my format is wrong or it is in the wrong place. /usr/local/freeswitch/conf/directory/default/01_test.xml I am running the command "group_call test" but get an error "NO_ROUTE_DESTINATION". Adding the group into the default.xml file works. All help greatly appreciated. Thanks Regards From X.Liu at hw.ac.uk Wed Nov 16 15:36:49 2011 From: X.Liu at hw.ac.uk (Liu, Xingkun) Date: Wed, 16 Nov 2011 12:36:49 -0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper><05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com><174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com><4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk><4EC1236E.3090402@hw.ac.uk><4EC14A92.1040704@hw.ac.uk> Message-ID: <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> Great work, thanks Christopher! I've quick tested it from the dialplan based on the example. It works on my setup. Now I am trying to put it into my Java ESL app, FS complains: switch_ivr_play_say.c:1167 Invalid Args A piece of codes I am using: SendMsg msg = new SendMsg(); msg.addCallCommand("execute"); msg.addExecuteAppName("play_and_detect_speech"); String arg = "say:" + utt + " detect:unimrcp:nuance5-mrcp1-1 {start-input-timers=false,no-input-timeout=5000," + "recognition-timeout=5000}" +this.crntGrammar; msg.addExecuteAppArg(arg); EslMessage response = sendSyncMultiLineCommand(channel, msg.getMsgLines()); Is there something I am missing? Cheers, Xing -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Christopher Rienzo Sent: Tue 11/15/2011 20:05 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to implement TTS barge-in using FS ESL Latest version of FreeSWITCH now has http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_detect_speech Give it a try and let us know if there is anything that can be improved. On Mon, Nov 14, 2011 at 12:06 PM, xl127 wrote: > The way you did and described here is similar to what I wanted. > I want to set start-input-timers=false in the initial setup, then start > the recognition timer once the TTS finishes playing the prompt. > I do not use input callback as I am using ESL event (Java ESL Client > 0.9.2). > > I've not yet figured out how to know the prompt playing has finished, how > to stop the TTS and how to start the input timer. > I didn't get many events when I did barge-in: only got two CHANNEL_EXECUTE > and CAHNNEL_EXECUTE_COMPLETE events. > so didn't get the DETECTED_SPEECH at this point. > > In principle it shouldn't stop FS generating events when barging in, so > there must be somethings wrong in my setup. > > From Examples_directory_lua_asr_tts in the wiki, it mentions the session > needs to sleep for a few seconds before getting the results. > I tried but it didn't help. > > > > > On 14/11/11 14:48, Christopher Rienzo wrote: > > I'm not sure why it locks up for you. It works for me and I know of at > least one other developer that got it all to work. > > The way I do it in my custom APP is to set {start-input-timers=false} in > the speech recognition request and register an input callback function to > deal with the recognition and DTMF events. The input callback reacts to > the start of speech and recognition complete events by stopping the > prompt. At start of speech, the input timers are started and silence is > played until the final result arrives from the recognizer (no match, match, > etc). > > An alternative way to do this is to watch for the recognition events over > ESL and react to them. > > > > On Mon, Nov 14, 2011 at 9:19 AM, xl127 wrote: > >> Hi Christopher, >> >> That will be great! Thanks and looking forward to your app! >> >> By the way, do you have a quick thoughts about the reasons the FS stucks >> when garge-in occurs? >> Otherwise if it is not straightforward to explain it, just leave it >> alone until your new app comes out. >> >> Cheers, >> Xing >> >> >> >> >> On 14/11/11 13:23, Christopher Rienzo wrote: >> >> I'll write up a dialplan app this week to deal with this. Getting tired >> of being asked the same question over and over since it's too complicated >> as is currently designed :) >> >> >> >> >> On Mon, Nov 14, 2011 at 8:07 AM, xl127 wrote: >> >>> Hi, >>> >>> The first problem I need to solve is that FS stucks when I speak during >>> the prompt is playing. >>> It looks like the recognizer is not listening any more. It works fine if >>> I speak after the playing prompt finishes. >>> >>> I tried Nuance and PocketSphinx recognizers and got same problem. >>> >>> Any idea about what the possible causes are? >>> >>> Thanks! >>> Xing >>> >>> >>> On 11/11/11 18:59, xl127 wrote: >>> > Hello, >>> > >>> > I found this is a question that was asked before by others, but I >>> didn't >>> > find the answer. >>> > >>> > Anyway, I am using FS ESL outbound mode connecting to my IVR app, >>> using >>> > FS's "speak" >>> > and "detect_speech" to access Nuance MRCP V1 server. >>> > >>> > I want the user to be able to barge-in during the system's speaking. >>> > How could I implement it? >>> > >>> > I tried to specify "kill-on-barge-in=true" in the mrcp config profile. >>> > The barge-in doesn't work. >>> > >>> > With or without setting kill-on-barge-in, FS stops responding to my >>> > phone call and >>> > eventually it hangs up my call if I speak somthing (do the barge-in) >>> > during the system's speaking. >>> > >>> > I made a turn-by-turn loop in my app, the ASR/TTS works fine if I do >>> not >>> > do barge-in ( I wait until the TTS finishes then I start to speak) >>> > >>> > Any advices please? >>> > >>> > Thanks! >>> > Xing >>> > >>> > >>> > >>> >>> >>> >>> -- >>> Heriot-Watt University is a Scottish charity >>> registered under charity number SC000278. >>> >>> Heriot-Watt University is the Sunday Times >>> Scottish University of the Year 2011-2012 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> ------------------------------ >> >> [image: Scottish University of the Year 2011-12] *Heriot-Watt >> University is the Sunday Times >> >> Scottish University of the Year 2011-2012 >> * >> >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > ------------------------------ > > [image: Scottish University of the Year 2011-12] *Heriot-Watt > University is the Sunday Times > Scottish University of the Year 2011-2012* > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/2ee31abe/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/2ee31abe/attachment-0003.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: hw_uni_of_year.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/2ee31abe/attachment-0004.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: ATT16976631.jpg Type: image/jpeg Size: 4803 bytes Desc: ATT16976631.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/2ee31abe/attachment-0005.jpg From miha at softnet.si Wed Nov 16 15:52:30 2011 From: miha at softnet.si (Miha Zoubek) Date: Wed, 16 Nov 2011 13:52:30 +0100 Subject: [Freeswitch-users] freeswitch radiusclinet (mod_radius_cdr) problems In-Reply-To: <4EC24102.90605@softnet.si> References: <4EBA8B5E.2050900@softnet.si> <4EBCD686.9040305@softnet.si> <4EC0C216.3040806@softnet.si> <4EC22C50.4090400@softnet.si> <4EC24102.90605@softnet.si> Message-ID: <4EC3B20E.2030308@softnet.si> HI, can any one can help me about this issue? Regards, Miha On 11/15/2011 11:37 AM, Miha Zoubek wrote: > Hi Hugh, > > and other question. For incoming calls... > Why there is also AAA request to radius server if I do not have it in my > public/dialplan? > > My freeradius sever is set for AAA (outgoing calls). Why NAS is sending > also accounting request for incoming calls? If the radius is set for > AAA, that I will also need o authenticate incoming calls which is stupid. > All CRD for incoming call we have on SBC. I would like to have only for > outgoing calls. > > Any idea? > > Thank you very much! > > BR, > Miha > > On 11/15/2011 10:09 AM, Miha Zoubek wrote: >> Hi Hugh, >> >> I have put log and my dialplan in pastebin >> (http://pastebin.freeswitch.org/17778). >> Could you please take a look due to my luck of skills with freeswith and >> inform me about your opinion. >> >> >> Regards, >> Miha >> >> On 11/14/2011 10:30 AM, Hugh Irvine wrote: >>> Hello Miha - >>> >>> You are probably getting a RADIUS accounting request for each call leg. >>> >>> The id's in the requests are different, as are the lengths, but you really need to check the contents of the requests to see exactly what they contain. >>> >>> regards >>> >>> Hugh >>> >>> >>> On 14 Nov 2011, at 18:24, Miha Zoubek wrote: >>> >>>> Hi, >>>> >>>> can anyone can help me with this issue? >>>> >>>> >>>> BR, >>>> Miha >>>> >>>> On 11/11/2011 9:02 AM, Miha Zoubek wrote: >>>>> Hi @ Ognjen, >>>>> >>>>> bellow you can see that my Accounting-Request(4), Accounting-Response(5) are twice send. >>>>> >>>>> Any idea? >>>>> >>>>> BR, >>>>> Miha >>>>> >>>>> >>>>> >>>>> 71.449050 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=235, l=265) >>>>> 71.517347 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=235, l=20) >>>>> 73.536126 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Request(1) (id=236, l=210) >>>>> 73.567412 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Accept(2) (id=236, l=20) >>>>> 73.572794 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=237, l=321) >>>>> 73.574156 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=237, l=20) >>>>> 83.482760 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=238, l=401) >>>>> 83.485670 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=238, l=20) >>>>> 83.514594 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=239, l=402) >>>>> 83.516404 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=239, l=20) >>>>> >>>>> On 11/10/2011 6:24 PM, Ognjen Seslija wrote: >>>>>> Radius clients generally send Start and Stop records, I guess this is why you get two records. >>>>>> >>>>>> On Wed, Nov 9, 2011 at 3:17 PM, Miha Zoubek wrote: >>>>>> Hi, >>>>>> >>>>>> radiusclient on freeswitch in sending more that once same response. So I >>>>>> am getting in my sql tables for every call two inputs which is wrong. >>>>>> How can I deal whit this issue? >>>>>> >>>>>> I have paste a log from freeradius server in pastebin that you can see >>>>>> what is freeswitch sending. >>>>>> >>>>>> http://pastebin.freeswitch.org/17730 >>>>>> >>>>>> >>>>>> Thank you! >>>>>> >>>>>> BR, >>>>>> Miha >>>>>> >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 cmrienzo at gmail.com Wed Nov 16 16:10:59 2011 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Wed, 16 Nov 2011 08:10:59 -0500 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> Message-ID: You are missing the tts_engine. Set the tts_engine channel variable to "unimrcp:nuance5-mrcp1-1" or set the default tts profile to nuance5-mrcp1-1 in the unimrcp.conf.xml file and use "say:unimrcp:text to speak" On Wed, Nov 16, 2011 at 7:36 AM, Liu, Xingkun wrote: > ** > > Great work, thanks Christopher! > > I've quick tested it from the dialplan based on the example. It works on > my setup. > > Now I am trying to put it into my Java ESL app, FS complains: > > switch_ivr_play_say.c:1167 Invalid Args > > A piece of codes I am using: > > SendMsg msg = new SendMsg(); > msg.addCallCommand("execute"); > msg.addExecuteAppName("play_and_detect_speech"); > String arg = "say:" + utt + " detect:unimrcp:nuance5-mrcp1-1 > {start-input-timers=false,no-input-timeout=5000," > + "recognition-timeout=5000}" +this.crntGrammar; > msg.addExecuteAppArg(arg); > EslMessage response = sendSyncMultiLineCommand(channel, > msg.getMsgLines()); > > Is there something I am missing? > > Cheers, > > Xing > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org on behalf of > Christopher Rienzo > Sent: Tue 11/15/2011 20:05 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to implement TTS barge-in using FS ESL > > Latest version of FreeSWITCH now has > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_detect_speech > Give it a try and let us know if there is anything that can be improved. > > > > On Mon, Nov 14, 2011 at 12:06 PM, xl127 wrote: > > > The way you did and described here is similar to what I wanted. > > I want to set start-input-timers=false in the initial setup, then start > > the recognition timer once the TTS finishes playing the prompt. > > I do not use input callback as I am using ESL event (Java ESL Client > > 0.9.2). > > > > I've not yet figured out how to know the prompt playing has finished, how > > to stop the TTS and how to start the input timer. > > I didn't get many events when I did barge-in: only got two > CHANNEL_EXECUTE > > and CAHNNEL_EXECUTE_COMPLETE events. > > so didn't get the DETECTED_SPEECH at this point. > > > > In principle it shouldn't stop FS generating events when barging in, so > > there must be somethings wrong in my setup. > > > > From Examples_directory_lua_asr_tts in the wiki, it mentions the session > > needs to sleep for a few seconds before getting the results. > > I tried but it didn't help. > > > > > > > > > > On 14/11/11 14:48, Christopher Rienzo wrote: > > > > I'm not sure why it locks up for you. It works for me and I know of at > > least one other developer that got it all to work. > > > > The way I do it in my custom APP is to set {start-input-timers=false} in > > the speech recognition request and register an input callback function to > > deal with the recognition and DTMF events. The input callback reacts to > > the start of speech and recognition complete events by stopping the > > prompt. At start of speech, the input timers are started and silence is > > played until the final result arrives from the recognizer (no match, > match, > > etc). > > > > An alternative way to do this is to watch for the recognition events over > > ESL and react to them. > > > > > > > > On Mon, Nov 14, 2011 at 9:19 AM, xl127 wrote: > > > >> Hi Christopher, > >> > >> That will be great! Thanks and looking forward to your app! > >> > >> By the way, do you have a quick thoughts about the reasons the FS stucks > >> when garge-in occurs? > >> Otherwise if it is not straightforward to explain it, just leave it > >> alone until your new app comes out. > >> > >> Cheers, > >> Xing > >> > >> > >> > >> > >> On 14/11/11 13:23, Christopher Rienzo wrote: > >> > >> I'll write up a dialplan app this week to deal with this. Getting tired > >> of being asked the same question over and over since it's too > complicated > >> as is currently designed :) > >> > >> > >> > >> > >> On Mon, Nov 14, 2011 at 8:07 AM, xl127 wrote: > >> > >>> Hi, > >>> > >>> The first problem I need to solve is that FS stucks when I speak during > >>> the prompt is playing. > >>> It looks like the recognizer is not listening any more. It works fine > if > >>> I speak after the playing prompt finishes. > >>> > >>> I tried Nuance and PocketSphinx recognizers and got same problem. > >>> > >>> Any idea about what the possible causes are? > >>> > >>> Thanks! > >>> Xing > >>> > >>> > >>> On 11/11/11 18:59, xl127 wrote: > >>> > Hello, > >>> > > >>> > I found this is a question that was asked before by others, but I > >>> didn't > >>> > find the answer. > >>> > > >>> > Anyway, I am using FS ESL outbound mode connecting to my IVR app, > >>> using > >>> > FS's "speak" > >>> > and "detect_speech" to access Nuance MRCP V1 server. > >>> > > >>> > I want the user to be able to barge-in during the system's speaking. > >>> > How could I implement it? > >>> > > >>> > I tried to specify "kill-on-barge-in=true" in the mrcp config > profile. > >>> > The barge-in doesn't work. > >>> > > >>> > With or without setting kill-on-barge-in, FS stops responding to my > >>> > phone call and > >>> > eventually it hangs up my call if I speak somthing (do the barge-in) > >>> > during the system's speaking. > >>> > > >>> > I made a turn-by-turn loop in my app, the ASR/TTS works fine if I do > >>> not > >>> > do barge-in ( I wait until the TTS finishes then I start to speak) > >>> > > >>> > Any advices please? > >>> > > >>> > Thanks! > >>> > Xing > >>> > > >>> > > >>> > > >>> > >>> > >>> > >>> -- > >>> Heriot-Watt University is a Scottish charity > >>> registered under charity number SC000278. > >>> > >>> Heriot-Watt University is the Sunday Times > >>> Scottish University of the Year 2011-2012 > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.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 > >>> > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services:consulting > @freeswitch.orghttp://www.freeswitchsolutions.com > > >> > >> http:// > www.cudatel.com > >> > >> Official FreeSWITCH > Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp:// > www.cluecon.com > >> > >> FreeSWITCH-users mailing listFreeSWITCH-users > @lists.freeswitch.orghttp:// > lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > >> > >> > >> > >> ------------------------------ > >> > >> [image: Scottish University of the Year 2011-12] *Heriot-Watt > > >> University is the Sunday Times > >> > >> Scottish University of the Year 2011-2012 > >> * > >> > >> Heriot-Watt University is a Scottish charity > >> registered under charity number SC000278. > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.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 > >> > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services:consulting > @freeswitch.orghttp://www.freeswitchsolutions.com > > > > > http:// > www.cudatel.com > > > > Official FreeSWITCH > Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp:// > www.cluecon.com > > > > FreeSWITCH-users mailing listFreeSWITCH-users > @lists.freeswitch.orghttp:// > lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > > > > ------------------------------ > > > > [image: Scottish University of the Year 2011-12] *Heriot-Watt > > > University is the Sunday Times > > Scottish University of the Year 2011-2012* > > > > > Heriot-Watt University is a Scottish charity > > registered under charity number SC000278. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > > > > > ------------------------------ > > [image: Scottish University of the Year 2011-12] *Heriot-Watt > University is the Sunday Times > Scottish University of the Year 2011-2012* > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111116/6b8f14e9/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/6b8f14e9/attachment-0001.jpe From acrow at integrafin.co.uk Wed Nov 16 16:16:24 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 16 Nov 2011 13:16:24 +0000 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> Message-ID: <4EC3B7A8.7010306@integrafin.co.uk> On 14/11/11 11:24, Tihomir Culjaga wrote: > > > On Fri, Nov 11, 2011 at 4:32 PM, Steve Underwood > wrote: > > Turning off ECM is equivalent to saying "I use FAX, but I couldn't > give > a damn if they are received". > > > and it always work if the network is stable (you might loose a pixel > here and there...)! If you are running it via open internet, sorry but > try something else (T.38 or switch to e-mail). You cannot expect > pass-through to work reliably in such scenarios. > > T. > Hi all, I built a new machine for this. I know faxing via audio is not the done thing, however this is only to the local Mitel box then out via ISDN until we get the licenses and/or E1 cards for both machine. On the new machine I built a new kernel without tickless and with 1000Hz - and lo and behold ECM faxing now works! Cheers Alex > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/2f06dbd3/attachment.html From fieldpeak at gmail.com Wed Nov 16 17:24:54 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Wed, 16 Nov 2011 22:24:54 +0800 Subject: [Freeswitch-users] execute external system cmd after 200 OK towards REGISTRATION Message-ID: hi friends, i need to excute an external system command once FS replied the 200 OK to the user towards REGISTRATION, could anyone advise where should i add the code, thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/552fa048/attachment.html From miha at softnet.si Wed Nov 16 17:37:01 2011 From: miha at softnet.si (Miha Zoubek) Date: Wed, 16 Nov 2011 15:37:01 +0100 Subject: [Freeswitch-users] freeswitch radiusclinet (mod_radius_cdr) problems In-Reply-To: <4EC3B20E.2030308@softnet.si> References: <4EBA8B5E.2050900@softnet.si> <4EBCD686.9040305@softnet.si> <4EC0C216.3040806@softnet.si> <4EC22C50.4090400@softnet.si> <4EC24102.90605@softnet.si> <4EC3B20E.2030308@softnet.si> Message-ID: <4EC3CA8D.4060406@softnet.si> Hi, I have also try this: http://wiki.freeswitch.org/wiki/Variable_process_cdr But still getting start and stop packet for each leg:( ~ Regards, Miha On 11/16/2011 1:52 PM, Miha Zoubek wrote: > HI, > > can any one can help me about this issue? > > Regards, > Miha > > On 11/15/2011 11:37 AM, Miha Zoubek wrote: >> Hi Hugh, >> >> and other question. For incoming calls... >> Why there is also AAA request to radius server if I do not have it in my >> public/dialplan? >> >> My freeradius sever is set for AAA (outgoing calls). Why NAS is sending >> also accounting request for incoming calls? If the radius is set for >> AAA, that I will also need o authenticate incoming calls which is stupid. >> All CRD for incoming call we have on SBC. I would like to have only for >> outgoing calls. >> >> Any idea? >> >> Thank you very much! >> >> BR, >> Miha >> >> On 11/15/2011 10:09 AM, Miha Zoubek wrote: >>> Hi Hugh, >>> >>> I have put log and my dialplan in pastebin >>> (http://pastebin.freeswitch.org/17778). >>> Could you please take a look due to my luck of skills with freeswith and >>> inform me about your opinion. >>> >>> >>> Regards, >>> Miha >>> >>> On 11/14/2011 10:30 AM, Hugh Irvine wrote: >>>> Hello Miha - >>>> >>>> You are probably getting a RADIUS accounting request for each call leg. >>>> >>>> The id's in the requests are different, as are the lengths, but you really need to check the contents of the requests to see exactly what they contain. >>>> >>>> regards >>>> >>>> Hugh >>>> >>>> >>>> On 14 Nov 2011, at 18:24, Miha Zoubek wrote: >>>> >>>>> Hi, >>>>> >>>>> can anyone can help me with this issue? >>>>> >>>>> >>>>> BR, >>>>> Miha >>>>> >>>>> On 11/11/2011 9:02 AM, Miha Zoubek wrote: >>>>>> Hi @ Ognjen, >>>>>> >>>>>> bellow you can see that my Accounting-Request(4), Accounting-Response(5) are twice send. >>>>>> >>>>>> Any idea? >>>>>> >>>>>> BR, >>>>>> Miha >>>>>> >>>>>> >>>>>> >>>>>> 71.449050 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=235, l=265) >>>>>> 71.517347 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=235, l=20) >>>>>> 73.536126 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Request(1) (id=236, l=210) >>>>>> 73.567412 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Access-Accept(2) (id=236, l=20) >>>>>> 73.572794 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=237, l=321) >>>>>> 73.574156 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=237, l=20) >>>>>> 83.482760 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=238, l=401) >>>>>> 83.485670 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=238, l=20) >>>>>> 83.514594 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Request(4) (id=239, l=402) >>>>>> 83.516404 xxx.xxx.xxx.xxx -> xxx.xxx.xxx.xxx RADIUS Accounting-Response(5) (id=239, l=20) >>>>>> >>>>>> On 11/10/2011 6:24 PM, Ognjen Seslija wrote: >>>>>>> Radius clients generally send Start and Stop records, I guess this is why you get two records. >>>>>>> >>>>>>> On Wed, Nov 9, 2011 at 3:17 PM, Miha Zoubek wrote: >>>>>>> Hi, >>>>>>> >>>>>>> radiusclient on freeswitch in sending more that once same response. So I >>>>>>> am getting in my sql tables for every call two inputs which is wrong. >>>>>>> How can I deal whit this issue? >>>>>>> >>>>>>> I have paste a log from freeradius server in pastebin that you can see >>>>>>> what is freeswitch sending. >>>>>>> >>>>>>> http://pastebin.freeswitch.org/17730 >>>>>>> >>>>>>> >>>>>>> Thank you! >>>>>>> >>>>>>> BR, >>>>>>> Miha >>>>>>> >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.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 >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111116/f52c30d8/attachment-0001.html From jeff at jefflenk.com Wed Nov 16 18:07:03 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 16 Nov 2011 07:07:03 -0800 (PST) Subject: [Freeswitch-users] tts on windows In-Reply-To: <1321395557.94056.YahooMailNeo@web65301.mail.ac2.yahoo.com> References: <1321395557.94056.YahooMailNeo@web65301.mail.ac2.yahoo.com> Message-ID: <1321456023222-7000661.post@n2.nabble.com> At this time the mod_flite.dll is not bundled into the prebuilt msi. This is mostly because the x64 build needs work so this file may be included in the future when this is resolved. If you are interested mod_cepstral is included. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/tts-on-windows-tp6998365p7000661.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mstockton at harqen.com Wed Nov 16 18:20:38 2011 From: mstockton at harqen.com (Matt Stockton) Date: Wed, 16 Nov 2011 09:20:38 -0600 Subject: [Freeswitch-users] rtp-timer-name issue and question Message-ID: Hi all, I am battling the following issue: My applications are primarily lua scripts that play files, accept DTMF , and make db calls and url callbacks to post information. At one point, I was having issues with an ever increasing delay for DTMF recognition as the application progressed (e.g. I would use playAndGetDigits, and the DTMF wouldn't be 'recognized' for a delayed amount of time. This time would increase and it appeared to be dependent on the number of db calls and url callbacks I made in the app -- as I made more, the delay would get worse). I asked around, and ended up changing this setting in my external sip profile: rtp-timer-name = none (it was soft before). This fixed my issue. Now, I am battling a separate issue, and I seem to have isolated it's cause to setting rtp-timer-name to none instead of soft. The problem now is that when I want to play an mp3 file, and I use streamFile with shout or playback with http_get, the beginning of the file is not played at all (playing starts probably about 200-400ms into the file. This is very repeatable for me - it's not happening to every streamed file, but it seems to be very deterministic on while line of lua script it's happening (the streamFile is happening in a loop, and it seems to always be happening to the first streamed file). If I change rtp-timer-name to soft, everything is fine, but then I'll have the first problem I described above. So, to fix one problem I need rtp-timer-name = none, for the other problem, I need rtp-timer-name = soft. I must be doing something wrong here, any suggestions? Thanks, Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/168e694f/attachment.html From jeff at jefflenk.com Wed Nov 16 18:21:12 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 16 Nov 2011 07:21:12 -0800 (PST) Subject: [Freeswitch-users] update freeswitch on windows In-Reply-To: References: <1321395303.55295.YahooMailNeo@web65314.mail.ac2.yahoo.com> Message-ID: <1321456872539-7000717.post@n2.nabble.com> As mercutioviz said its probably best that you build the code yourself (you are in control). With that said the weekly msi's could be used to do the update but you will need to keep your configuration files backed up as the msi may overwrite them(cant remember). You could also do copy deployment of the weekly install to other machines or perhaps to another file directory on the same machine(just dont copy the \conf folder). -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/update-freeswitch-on-windows-tp6998366p7000717.html Sent from the freeswitch-users mailing list archive at Nabble.com. From x.liu at hw.ac.uk Wed Nov 16 18:24:29 2011 From: x.liu at hw.ac.uk (xl127) Date: Wed, 16 Nov 2011 15:24:29 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> Message-ID: <4EC3D5AD.5070507@hw.ac.uk> Hi Christopher, Now it works in my ESL app though I am just able to do one dialogue ( I need to add the event catching for furthur dialgoues). I have a couple of questions here: 1. In the first try, my Nuance server was able to be accessed somehow (FS says the MRCP is not responding in 5000ms, something like that), then FS says: [WARNING] rtsp_client.c:386 () Failed to Connect to RTSP Server 99.185.85.31:554, later FS says: [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER channel error! [ERR] switch_ivr_play_say.c:2439 Invalid TTS module! The SYNTHESIZER channel error and Invalid TTS module error are obvious. What I don't understand is why it went to this stange address: 99.185.85.31:554? 2. I specified TTS engine in play_and_detect_speech as "say:unimrcp:nuance5-mrcp1-1: the text to speak" It works though I didn't specify the TTS voice. How do I specify the TTS voice? In the mrcp profile (how?)? or something like: "say:unimrcp:nuance5-mrcp1-1:Serena: the text to speak" (this seems not right.) 3. The barge-in works well, thanks!. Is the barge-in configurable? In some scenarios, we might not allow barge-in. 4. How could I get the text which has spoken to the user when barge-in occurs? Or Could I get the time when barge-in occurs? If I know the barge-in time and rough totale time for the whole text to be spoken I can figure out the spoken text by manually checking the recorded audio file later, which would be painful. 5. when I use "speak" and "detect_speech" apps in ESL, I can catch event: DETECTED_SPEECH and speech-type: begin-speaking and "detected-speech", then I do the recognition results processing. The new app play_and_detect_speech seems not generate these events any more. The way that I can think of to get the results is to catch event:CHANNEL_EXECUTE_COMPLETE then check if variable_current_application=play_and_detect_speech, then get the results from variable_detect_speech_result. Is this the proper way to get the results in ESL app? Or will play_and_detect_speech later on be consistent with detect_speech in term of ASR events? 6. I'd like to set start-input-timers=false in the initial request then start the recognition timers (start-input-timers=true) after the TTS finishes. How possibly could I do this? Apologies for so many questions! Cheers, Xing On 16/11/11 13:10, Christopher Rienzo wrote: > You are missing the tts_engine. Set the tts_engine channel variable > to "unimrcp:nuance5-mrcp1-1" or set the default tts profile to > nuance5-mrcp1-1 in the unimrcp.conf.xml file and use "say:unimrcp:text > to speak" > > > On Wed, Nov 16, 2011 at 7:36 AM, Liu, Xingkun > wrote: > > Great work, thanks Christopher! > > I've quick tested it from the dialplan based on the example. It > works on my setup. > > Now I am trying to put it into my Java ESL app, FS complains: > > switch_ivr_play_say.c:1167 Invalid Args > > A piece of codes I am using: > > SendMsg msg = new SendMsg(); > msg.addCallCommand("execute"); > msg.addExecuteAppName("play_and_detect_speech"); > String arg = "say:" + utt + " > detect:unimrcp:nuance5-mrcp1-1 > {start-input-timers=false,no-input-timeout=5000," > + "recognition-timeout=5000}" +this.crntGrammar; > msg.addExecuteAppArg(arg); > EslMessage response = sendSyncMultiLineCommand(channel, > msg.getMsgLines()); > > Is there something I am missing? > > Cheers, > > Xing > > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > on behalf > of Christopher Rienzo > Sent: Tue 11/15/2011 20:05 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to implement TTS barge-in > using FS ESL > > Latest version of FreeSWITCH now has > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_detect_speech > Give it a try and let us know if there is anything that can be > improved. > > > > On Mon, Nov 14, 2011 at 12:06 PM, xl127 > wrote: > > > The way you did and described here is similar to what I wanted. > > I want to set start-input-timers=false in the initial setup, > then start > > the recognition timer once the TTS finishes playing the prompt. > > I do not use input callback as I am using ESL event (Java ESL Client > > 0.9.2). > > > > I've not yet figured out how to know the prompt playing has > finished, how > > to stop the TTS and how to start the input timer. > > I didn't get many events when I did barge-in: only got two > CHANNEL_EXECUTE > > and CAHNNEL_EXECUTE_COMPLETE events. > > so didn't get the DETECTED_SPEECH at this point. > > > > In principle it shouldn't stop FS generating events when barging > in, so > > there must be somethings wrong in my setup. > > > > From Examples_directory_lua_asr_tts in the wiki, it mentions the > session > > needs to sleep for a few seconds before getting the results. > > I tried but it didn't help. > > > > > > > > > > On 14/11/11 14:48, Christopher Rienzo wrote: > > > > I'm not sure why it locks up for you. It works for me and I > know of at > > least one other developer that got it all to work. > > > > The way I do it in my custom APP is to set > {start-input-timers=false} in > > the speech recognition request and register an input callback > function to > > deal with the recognition and DTMF events. The input callback > reacts to > > the start of speech and recognition complete events by stopping the > > prompt. At start of speech, the input timers are started and > silence is > > played until the final result arrives from the recognizer (no > match, match, > > etc). > > > > An alternative way to do this is to watch for the recognition > events over > > ESL and react to them. > > > > > > > > On Mon, Nov 14, 2011 at 9:19 AM, xl127 > wrote: > > > >> Hi Christopher, > >> > >> That will be great! Thanks and looking forward to your app! > >> > >> By the way, do you have a quick thoughts about the reasons the > FS stucks > >> when garge-in occurs? > >> Otherwise if it is not straightforward to explain it, just > leave it > >> alone until your new app comes out. > >> > >> Cheers, > >> Xing > >> > >> > >> > >> > >> On 14/11/11 13:23, Christopher Rienzo wrote: > >> > >> I'll write up a dialplan app this week to deal with this. > Getting tired > >> of being asked the same question over and over since it's too > complicated > >> as is currently designed :) > >> > >> > >> > >> > >> On Mon, Nov 14, 2011 at 8:07 AM, xl127 > wrote: > >> > >>> Hi, > >>> > >>> The first problem I need to solve is that FS stucks when I > speak during > >>> the prompt is playing. > >>> It looks like the recognizer is not listening any more. It > works fine if > >>> I speak after the playing prompt finishes. > >>> > >>> I tried Nuance and PocketSphinx recognizers and got same problem. > >>> > >>> Any idea about what the possible causes are? > >>> > >>> Thanks! > >>> Xing > >>> > >>> > >>> On 11/11/11 18:59, xl127 wrote: > >>> > Hello, > >>> > > >>> > I found this is a question that was asked before by others, > but I > >>> didn't > >>> > find the answer. > >>> > > >>> > Anyway, I am using FS ESL outbound mode connecting to my IVR > app, > >>> using > >>> > FS's "speak" > >>> > and "detect_speech" to access Nuance MRCP V1 server. > >>> > > >>> > I want the user to be able to barge-in during the system's > speaking. > >>> > How could I implement it? > >>> > > >>> > I tried to specify "kill-on-barge-in=true" in the mrcp > config profile. > >>> > The barge-in doesn't work. > >>> > > >>> > With or without setting kill-on-barge-in, FS stops > responding to my > >>> > phone call and > >>> > eventually it hangs up my call if I speak somthing (do the > barge-in) > >>> > during the system's speaking. > >>> > > >>> > I made a turn-by-turn loop in my app, the ASR/TTS works fine > if I do > >>> not > >>> > do barge-in ( I wait until the TTS finishes then I start to > speak) > >>> > > >>> > Any advices please? > >>> > > >>> > Thanks! > >>> > Xing > >>> > > >>> > > >>> > > >>> > >>> > >>> > >>> -- > >>> Heriot-Watt University is a Scottish charity > >>> registered under charity number SC000278. > >>> > >>> Heriot-Watt University is the Sunday Times > >>> Scottish University of the Year 2011-2012 > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.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 > >>> > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting > Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication > Server <> > >> > >> Official FreeSWITCH > Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > >> > >> FreeSWITCH-users mailing > listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > >> > >> > >> > >> ------------------------------ > >> > >> [image: Scottish University of the Year 2011-12] *Heriot-Watt > > >> University is the Sunday Times > >> > >> Scottish University of the Year 2011-2012 > >> * > >> > >> Heriot-Watt University is a Scottish charity > >> registered under charity number SC000278. > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.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 > >> > >> > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting > Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server <> > > > > Official FreeSWITCH > Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > > > > FreeSWITCH-users mailing > listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > > > > ------------------------------ > > > > [image: Scottish University of the Year 2011-12] *Heriot-Watt > > > University is the Sunday Times > > Scottish University of the Year 2011-2012* > > > > > Heriot-Watt University is a Scottish charity > > registered under charity number SC000278. > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > > > > > ------------------------------------------------------------------------ > > Scottish University of the Year 2011-12 *Heriot-Watt University is > the Sunday Times > Scottish University of the Year 2011-2012* > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/10ac67e8/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/10ac67e8/attachment-0001.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/10ac67e8/attachment-0001.jpg From adam.kelloway at newpace.ca Wed Nov 16 18:39:58 2011 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Wed, 16 Nov 2011 11:39:58 -0400 Subject: [Freeswitch-users] sendevent issues In-Reply-To: References: <4EC284AB.7070509@newpace.ca> Message-ID: <4EC3D94E.5070900@newpace.ca> Thanks Michael, would you mind trying the following? I've discovered that I never see the event when I add a "Unique-ID" header: does not work: sendevent CUSTOM Event-Name: CUSTOM Event-Subclass: my::event Unique-ID: ec13a91a-1067-11e1-a184-efefb30e1896 works: sendevent CUSTOM Event-Name: CUSTOM Event-Subclass: my::event works: sendevent CUSTOM Event-Name: CUSTOM Event-Subclass: my::event my-header: my-value I could probably add the uuid to a different header name, but I'm trying to keep the names I use consistent with other events. Thanks, Adam On 3:59 PM, Michael Collins wrote: > I just tested this and it worked for me. However, instead of doing > "/events all" i did "/events plain all". I see all the events on the > system, including the ones injected. > > -MC > > On Tue, Nov 15, 2011 at 7:26 AM, Adam Kelloway > > wrote: > > Hi there, > > I have client connection to a freeswitch instance, and send the > following message to freeswitch from it: > > sendevent CUSTOM\n > Event-Name: CUSTOM\n > Event-Subclass: my::event\n > my-header: my-value\n > \n > > I receive the following reply: > > Content-Type: command/reply > Reply-Text: +OK > > I subscribe to all events from fs_cli, using: > > /events all > > However, I never see the event that was sent. Am I misinterpreting how > sendevent is used? I would expect that the console application > would see > this event. Am I perhaps missing some mandatory headers? > > Thanks, > > Adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- Adam -- NewPace Logo Adam Kelloway Software Engineer, NewPace phone +1 (902) 406--8375 x1031 email Adam.Kelloway at NewPace.com aim /msn Adam.Kelloway @NewPace.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/aedc0028/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Newpace_50x50.png Type: image/png Size: 4620 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/aedc0028/attachment.png From curriegrad2004 at gmail.com Wed Nov 16 18:39:07 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 16 Nov 2011 07:39:07 -0800 Subject: [Freeswitch-users] update freeswitch on windows In-Reply-To: <1321456872539-7000717.post@n2.nabble.com> References: <1321395303.55295.YahooMailNeo@web65314.mail.ac2.yahoo.com> <1321456872539-7000717.post@n2.nabble.com> Message-ID: One way to build your own FreeSWITCH builds is to get your own copy of VC 2010 express. It's free from Microsoft's website. The wiki has details on how to do this. On Wed, Nov 16, 2011 at 7:21 AM, Jeff Lenk wrote: > As mercutioviz said its probably best that you build the code yourself (you > are in control). With that said the weekly msi's could be used to do the > update but you will need to keep your configuration files backed up as the > msi may overwrite them(cant remember). You could also do copy deployment of > the weekly install to other machines or perhaps to another file directory on > the same machine(just dont copy the \conf folder). > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/update-freeswitch-on-windows-tp6998366p7000717.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 cmrienzo at gmail.com Wed Nov 16 18:52:23 2011 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Wed, 16 Nov 2011 10:52:23 -0500 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <4EC3D5AD.5070507@hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> <4EC3D5AD.5070507@hw.ac.uk> Message-ID: Responses inline Now it works in my ESL app though I am just able to do one dialogue ( I > need to add the event catching for furthur dialgoues). > > I have a couple of questions here: > > 1. In the first try, my Nuance server was able to be accessed somehow > (FS says the MRCP is not responding in 5000ms, > something like that), then FS says: [WARNING] rtsp_client.c:386 () > Failed to Connect to RTSP Server 99.185.85.31:554, > later FS says: > [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER channel error! > [ERR] switch_ivr_play_say.c:2439 Invalid TTS module! > > The SYNTHESIZER channel error and Invalid TTS module error are obvious. > > What I don't understand is why it went to this stange address: > 99.185.85.31:554? > check your unimrcp configuration. Make sure the default TTS and ASR profiles are set to actual servers. > 2. I specified TTS engine in play_and_detect_speech as > "say:unimrcp:nuance5-mrcp1-1: the text to speak" > It works though I didn't specify the TTS voice. > > How do I specify the TTS voice? In the mrcp profile (how?)? or > something like: > "say:unimrcp:nuance5-mrcp1-1:Serena: the text to speak" (this > seems not right.) > That won't work. Set the tts_engine variable as I explained previously, or use say:unimrcp:voice:text to speak with the desired voice and the correct default TTS profile defined in unimrcp.conf.xml. This is a limitation of the say: notation. Alternatively, the voice can be defined with the tts_voice channel variable. > 3. The barge-in works well, thanks!. Is the barge-in configurable? In > some scenarios, we might not allow barge-in. > If you don't want to barge in, just do "playback (or speak)" first, then "play_and_detect_speech" with a silence prompt. > > 4. How could I get the text which has spoken to the user when barge-in > occurs? > Or Could I get the time when barge-in occurs? If I know the barge-in > time and rough totale time for the whole text > to be spoken I can figure out the spoken text by manually checking > the recorded audio file later, which would be painful. > If this is necessary, you might want to use the lower-level functions instead to watch for the begin-speaking event. > > 5. when I use "speak" and "detect_speech" apps in ESL, I can catch > event: DETECTED_SPEECH and speech-type: begin-speaking > and "detected-speech", then I do the recognition results processing. > > The new app play_and_detect_speech seems not generate these events any > more. The way that I can think of to get the results > is to catch event:CHANNEL_EXECUTE_COMPLETE then check if > variable_current_application=play_and_detect_speech, then get > the results from variable_detect_speech_result. > > Is this the proper way to get the results in ESL app? Or will > play_and_detect_speech later on be consistent with detect_speech > in term of ASR events? > play_and_detect_speech is a higher level abstraction to simplify things. If you want to have more control, go back to using the ESL events. Reading the code in mod_dptools and switch_ivr_async will give you hints about how to do it correctly. > > 6. I'd like to set start-input-timers=false in the initial request then > start the recognition timers (start-input-timers=true) > after the TTS finishes. > How possibly could I do this? > This is automatically done in the switch_ivr_play_and_detect_speech() function. You just need to specify start-input-timers=false in the beginning. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/0bd5d893/attachment-0001.html From msc at freeswitch.org Wed Nov 16 19:38:54 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Nov 2011 08:38:54 -0800 Subject: [Freeswitch-users] FreeSWITCH Conf Call Today Message-ID: Hello all, We have a really light agenda for the conf call today: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_16 Today would be a good day to come talk about your FreeSWITCH questions! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/4086773a/attachment.html From B.Tietz at pinguin.ag Wed Nov 16 13:00:44 2011 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Wed, 16 Nov 2011 11:00:44 +0100 Subject: [Freeswitch-users] sofia recover can not recover calls Message-ID: <07BF4904977CC645B485E970424193AD0E68BD11F3@localhost> Hi, I'm trying to make a 'sofia recover' with the 'fssofia' script from wiki. If the sofia recover is fired I get the following messages and the calls are not recovered: 2011-11-16 10:49:29.211897 [NOTICE] switch_core_session.c:1754 Close Channel N/A [CS_NEW] 2011-11-16 10:49:29.211897 [DEBUG] switch_core_state_machine.c:494 () Running State Change CS_DESTROY 2011-11-16 10:49:29.211897 [DEBUG] switch_core_state_machine.c:504 (N/A) State DESTROY 2011-11-16 10:49:29.211897 [DEBUG] mod_sofia.c:370 N/A SOFIA DESTROY 2011-11-16 10:49:29.211897 [DEBUG] switch_core_state_machine.c:504 (N/A) State DESTROY going to sleep 2011-11-16 10:49:29.211897 [WARNING] sofia_glue.c:5400 Invalid cdr data, call not recovered 2011-11-16 10:49:29.211897 [NOTICE] switch_core_session.c:1754 Close Channel N/A [CS_NEW] 2011-11-16 10:49:29.211897 [DEBUG] switch_core_state_machine.c:494 () Running State Change CS_DESTROY 2011-11-16 10:49:29.211897 [DEBUG] switch_core_state_machine.c:504 (N/A) State DESTROY 2011-11-16 10:49:29.211897 [DEBUG] mod_sofia.c:370 N/A SOFIA DESTROY 2011-11-16 10:49:29.211897 [DEBUG] switch_core_state_machine.c:504 (N/A) State DESTROY going to sleep 2011-11-16 10:49:29.211897 [WARNING] sofia_glue.c:5400 Invalid cdr data, call not recovered Any ideas?! regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/d9ec4c49/attachment.html From B.Tietz at pinguin.ag Wed Nov 16 13:02:44 2011 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Wed, 16 Nov 2011 11:02:44 +0100 Subject: [Freeswitch-users] sofia recover can not recover calls Message-ID: <07BF4904977CC645B485E970424193AD0E68BD11F9@localhost> Hi, forgot some details about the systems. Here we go: FreeSWITCH Version 1.0.head (git-d93ed90 2011-11-11 20-17-21 -0600) And both machines have the same name in switch.con.xml regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/d3b6fc56/attachment.html From jbaclor at ezuce.com Wed Nov 16 07:33:36 2011 From: jbaclor at ezuce.com (Joegen Baclor) Date: Wed, 16 Nov 2011 12:33:36 +0800 Subject: [Freeswitch-users] Comfort noise or silence rtp packets during recording Message-ID: <4EC33D20.3060604@ezuce.com> I tried asking in the freeswitch irc channel but got no response. I'm trying my luck here. I have recently received a packet capture from the sipx community that seems to indicate that when FreeSwitch is recording a voice mail message via our custom IVR, that it does not send either comfort noise or silence packets back to the caller. This results to the call getting dropped by the ITSP after 30 seconds because of RTP time out. Is there a configurable parameter to make the IVR send some voice activity back? Joegen From mpicher at gmail.com Wed Nov 16 18:43:03 2011 From: mpicher at gmail.com (Michael Picher) Date: Wed, 16 Nov 2011 10:43:03 -0500 Subject: [Freeswitch-users] Timing issues in AWS? Message-ID: Hi guys, Trying to get to the bottom of some conference bridge issues I'm having with running the system in AWS. We're hearing a bunch of snap-crackle-pops in conference bridges and when I tcpdum on the server itself I see them in the RTP and see RTP timestamp problems. I've run the following: timer_test freeswitch at 127.0.0.1@internal> timer_test 120 10 Avg: 120.004ms Total Time: 1200.315ms 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120006 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 120040 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119976 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120007 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 120001 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 120031 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119996 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 119994 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 120005 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119991 test_time freeswitch at 127.0.0.1@internal> time_test 600 10 test 1 sleep 600 1592 test 2 sleep 600 986 test 3 sleep 600 1018 test 4 sleep 600 980 test 5 sleep 600 1005 test 6 sleep 600 1000 test 7 sleep 600 972 test 8 sleep 600 990 test 9 sleep 600 1006 test 10 sleep 600 994 avg 1054 For kernel: [root at openuc bin]# uname -r 2.6.21.7-2.fc8xen CONFIG_HZ: [root at openuc bin]# grep CONFIG_HZ /boot/config-* /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 are the xenU kernel settings screwing me here? Thanks, Mike -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/66b0eb04/attachment.html From krice at freeswitch.org Wed Nov 16 19:47:08 2011 From: krice at freeswitch.org (Ken Rice) Date: Wed, 16 Nov 2011 10:47:08 -0600 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: Message-ID: Actually, consider this, you have 1 server running 10 different instances for 10 different people and as AWS you have to try and be level handed about how you hand out instances on a single node in your cloud cluster... How do you make sure that instance 1 isnt hogging all the CPU time that instances 9 ? 10 need? This is the nature of a timeshare system... This goes back to AS400s SYS 34?s and 36?s... Don?t get me wrong I think AWS, and cloud computing in general can be great for certain things (store and forward), but do you really want to chance the interruptions in RealTime media streaming where any delay in processing causes missed or dropped packets... On 11/16/11 9:43 AM, "Michael Picher" wrote: > Hi guys, > > Trying to get to the bottom of some conference bridge issues I'm having with > running the system in AWS. > > We're hearing a bunch of snap-crackle-pops in conference bridges and when I > tcpdum on the server itself I see them in the RTP and see RTP timestamp > problems. > > I've run the following: > > > timer_test > freeswitch at 127.0.0.1@internal> timer_test 120 10 > > Avg: 120.004ms Total Time: 1200.315ms > > 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: > samplecount after init: 1 > freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] > mod_commands.c:466 Timer Test: samplecount after first step: 2 > 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep > 120 120006 > 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep > 120 120040 > 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep > 120 119976 > 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep > 120 120007 > 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep > 120 120001 > 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep > 120 120031 > 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep > 120 119996 > 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep > 120 119994 > 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep > 120 120005 > 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep > 120 119991 > > test_time > freeswitch at 127.0.0.1@internal> time_test 600 10 > > test 1 sleep 600 1592 > test 2 sleep 600 986 > test 3 sleep 600 1018 > test 4 sleep 600 980 > test 5 sleep 600 1005 > test 6 sleep 600 1000 > test 7 sleep 600 972 > test 8 sleep 600 990 > test 9 sleep 600 1006 > test 10 sleep 600 994 > avg 1054 > > For kernel: > [root at openuc bin]# uname -r > > 2.6.21.7-2.fc8xen > > CONFIG_HZ: > [root at openuc bin]# grep CONFIG_HZ /boot/config-* > > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 > > are the xenU kernel settings screwing me here? > > Thanks, > ? Mike > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/c30c45b9/attachment-0001.html From x.liu at hw.ac.uk Wed Nov 16 19:51:16 2011 From: x.liu at hw.ac.uk (xl127) Date: Wed, 16 Nov 2011 16:51:16 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> <4EC3D5AD.5070507@hw.ac.uk> Message-ID: <4EC3EA04.7030605@hw.ac.uk> Hi Christopher, The questions are cleared to me now. Many thanks for your explanations! Best regards, Xing On 16/11/11 15:52, Christopher Rienzo wrote: > > Responses inline > > > Now it works in my ESL app though I am just able to do one > dialogue ( I need to add the event catching for furthur dialgoues). > > I have a couple of questions here: > > 1. In the first try, my Nuance server was able to be accessed > somehow (FS says the MRCP is not responding in 5000ms, > something like that), then FS says: [WARNING] > rtsp_client.c:386 () Failed to Connect to RTSP Server > 99.185.85.31:554 , > later FS says: > [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER channel error! > [ERR] switch_ivr_play_say.c:2439 Invalid TTS module! > > The SYNTHESIZER channel error and Invalid TTS module error are > obvious. > > What I don't understand is why it went to this stange address: > 99.185.85.31:554 ? > > > check your unimrcp configuration. Make sure the default TTS and ASR > profiles are set to actual servers. > > 2. I specified TTS engine in play_and_detect_speech as > "say:unimrcp:nuance5-mrcp1-1: the text to speak" > It works though I didn't specify the TTS voice. > > How do I specify the TTS voice? In the mrcp profile (how?)? > or something like: > "say:unimrcp:nuance5-mrcp1-1:Serena: the text to speak" > (this seems not right.) > > > That won't work. Set the tts_engine variable as I explained > previously, or use say:unimrcp:voice:text to speak with the desired > voice and the correct default TTS profile defined in > unimrcp.conf.xml. This is a limitation of the say: notation. > Alternatively, the voice can be defined with the tts_voice channel > variable. > > 3. The barge-in works well, thanks!. Is the barge-in > configurable? In some scenarios, we might not allow barge-in. > > > If you don't want to barge in, just do "playback (or speak)" first, > then "play_and_detect_speech" with a silence prompt. > > > 4. How could I get the text which has spoken to the user when > barge-in occurs? > Or Could I get the time when barge-in occurs? If I know the > barge-in time and rough totale time for the whole text > to be spoken I can figure out the spoken text by manually > checking the recorded audio file later, which would be painful. > > > If this is necessary, you might want to use the lower-level functions > instead to watch for the begin-speaking event. > > > 5. when I use "speak" and "detect_speech" apps in ESL, I can > catch event: DETECTED_SPEECH and speech-type: begin-speaking > and "detected-speech", then I do the recognition results > processing. > > The new app play_and_detect_speech seems not generate these > events any more. The way that I can think of to get the results > is to catch event:CHANNEL_EXECUTE_COMPLETE then check if > variable_current_application=play_and_detect_speech, then get > the results from variable_detect_speech_result. > > Is this the proper way to get the results in ESL app? Or will > play_and_detect_speech later on be consistent with detect_speech > in term of ASR events? > > > play_and_detect_speech is a higher level abstraction to simplify > things. If you want to have more control, go back to using the ESL > events. Reading the code in mod_dptools and switch_ivr_async will > give you hints about how to do it correctly. > > > 6. I'd like to set start-input-timers=false in the initial > request then start the recognition timers (start-input-timers=true) > after the TTS finishes. > How possibly could I do this? > > > This is automatically done in the switch_ivr_play_and_detect_speech() > function. You just need to specify start-input-timers=false in the > beginning. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/c16f3769/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/c16f3769/attachment.jpg From chris.chen2004 at gmail.com Wed Nov 16 19:52:35 2011 From: chris.chen2004 at gmail.com (Chris Chen) Date: Wed, 16 Nov 2011 11:52:35 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: Message-ID: Just a simple question, what kind of AWS instance are you running your FreeSWITCH? It makes huge difference. Thanks, Chris On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher wrote: > Hi guys, > > Trying to get to the bottom of some conference bridge issues I'm having > with running the system in AWS. > > We're hearing a bunch of snap-crackle-pops in conference bridges and when > I tcpdum on the server itself I see them in the RTP and see RTP timestamp > problems. > > I've run the following: > > > timer_test > freeswitch at 127.0.0.1@internal> timer_test 120 10 > > Avg: 120.004ms Total Time: 1200.315ms > > 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: > samplecount after init: 1 > freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] > mod_commands.c:466 Timer Test: samplecount after first step: 2 > 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 > sleep 120 120006 > 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 > sleep 120 120040 > 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 > sleep 120 119976 > 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 > sleep 120 120007 > 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 > sleep 120 120001 > 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 > sleep 120 120031 > 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 > sleep 120 119996 > 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 > sleep 120 119994 > 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 > sleep 120 120005 > 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 > sleep 120 119991 > > test_time > freeswitch at 127.0.0.1@internal> time_test 600 10 > > test 1 sleep 600 1592 > test 2 sleep 600 986 > test 3 sleep 600 1018 > test 4 sleep 600 980 > test 5 sleep 600 1005 > test 6 sleep 600 1000 > test 7 sleep 600 972 > test 8 sleep 600 990 > test 9 sleep 600 1006 > test 10 sleep 600 994 > avg 1054 > > For kernel: > [root at openuc bin]# uname -r > > 2.6.21.7-2.fc8xen > > CONFIG_HZ: > [root at openuc bin]# grep CONFIG_HZ /boot/config-* > > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 > > are the xenU kernel settings screwing me here? > > Thanks, > Mike > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111116/41b23736/attachment-0001.html From mpicher at gmail.com Wed Nov 16 19:56:03 2011 From: mpicher at gmail.com (Michael Picher) Date: Wed, 16 Nov 2011 11:56:03 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: Message-ID: I fully understand the nature of a timeshare system... they go way back before AS400's... I used them on DEC PDP11's. Indeed you can't make sure that one instance isn't 'hogging' CPU unless there's an even distribution of time / cpu unit. That being said, it has been implied that FS should work on a large AWS instance... I'm trying to get to the bottom of how many users and how large a system questions that we have. I just have basic timing problems at this point with very small conference bridges on a m1.large AWS instance. Granted, I don't know what the rest of the server is doing but these are pretty consistent problems. Mike On Wed, Nov 16, 2011 at 11:47 AM, Ken Rice wrote: > Actually, consider this, you have 1 server running 10 different > instances for 10 different people and as AWS you have to try and be level > handed about how you hand out instances on a single node in your cloud > cluster... > > How do you make sure that instance 1 isnt hogging all the CPU time that > instances 9 ? 10 need? > > This is the nature of a timeshare system... This goes back to AS400s SYS > 34?s and 36?s... Don?t get me wrong I think AWS, and cloud computing in > general can be great for certain things (store and forward), but do you > really want to chance the interruptions in RealTime media streaming where > any delay in processing causes missed or dropped packets... > > > > On 11/16/11 9:43 AM, "Michael Picher" wrote: > > Hi guys, > > Trying to get to the bottom of some conference bridge issues I'm having > with running the system in AWS. > > We're hearing a bunch of snap-crackle-pops in conference bridges and when > I tcpdum on the server itself I see them in the RTP and see RTP timestamp > problems. > > I've run the following: > > > timer_test > freeswitch at 127.0.0.1@internal> timer_test 120 10 > > Avg: 120.004ms Total Time: 1200.315ms > > 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: > samplecount after init: 1 > freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] > mod_commands.c:466 Timer Test: samplecount after first step: 2 > 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 > sleep 120 120006 > 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 > sleep 120 120040 > 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 > sleep 120 119976 > 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 > sleep 120 120007 > 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 > sleep 120 120001 > 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 > sleep 120 120031 > 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 > sleep 120 119996 > 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 > sleep 120 119994 > 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 > sleep 120 120005 > 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 > sleep 120 119991 > > test_time > freeswitch at 127.0.0.1@internal> time_test 600 10 > > test 1 sleep 600 1592 > test 2 sleep 600 986 > test 3 sleep 600 1018 > test 4 sleep 600 980 > test 5 sleep 600 1005 > test 6 sleep 600 1000 > test 7 sleep 600 972 > test 8 sleep 600 990 > test 9 sleep 600 1006 > test 10 sleep 600 994 > avg 1054 > > For kernel: > [root at openuc bin]# uname -r > > 2.6.21.7-2.fc8xen > > CONFIG_HZ: > [root at openuc bin]# grep CONFIG_HZ /boot/config-* > > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 > > are the xenU kernel settings screwing me here? > > Thanks, > Mike > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/de1465f7/attachment.html From mpicher at gmail.com Wed Nov 16 19:56:44 2011 From: mpicher at gmail.com (Michael Picher) Date: Wed, 16 Nov 2011 11:56:44 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: Message-ID: m1.large I have a c1.xlarge queued up and ready to test... On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen wrote: > Just a simple question, what kind of AWS instance are you running your > FreeSWITCH? > It makes huge difference. > Thanks, > Chris > > On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher wrote: > >> Hi guys, >> >> Trying to get to the bottom of some conference bridge issues I'm having >> with running the system in AWS. >> >> We're hearing a bunch of snap-crackle-pops in conference bridges and when >> I tcpdum on the server itself I see them in the RTP and see RTP timestamp >> problems. >> >> I've run the following: >> >> >> timer_test >> freeswitch at 127.0.0.1@internal> timer_test 120 10 >> >> Avg: 120.004ms Total Time: 1200.315ms >> >> 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: >> samplecount after init: 1 >> freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] >> mod_commands.c:466 Timer Test: samplecount after first step: 2 >> 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 >> sleep 120 120006 >> 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 >> sleep 120 120040 >> 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 >> sleep 120 119976 >> 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 >> sleep 120 120007 >> 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 >> sleep 120 120001 >> 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 >> sleep 120 120031 >> 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 >> sleep 120 119996 >> 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 >> sleep 120 119994 >> 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 >> sleep 120 120005 >> 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 >> sleep 120 119991 >> >> test_time >> freeswitch at 127.0.0.1@internal> time_test 600 10 >> >> test 1 sleep 600 1592 >> test 2 sleep 600 986 >> test 3 sleep 600 1018 >> test 4 sleep 600 980 >> test 5 sleep 600 1005 >> test 6 sleep 600 1000 >> test 7 sleep 600 972 >> test 8 sleep 600 990 >> test 9 sleep 600 1006 >> test 10 sleep 600 994 >> avg 1054 >> >> For kernel: >> [root at openuc bin]# uname -r >> >> 2.6.21.7-2.fc8xen >> >> CONFIG_HZ: >> [root at openuc bin]# grep CONFIG_HZ /boot/config-* >> >> /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y >> /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set >> /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set >> /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 >> /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y >> /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 >> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set >> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set >> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set >> /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y >> /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 >> >> are the xenU kernel settings screwing me here? >> >> Thanks, >> Mike >> >> >> -- >> There are 10 kinds of people in this world, those who understand binary >> and those who don't. >> >> mpicher at gmail.com >> blog: http://www.sipxecs.info >> call: sip:mpicher at sipxecs.info >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/6dca267c/attachment.html From SPapineni at enghouse.com Wed Nov 16 20:20:28 2011 From: SPapineni at enghouse.com (Papineni, Suneel) Date: Wed, 16 Nov 2011 17:20:28 +0000 Subject: [Freeswitch-users] Issue with mod_diangling during compilation Message-ID: <9438D04074E0DE45A49CD76099821272F86BDF@CORP-MAIL-002.edge.local> Hi, I am trying to understand mod_diangling and followed documentation at http://wiki.freeswitch.org/wiki/Dingaling As I am running on Windows, Installed GnuTls and generated libgnutls-26.lib file. Added additional include directory and dependencies to iksemel library project. Now when I compile, it should generate respective DLLs right. Unfortunately I am not finding the TLS DLLs' as mentioned in the link. I tried to compile whole solution and also compiled iksemel project and mod_diangling project separately. Still I didn't find the TLS DLLs in output (Release) folder. Can someone please let me know what I am missing here. Also please let me know if I need to follow any other procedure. Note: I am using latest version of FreeSwitch from GIT and using Visual Studio 2010. Thanks & Regards Suneel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/db573535/attachment-0001.html From Nabble_01394 at slickdeals.endjunk.com Wed Nov 16 22:07:26 2011 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Wed, 16 Nov 2011 11:07:26 -0800 (PST) Subject: [Freeswitch-users] Issue with mod_diangling during compilation In-Reply-To: <9438D04074E0DE45A49CD76099821272F86BDF@CORP-MAIL-002.edge.local> References: <9438D04074E0DE45A49CD76099821272F86BDF@CORP-MAIL-002.edge.local> Message-ID: <1321470446167-7001513.post@n2.nabble.com> Papineni, Suneel wrote: > Unfortunately I am not finding the TLS DLLs' as mentioned in the link. > > I tried to compile whole solution and also compiled iksemel project and > mod_diangling project separately. Still I didn't find the TLS DLLs in > output (Release) folder.I thought starting from git commit f506e19e152e070f04574ad67e282af8691606f5 libdingaling no longer uses GNUTLS but rather OpenSSL. See this http://jira.freeswitch.com/browse/FS-3471 FS-3471 . ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-mod-diangling-during-compilation-tp7001164p7001513.html Sent from the freeswitch-users mailing list archive at Nabble.com. From don.dawson at voice-ring.com Wed Nov 16 23:36:53 2011 From: don.dawson at voice-ring.com (Don Dawson) Date: Wed, 16 Nov 2011 14:36:53 -0600 Subject: [Freeswitch-users] call multiple contacts for ext Message-ID: <4EC41EE5.8050307@voice-ring.com> We have been working on getting the SLA features to work on freeswitch. We have been successful, however, when we have an EXT that 2 phones register with, the SLA works but is only calling one of the phones(sofia/internal/121%domain). If we try using "$SOFIA_CONTACT" in the bridge app then both phones are called but the SLA lights do not work. I presume its because of bypassing the "internal" profile. Is there a way to call all of an extensions contacts using the profile(internal in this case)? From avi at avimarcus.net Thu Nov 17 01:40:51 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 17 Nov 2011 00:40:51 +0200 Subject: [Freeswitch-users] calling limit_usage via ESL Message-ID: I'm using the php esl implementation in fusionpbx.. I'm able to pass "api show channels count" and "api show calls count" and the like, but when I do: "api limit_usage hash origination usa_pd" it's just blank. Pasting "limit_usage hash origination usa_pd" into fs_cli yields 0 (or higher as the case may be). There DOES seem to be a \n returned, but nothing else. I've got no 'friggin clue. Help, please? Thanks, Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/16d5b977/attachment.html From Hector.Geraldino at ip-soft.net Thu Nov 17 02:00:53 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Wed, 16 Nov 2011 18:00:53 -0500 Subject: [Freeswitch-users] calling limit_usage via ESL In-Reply-To: References: Message-ID: <6A6B4C284AD15042B429EB9D904544AD0224D44767@NY1-EXMB-01.ip-soft.net> I think that, if it's an application (i.e. can be executed directly from the dialplan), you shouldn't use the 'api' prefix. I'm not familiar with the php esl module, but in the Java ESL there's a difference when you try to call an api command vs and application: API: "api " + command + " " + arguments APP: SendMsg message = new SendMsg(uuid); message.addCallCommand("execute"); message.addExecuteAppName(appName); message.addExecuteAppArg(arguments); Also, you can always inspect the execution result if you're executing the command in sync mode. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Wednesday, November 16, 2011 5:41 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] calling limit_usage via ESL I'm using the php esl implementation in fusionpbx.. I'm able to pass "api show channels count" and "api show calls count" and the like, but when I do: "api limit_usage hash origination usa_pd" it's just blank. Pasting "limit_usage hash origination usa_pd" into fs_cli yields 0 (or higher as the case may be). There DOES seem to be a \n returned, but nothing else. I've got no 'friggin clue. Help, please? Thanks, Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/6bfd9f82/attachment.html From avi at avimarcus.net Thu Nov 17 02:19:55 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 17 Nov 2011 01:19:55 +0200 Subject: [Freeswitch-users] calling limit_usage via ESL In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0224D44767@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD0224D44767@NY1-EXMB-01.ip-soft.net> Message-ID: The wiki says it's an API also. I have frequently executed it directly in fs_cli. When I try it without "api" at the start, it hangs. -Avi On Thu, Nov 17, 2011 at 1:00 AM, Hector Geraldino < Hector.Geraldino at ip-soft.net> wrote: > I think that, if it?s an application (i.e. can be executed directly from > the dialplan), you shouldn?t use the ?api? prefix.**** > > ** ** > > I?m not familiar with the php esl module, but in the Java ESL there?s a > difference when you try to call an api command vs and application:**** > > ** ** > > API:**** > > "api " + command + " " + arguments**** > > ** ** > > APP:**** > > SendMsg message = new SendMsg(uuid);**** > > message.addCallCommand("execute");**** > > message.addExecuteAppName(appName);**** > > message.addExecuteAppArg(arguments);**** > > ** ** > > Also, you can always inspect the execution result if you?re executing the > command in sync mode.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Wednesday, November 16, 2011 5:41 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] calling limit_usage via ESL**** > > ** ** > > I'm using the php esl implementation in fusionpbx.. I'm able to pass "api > show channels count" and "api show calls count" and the like, but when I do: > **** > > "api limit_usage hash origination usa_pd" it's just blank.**** > > Pasting "limit_usage hash origination usa_pd" into fs_cli yields 0 (or > higher as the case may be).**** > > ** ** > > There DOES seem to be a \n returned, but nothing else. I've got no > 'friggin clue.**** > > Help, please?**** > > > **** > > Thanks,**** > > Avi**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/5bb6931c/attachment-0001.html From pete at privateconnect.com Thu Nov 17 03:16:36 2011 From: pete at privateconnect.com (Pete Mueller) Date: Wed, 16 Nov 2011 17:16:36 -0700 Subject: [Freeswitch-users] =?utf-8?q?mod=5Frtmp_and_a_security_camera?= Message-ID: <20111116171633.2ad02225396a31c9de30536f2e338977.9b974c35a1.wbe@email13.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111116/ce7a7a03/attachment.html From david at styleflare.com Thu Nov 17 03:23:52 2011 From: david at styleflare.com (David J) Date: Wed, 16 Nov 2011 19:23:52 -0500 Subject: [Freeswitch-users] mod_rtmp and a security camera In-Reply-To: <20111116171633.2ad02225396a31c9de30536f2e338977.9b974c35a1.wbe@email13.secureserver.net> References: <20111116171633.2ad02225396a31c9de30536f2e338977.9b974c35a1.wbe@email13.secureserver.net> Message-ID: Why would you want to do this. Just use flash in the browser. What purpose does freeswitch serve? On Nov 16, 2011 7:18 PM, "Pete Mueller" wrote: > Hey all, question for you. > > I have an IP-based security camera (made by AXIS) that produces a RTSP > stream containing AAC audio and H.264 video. What I'd like to do is view > that audio/video on the web using a flash player. It seems that FreeSWITCH > has the necessary components to do this. I'm just not sure how to hook it > all together. > > My line of thinking is: > 1. Create an extension that when called streams the audio and video from > the static url. The url is in the form of: > rtsp://10.1.1.1/axis-media/media.amp > > 2. Create an extension associated with mod_rtmp, and connect the flex > client. > > 3. Have the flex client "call" the rtsp extension. > > I've seen instructions for #2 and #3, but I'm not sure how to do #1. Any > ideas? Or am I way off? > > -pete > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111116/91c5f6ab/attachment.html From david at styleflare.com Thu Nov 17 03:25:16 2011 From: david at styleflare.com (David J) Date: Wed, 16 Nov 2011 19:25:16 -0500 Subject: [Freeswitch-users] mod_rtmp and a security camera In-Reply-To: <20111116171633.2ad02225396a31c9de30536f2e338977.9b974c35a1.wbe@email13.secureserver.net> References: <20111116171633.2ad02225396a31c9de30536f2e338977.9b974c35a1.wbe@email13.secureserver.net> Message-ID: Also the flex client does not support video yet. I think there is better ways to do this using web technologies On Nov 16, 2011 7:18 PM, "Pete Mueller" wrote: > Hey all, question for you. > > I have an IP-based security camera (made by AXIS) that produces a RTSP > stream containing AAC audio and H.264 video. What I'd like to do is view > that audio/video on the web using a flash player. It seems that FreeSWITCH > has the necessary components to do this. I'm just not sure how to hook it > all together. > > My line of thinking is: > 1. Create an extension that when called streams the audio and video from > the static url. The url is in the form of: > rtsp://10.1.1.1/axis-media/media.amp > > 2. Create an extension associated with mod_rtmp, and connect the flex > client. > > 3. Have the flex client "call" the rtsp extension. > > I've seen instructions for #2 and #3, but I'm not sure how to do #1. Any > ideas? Or am I way off? > > -pete > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111116/12a10b8c/attachment.html From david at styleflare.com Thu Nov 17 03:28:17 2011 From: david at styleflare.com (David J) Date: Wed, 16 Nov 2011 19:28:17 -0500 Subject: [Freeswitch-users] mod_rtmp and a security camera In-Reply-To: References: <20111116171633.2ad02225396a31c9de30536f2e338977.9b974c35a1.wbe@email13.secureserver.net> Message-ID: 3 years ago I worked on a project that did this with axis 210 cameras. We used red5 server. Hope that helps On Nov 16, 2011 7:25 PM, "David J" wrote: > Also the flex client does not support video yet. > > I think there is better ways to do this using web technologies > On Nov 16, 2011 7:18 PM, "Pete Mueller" wrote: > >> Hey all, question for you. >> >> I have an IP-based security camera (made by AXIS) that produces a RTSP >> stream containing AAC audio and H.264 video. What I'd like to do is view >> that audio/video on the web using a flash player. It seems that FreeSWITCH >> has the necessary components to do this. I'm just not sure how to hook it >> all together. >> >> My line of thinking is: >> 1. Create an extension that when called streams the audio and video from >> the static url. The url is in the form of: >> rtsp://10.1.1.1/axis-media/media.amp >> >> 2. Create an extension associated with mod_rtmp, and connect the flex >> client. >> >> 3. Have the flex client "call" the rtsp extension. >> >> I've seen instructions for #2 and #3, but I'm not sure how to do #1. Any >> ideas? Or am I way off? >> >> -pete >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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/20111116/abeee857/attachment.html From freeswitch at earthspike.net Thu Nov 17 03:54:18 2011 From: freeswitch at earthspike.net (John) Date: Thu, 17 Nov 2011 00:54:18 +0000 Subject: [Freeswitch-users] Help with choppy audio after attended transfer In-Reply-To: <41b80a3a-4e9e-410f-8c9e-c1eb5cc02f91@Raimund-ThinkPad-X61s> References: <41b80a3a-4e9e-410f-8c9e-c1eb5cc02f91@Raimund-ThinkPad-X61s> Message-ID: <4EC45B3A.1030600@earthspike.net> I also get the problem intermittently, on the outgoing leg only. I have a Sangoma B700 dual BRI card talking to the PSTN and Yealink T20Ps on the inside. Everything is G.711 so there is no transcoding. The ISDN incoming calls are answered by a Lua-based IVR, but the choppiness starts after MOH and the call is picked up, or when the extension voicemail greeting is played. Outgoing calls also have the outbound leg experiencing chopping. I estimated the chopping to be at around 5Hz or so, which matches the 160octet size being fed into the ISDN G.711. A reboot cleared all the symptoms (drastic, I know, but I was on a dialup connection on another job and didn't have time to debug). I will go through and check that all the ptimes match (eg all at 20ms), but it appears that this bug still lives, at least on this git: > version FreeSWITCH Version 1.0.head (git-6fe6d8d 2011-11-01 11-52-24 -0500) John On 17/10/11 08:55, Raimund Sacherer wrote: > Are you still gettint this Problem? I ran into similiar issues, but > it's intermittent ... > > best > Ray > > ------------------------------------------------------------------------ > *From: *"Chris Cureau" > *To: *"FreeSWITCH Users Help" > *Sent: *Wednesday, August 3, 2011 1:28:22 AM > *Subject: *Re: [Freeswitch-users] Help with choppy audio after > attended transfer > > Latest git doesn't help unfortunately. > > More research is beginning to confirm its a codec issue. When I limit > the phone to codes that use 20ms packetization, I hear no incoming > audio on outgoing calls. Incoming calls are okay. When using All for > codes (and the 30ms packetization) I get the choppy voice when leaving > MOH. > > > > On Mon, Aug 1, 2011 at 8:34 PM, Anthony Minessale > > wrote: > > can you do make current and try latest git? > > > On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau > wrote: > > Anthony, > > > > Thanks for answering...and sorry for the delay. I've already > checked all of > > the ptime settings I can, and all phones plus freeswitch are set > to use 20ms > > packetization. I've even set "scrooge" in the codec > negotiation, but I keep > > running into this issue. I've updated my post with "sofia > global siptrace > > on". > > > > I am assuming that the ptime issue happens around line 2462 > > (http://pastebin.freeswitch.org/16935) > > > > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec > Compare > > [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000] > > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already > using PCMU > > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 > dtmf send > > payload to 101 > > 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing > updated SDP > > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio > params are > > unchanged for sofia/internal/sip:1003 at 10.0.1.205:5060 > . > > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052 > > sofia/internal/sip:1003 at 10.0.1.205:5060 > Setting audio receive payload in > > Re-INVITE to 0 > > > > Could this be an issue with the Aastra'a firmware? Or maybe the > MOH is > > being processed at 30ms instead of 20ms, and the negotiation is > not updated > > somehow? > > > > I don't mean to sound ignorant, but I'm really at a loss > here...and thanks > > again for any help! > > > > Cheers, > > Chris > > > > On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale > > > wrote: > >> > >> probably ptime related thing. > >> you should have included the sip trace "sofia global siptrace on" > >> > >> > >> On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau > > wrote: > >> > I'm having some issues with extremely choppy audio after a > call has been > >> > sent to another extension via an automated transfer. The > audio is great > >> > when the call is answered. Shortly after, the transfer button is > >> > pressed > >> > and the incoming call hears music on hold. The music on hold > is sent to > >> > the > >> > caller sounds fine as does the conversation between > extensions. When > >> > the > >> > transfer is completed, the caller hears what sounds like someone > >> > speaking > >> > through a fan (though slower) but incoming audio sounds fine. > >> > > >> > Since it's such a large log, I posted it to the FreeSWITCH > pastebin: > >> > http://pastebin.freeswitch.org/16911 > >> > > >> > I'm thinking that it has something to do with the transition > from MOH to > >> > the > >> > internal extension, but I can't figure out what is happening. > >> > > >> > Any ideas? > >> > > >> > _______________________________________________ > >> > Join us at ClueCon 2011, Aug 9-11, Chicago > >> > http://www.cluecon.com 877-7-4ACLUE > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > > >> googletalk:conf+888 at conference.freeswitch.org > > >> pstn:+19193869900 > >> > >> _______________________________________________ > >> Join us at ClueCon 2011, Aug 9-11, Chicago > >> http://www.cluecon.com 877-7-4ACLUE > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Join us at ClueCon 2011, Aug 9-11, Chicago > > http://www.cluecon.com 877-7-4ACLUE > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Logitravel.com > Raimund Sacherer > /Sistemas/ > Agencia de Viajes Online > www.logitravel.com > Edificio Logitravel, Parcela 3B (Parc Bit) > Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca > Tel 902 366 847 | Fax 971 213 495 > S?guenos en: Facebook de Logitravel > Twitter de Logitravel > Blog de Logitravel > Logitravel en Youtube > Logitravel en Foursquare > > > > Descarga nuestras *aplicaciones para m?vil* Logitravel.com > > > Este correo electr?nico y, en su caso, cualquier fichero anexo, > contiene informaci?n de car?cter confidencial exclusivamente dirigida > a su destinatario. Queda prohibida su divulgaci?n, copia o > distribuci?n a terceros sin la previa autorizaci?n escrita de > LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por > error, se ruega notif?quese inmediatamente esta circunstancia mediante > reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA > EMPRESA le recuerda que sus datos forman o formar?n parte de un > fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 > en la Agencia General de Protecci?n de Datos, propiedad de la empresa > LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - > Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de > acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que > deber? ejercer mediante escrito a la direcci?n anteriormente citada. > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/5061c9a6/attachment-0001.html From don.dawson at voice-ring.com Thu Nov 17 01:35:51 2011 From: don.dawson at voice-ring.com (Don Dawson) Date: Wed, 16 Nov 2011 16:35:51 -0600 Subject: [Freeswitch-users] call multiple contacts for ext In-Reply-To: <4EC41EE5.8050307@voice-ring.com> References: <4EC41EE5.8050307@voice-ring.com> Message-ID: <4EC43AC7.4030107@voice-ring.com> On 11/16/2011 2:36 PM, Don Dawson wrote: > We have been working on getting the SLA features to work on freeswitch. > We have been successful, however, when we have an EXT that 2 phones > register with, > the SLA works but is only calling one of the > phones(sofia/internal/121%domain). If we try using "$SOFIA_CONTACT" in > the bridge app then both phones are called but the SLA lights do not > work. I presume its because of bypassing the "internal" profile. Is > there a way to call all of an extensions contacts using the > profile(internal in this case)? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 Its working now. I was missing {presence_id=}. From jeff at jefflenk.com Thu Nov 17 06:58:41 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 16 Nov 2011 19:58:41 -0800 (PST) Subject: [Freeswitch-users] Issue with mod_diangling during compilation In-Reply-To: <1321470446167-7001513.post@n2.nabble.com> References: <9438D04074E0DE45A49CD76099821272F86BDF@CORP-MAIL-002.edge.local> <1321470446167-7001513.post@n2.nabble.com> Message-ID: <1321502321595-7002749.post@n2.nabble.com> That is correct. Git as of around Oct 17 no longer uses GnuTLS and now uses OpenSSL. The build for windows now builds correctly for ssl out of the box so to speak. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-mod-diangling-during-compilation-tp7001164p7002749.html Sent from the freeswitch-users mailing list archive at Nabble.com. From shouldbeq931 at gmail.com Thu Nov 17 13:57:38 2011 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Thu, 17 Nov 2011 10:57:38 +0000 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: <4EC3B7A8.7010306@integrafin.co.uk> References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> <4EC3B7A8.7010306@integrafin.co.uk> Message-ID: On Wed, Nov 16, 2011 at 1:16 PM, Alex Crow wrote: > On 14/11/11 11:24, Tihomir Culjaga wrote: > > On Fri, Nov 11, 2011 at 4:32 PM, Steve Underwood wrote: >> >> Turning off ECM is equivalent to saying "I use FAX, but I couldn't give >> a damn if they are received". > > and it always work if the network is stable (you might loose a pixel here > and there...)! If you are running it via open internet, sorry but try > something else (T.38 or switch to e-mail). You cannot expect pass-through to > work reliably in such scenarios. > > T. > > > > Hi all, > > I built a new machine for this. I know faxing via audio is not the done > thing, however this is only to the local Mitel box then out via ISDN until > we get the licenses and/or E1 cards for both machine. On the new machine I > built a new kernel without tickless and with 1000Hz - and lo and behold ECM > faxing now works! > > Cheers > > Alex > That's really good to hear! From fieldpeak at gmail.com Thu Nov 17 14:58:48 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Thu, 17 Nov 2011 19:58:48 +0800 Subject: [Freeswitch-users] How to identify the first REGISTERATION and do sth. Message-ID: Dear friends, i would to excute an script after the 200 OK agaist teh first REGISTERATION from the user, so i need identify which one is the first REGISTERATION message, in the sofia_reg.c, i found below lines, however, it looks no help, could anyone can help advise, thanks. "if (sofia_test_pflag(profile, PFLAG_MESSAGE_QUERY_ON_REGISTER) || (reg_count == 1 && sofia_test_pflag(profile, PFLAG_MESSAGE_QUERY_ON_FIRST_REGISTER)))" -- Regards, Charles From davidwaf at gmail.com Thu Nov 17 15:34:05 2011 From: davidwaf at gmail.com (David Wafula) Date: Thu, 17 Nov 2011 14:34:05 +0200 Subject: [Freeswitch-users] listing conference participants Message-ID: Hello all, How does one get a list of participants currently in a conference? I want to do something similar like what is on http://conference.freeswitch.org regards, -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/add631f0/attachment.html From cmrienzo at gmail.com Thu Nov 17 15:40:11 2011 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Thu, 17 Nov 2011 07:40:11 -0500 Subject: [Freeswitch-users] listing conference participants In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference see "list" On Thu, Nov 17, 2011 at 7:34 AM, David Wafula wrote: > Hello all, > How does one get a list of participants currently in a conference? I want > to do something similar like what is on http://conference.freeswitch.org > > regards, > -- > David Wafula > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/d1954aea/attachment.html From davidwaf at gmail.com Thu Nov 17 15:42:53 2011 From: davidwaf at gmail.com (David Wafula) Date: Thu, 17 Nov 2011 14:42:53 +0200 Subject: [Freeswitch-users] listing conference participants In-Reply-To: References: Message-ID: On Thu, Nov 17, 2011 at 2:40 PM, Christopher Rienzo wrote: > http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference > > see "list" > Exactly what i was looking for. Thanks. -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/1a3c79c1/attachment.html From mpicher at gmail.com Thu Nov 17 16:03:22 2011 From: mpicher at gmail.com (Michael Picher) Date: Thu, 17 Nov 2011 08:03:22 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: Message-ID: Not having anywhere near the same trouble in a Xen Server in my lab... Although test_time is skewed 180 degrees from what it was in AWS.... freeswitch at internal> timer_test 120 10 Avg: 119.945ms Total Time: 1199.727ms 2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120993 2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 118914 2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119919 2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120032 2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 119802 2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 119999 2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119975 2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 120003 2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 119849 2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119971 freeswitch at internal> time_test 600 10 test 1 sleep 600 1 test 2 sleep 600 1 test 3 sleep 600 0 test 4 sleep 600 1 test 5 sleep 600 0 test 6 sleep 600 1 test 7 sleep 600 1 test 8 sleep 600 0 test 9 sleep 600 0 test 10 sleep 600 1 avg 0 [root at openuc bin]# uname -r 2.6.18-274.7.1.el5 [root at openuc bin]# grep CONFIG_HZ /boot/config-* /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000 Thoughts as to why AWS results are so different from XenServer? Other than not knowing who else is on the AWS box? Thanks, Mike On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher wrote: > m1.large > > I have a c1.xlarge queued up and ready to test... > > > On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen wrote: > >> Just a simple question, what kind of AWS instance are you running your >> FreeSWITCH? >> It makes huge difference. >> Thanks, >> Chris >> >> On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher wrote: >> >>> Hi guys, >>> >>> Trying to get to the bottom of some conference bridge issues I'm having >>> with running the system in AWS. >>> >>> We're hearing a bunch of snap-crackle-pops in conference bridges and >>> when I tcpdum on the server itself I see them in the RTP and see RTP >>> timestamp problems. >>> >>> I've run the following: >>> >>> >>> timer_test >>> freeswitch at 127.0.0.1@internal> timer_test 120 10 >>> >>> Avg: 120.004ms Total Time: 1200.315ms >>> >>> 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: >>> samplecount after init: 1 >>> freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] >>> mod_commands.c:466 Timer Test: samplecount after first step: 2 >>> 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 >>> sleep 120 120006 >>> 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 >>> sleep 120 120040 >>> 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 >>> sleep 120 119976 >>> 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 >>> sleep 120 120007 >>> 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 >>> sleep 120 120001 >>> 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 >>> sleep 120 120031 >>> 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 >>> sleep 120 119996 >>> 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 >>> sleep 120 119994 >>> 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 >>> sleep 120 120005 >>> 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 >>> sleep 120 119991 >>> >>> test_time >>> freeswitch at 127.0.0.1@internal> time_test 600 10 >>> >>> test 1 sleep 600 1592 >>> test 2 sleep 600 986 >>> test 3 sleep 600 1018 >>> test 4 sleep 600 980 >>> test 5 sleep 600 1005 >>> test 6 sleep 600 1000 >>> test 7 sleep 600 972 >>> test 8 sleep 600 990 >>> test 9 sleep 600 1006 >>> test 10 sleep 600 994 >>> avg 1054 >>> >>> For kernel: >>> [root at openuc bin]# uname -r >>> >>> 2.6.21.7-2.fc8xen >>> >>> CONFIG_HZ: >>> [root at openuc bin]# grep CONFIG_HZ /boot/config-* >>> >>> /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y >>> /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set >>> /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set >>> /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 >>> /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set >>> /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set >>> /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y >>> /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 >>> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set >>> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set >>> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set >>> /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y >>> /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 >>> >>> are the xenU kernel settings screwing me here? >>> >>> Thanks, >>> Mike >>> >>> >>> -- >>> There are 10 kinds of people in this world, those who understand binary >>> and those who don't. >>> >>> mpicher at gmail.com >>> blog: http://www.sipxecs.info >>> call: sip:mpicher at sipxecs.info >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info > -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/b52fbd99/attachment-0001.html From avi at avimarcus.net Thu Nov 17 16:06:08 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 17 Nov 2011 15:06:08 +0200 Subject: [Freeswitch-users] iptables dropping SIP packets..? Message-ID: Service: sip (udp/5060) (nf_ct_sip: dropping packet) - 9 packets I see this every few days - it's via an IP with a Yealink T20p phone. I've tried looking this up in the past.. but I don't understand what rules would be dropping this, and what is being dropped. Any clues on how to investigate? -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/85453158/attachment.html From turqmr2 at gmail.com Thu Nov 17 16:07:28 2011 From: turqmr2 at gmail.com (Jacob Smith) Date: Thu, 17 Nov 2011 08:07:28 -0500 Subject: [Freeswitch-users] Getting past version through GIT In-Reply-To: References: <002701cc9f2e$d47dfbc0$7d79f340$@com> Message-ID: <4EC50710.2010102@gmail.com> I have run into a bug (http://jira.freeswitch.org/browse/FS-3689) and since mod_dingaling recently moved from TLS to SSL I thought I might be able to run an older version. So, I have a two part question: How can I download an older version of FS and is there a known working version that someone can recommend? Thanks, Jacob From x.liu at hw.ac.uk Thu Nov 17 16:28:19 2011 From: x.liu at hw.ac.uk (xl127) Date: Thu, 17 Nov 2011 13:28:19 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <4EC3EA04.7030605@hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> <4EC3D5AD.5070507@hw.ac.uk> <4EC3EA04.7030605@hw.ac.uk> Message-ID: <4EC50BF3.1070905@hw.ac.uk> Hi Christopher, After some more tests, I found followings: 1. Testing from dialplan, the log output is the string of "CRIT ${speech_detect_result}" rather than the recognition results. 2. Regarding to non barge in, I first send "speak" command then send play_and_detect_speech with parameter: detect:unimrcp:nuance5-mrcp1-1 {start-input-timers=false,no-input-timeout=25000,recognition-timeout=25000}dudeYNNC_Nuance in which I removed the "say:" part. But very soon I received event CHANNEL_EXECUTE_COMPLETE for play_and_detect_speech before the text playing finishes, and speech_detect_result is null (actually the event header does not contain this variable) I tried with "say:" part with empty text like "say:unimrcp:en-GB: " but it doesn't work, see issue 3 below. 3. It seems that I can not send "speak" command twice before the first one finishes. In my case I run one port TTS server on one machine. If I send the command twice FS will give me Synthesizer Error/Invalid TTS Module. I thought the TTS request would be queued rather than it immediately looks for the TTS resource. If I send the second command to another TTS machine, no error occurs but I can only hear one utterance being spoken, it looks like one utteraance was dropped somehow. Any ideas? Thanks! Xing On 16/11/11 16:51, xl127 wrote: > Hi Christopher, > > The questions are cleared to me now. Many thanks for your explanations! > > Best regards, > > Xing > > > On 16/11/11 15:52, Christopher Rienzo wrote: >> >> Responses inline >> >> >> Now it works in my ESL app though I am just able to do one >> dialogue ( I need to add the event catching for furthur dialgoues). >> >> I have a couple of questions here: >> >> 1. In the first try, my Nuance server was able to be accessed >> somehow (FS says the MRCP is not responding in 5000ms, >> something like that), then FS says: [WARNING] >> rtsp_client.c:386 () Failed to Connect to RTSP Server >> *MailScanner warning: numerical links are often malicious:* >> 99.185.85.31:554 , >> later FS says: >> [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER channel error! >> [ERR] switch_ivr_play_say.c:2439 Invalid TTS module! >> >> The SYNTHESIZER channel error and Invalid TTS module error are >> obvious. >> >> What I don't understand is why it went to this stange >> address: *MailScanner warning: numerical links are often >> malicious:* 99.185.85.31:554 ? >> >> >> check your unimrcp configuration. Make sure the default TTS and ASR >> profiles are set to actual servers. >> >> 2. I specified TTS engine in play_and_detect_speech as >> "say:unimrcp:nuance5-mrcp1-1: the text to speak" >> It works though I didn't specify the TTS voice. >> >> How do I specify the TTS voice? In the mrcp profile (how?)? >> or something like: >> "say:unimrcp:nuance5-mrcp1-1:Serena: the text to speak" >> (this seems not right.) >> >> >> That won't work. Set the tts_engine variable as I explained >> previously, or use say:unimrcp:voice:text to speak with the desired >> voice and the correct default TTS profile defined in >> unimrcp.conf.xml. This is a limitation of the say: notation. >> Alternatively, the voice can be defined with the tts_voice channel >> variable. >> >> 3. The barge-in works well, thanks!. Is the barge-in >> configurable? In some scenarios, we might not allow barge-in. >> >> >> If you don't want to barge in, just do "playback (or speak)" first, >> then "play_and_detect_speech" with a silence prompt. >> >> >> 4. How could I get the text which has spoken to the user when >> barge-in occurs? >> Or Could I get the time when barge-in occurs? If I know the >> barge-in time and rough totale time for the whole text >> to be spoken I can figure out the spoken text by manually >> checking the recorded audio file later, which would be painful. >> >> >> If this is necessary, you might want to use the lower-level functions >> instead to watch for the begin-speaking event. >> >> >> 5. when I use "speak" and "detect_speech" apps in ESL, I can >> catch event: DETECTED_SPEECH and speech-type: begin-speaking >> and "detected-speech", then I do the recognition results >> processing. >> >> The new app play_and_detect_speech seems not generate these >> events any more. The way that I can think of to get the results >> is to catch event:CHANNEL_EXECUTE_COMPLETE then check if >> variable_current_application=play_and_detect_speech, then get >> the results from variable_detect_speech_result. >> >> Is this the proper way to get the results in ESL app? Or will >> play_and_detect_speech later on be consistent with detect_speech >> in term of ASR events? >> >> >> play_and_detect_speech is a higher level abstraction to simplify >> things. If you want to have more control, go back to using the ESL >> events. Reading the code in mod_dptools and switch_ivr_async will >> give you hints about how to do it correctly. >> >> >> 6. I'd like to set start-input-timers=false in the initial >> request then start the recognition timers (start-input-timers=true) >> after the TTS finishes. >> How possibly could I do this? >> >> >> This is automatically done in the >> switch_ivr_play_and_detect_speech() function. You just need to >> specify start-input-timers=false in the beginning. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > ------------------------------------------------------------------------ > > Scottish University of the Year 2011-12 *Heriot-Watt University is the > Sunday Times > Scottish University of the Year 2011-2012* > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/d8e8b047/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/d8e8b047/attachment-0001.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/d8e8b047/attachment-0001.jpg From thomas at chaschperli.ch Thu Nov 17 16:32:38 2011 From: thomas at chaschperli.ch (Thomas Mueller) Date: Thu, 17 Nov 2011 14:32:38 +0100 Subject: [Freeswitch-users] iptables dropping SIP packets..? In-Reply-To: References: Message-ID: <4EC50CF6.8090103@chaschperli.ch> On 17.11.2011 14:06, Avi Marcus wrote: > Service: sip (udp/5060) (nf_ct_sip: dropping packet) - 9 packets > > I see this every few days - it's via an IP with a Yealink T20p phone. > I've tried looking this up in the past.. but I don't understand what > rules would be dropping this, and what is being dropped. > Any clues on how to investigate? Maybe this is related to this bugreport? nf_ct_sip dropping SIP messages larger then MTU http://bugzilla.netfilter.org/show_bug.cgi?id=760 There is some possibility that you don't need the nf_ct_sip - just unload it (modprobe -r) and if it works, do blacklist the module in (for exmple in /etc/modprobe.d/00-my-local.conf ) - Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/1fed25cc/attachment.html From cmrienzo at gmail.com Thu Nov 17 16:37:45 2011 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Thu, 17 Nov 2011 08:37:45 -0500 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <4EC50BF3.1070905@hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> <4EC3D5AD.5070507@hw.ac.uk> <4EC3EA04.7030605@hw.ac.uk> <4EC50BF3.1070905@hw.ac.uk> Message-ID: > After some more tests, I found followings: > > 1. Testing from dialplan, the log output is the string of "CRIT > ${speech_detect_result}" rather than the recognition results. > oops > > 2. Regarding to non barge in, I first send "speak" command then send > play_and_detect_speech with parameter: > detect:unimrcp:nuance5-mrcp1-1 > {start-input-timers=false,no-input-timeout=25000,recognition-timeout=25000}dudeYNNC_Nuance > > in which I removed the "say:" part. > > But very soon I received event CHANNEL_EXECUTE_COMPLETE for > play_and_detect_speech before the text playing finishes, > and speech_detect_result is null (actually the event header does not > contain this variable) > > I tried with "say:" part with empty text like "say:unimrcp:en-GB: " > but it doesn't work, see issue 3 below. > Try with silence as I originally suggested: silence_stream://1000 detect:unimrcp:nuance5-mrcp1-1 {start-input-timers=false,no-input-timeout=25000, recognition-timeout=25000}dudeYNNC_Nuance > > 3. It seems that I can not send "speak" command twice before the first > one finishes. In my case I run one port TTS server on one machine. > > If I send the command twice FS will give me Synthesizer Error/Invalid > TTS Module. > I thought the TTS request would be queued rather than it immediately > looks for the TTS resource. > > If I send the second command to another TTS machine, no error occurs > but I can only hear one utterance being spoken, > it looks like one utteraance was dropped somehow. > Wait for speak to finish before starting a new one. > > On 16/11/11 16:51, xl127 wrote: > > Hi Christopher, > > The questions are cleared to me now. Many thanks for your explanations! > > Best regards, > > Xing > > > On 16/11/11 15:52, Christopher Rienzo wrote: > > > Responses inline > > > Now it works in my ESL app though I am just able to do one dialogue ( I >> need to add the event catching for furthur dialgoues). >> >> I have a couple of questions here: >> >> 1. In the first try, my Nuance server was able to be accessed somehow >> (FS says the MRCP is not responding in 5000ms, >> something like that), then FS says: [WARNING] rtsp_client.c:386 () >> Failed to Connect to RTSP Server *MailScanner has detected a possible >> fraud attempt from "99.185.85.31:554" claiming to be* *MailScanner >> warning: numerical links are often malicious:* 99.185.85.31:554 >> , >> >> later FS says: >> [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER channel error! >> [ERR] switch_ivr_play_say.c:2439 Invalid TTS module! >> >> The SYNTHESIZER channel error and Invalid TTS module error are obvious. >> >> What I don't understand is why it went to this stange address: *MailScanner >> has detected a possible fraud attempt from "99.185.85.31:554" claiming to be >> * *MailScanner warning: numerical links are often malicious:*99.185.85.31:554 >> ? >> > > check your unimrcp configuration. Make sure the default TTS and ASR > profiles are set to actual servers. > > >> 2. I specified TTS engine in play_and_detect_speech as >> "say:unimrcp:nuance5-mrcp1-1: the text to speak" >> It works though I didn't specify the TTS voice. >> >> How do I specify the TTS voice? In the mrcp profile (how?)? or >> something like: >> "say:unimrcp:nuance5-mrcp1-1:Serena: the text to speak" (this >> seems not right.) >> > > That won't work. Set the tts_engine variable as I explained previously, > or use say:unimrcp:voice:text to speak with the desired voice and the > correct default TTS profile defined in unimrcp.conf.xml. This is a > limitation of the say: notation. Alternatively, the voice can be defined > with the tts_voice channel variable. > > > >> 3. The barge-in works well, thanks!. Is the barge-in configurable? In >> some scenarios, we might not allow barge-in. >> > > If you don't want to barge in, just do "playback (or speak)" first, then > "play_and_detect_speech" with a silence prompt. > > >> >> 4. How could I get the text which has spoken to the user when barge-in >> occurs? >> Or Could I get the time when barge-in occurs? If I know the barge-in >> time and rough totale time for the whole text >> to be spoken I can figure out the spoken text by manually checking >> the recorded audio file later, which would be painful. >> > > If this is necessary, you might want to use the lower-level functions > instead to watch for the begin-speaking event. > > >> >> 5. when I use "speak" and "detect_speech" apps in ESL, I can catch >> event: DETECTED_SPEECH and speech-type: begin-speaking >> and "detected-speech", then I do the recognition results processing. >> >> The new app play_and_detect_speech seems not generate these events >> any more. The way that I can think of to get the results >> is to catch event:CHANNEL_EXECUTE_COMPLETE then check if >> variable_current_application=play_and_detect_speech, then get >> the results from variable_detect_speech_result. >> >> Is this the proper way to get the results in ESL app? Or will >> play_and_detect_speech later on be consistent with detect_speech >> in term of ASR events? >> > > play_and_detect_speech is a higher level abstraction to simplify things. > If you want to have more control, go back to using the ESL events. Reading > the code in mod_dptools and switch_ivr_async will give you hints about how > to do it correctly. > > >> >> 6. I'd like to set start-input-timers=false in the initial request then >> start the recognition timers (start-input-timers=true) >> after the TTS finishes. >> How possibly could I do this? >> > > This is automatically done in the switch_ivr_play_and_detect_speech() > function. You just need to specify start-input-timers=false in the > beginning. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > ------------------------------ > > [image: Scottish University of the Year 2011-12] *Heriot-Watt > University is the Sunday Times > Scottish University of the Year 2011-2012* > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > ------------------------------ > > [image: MailScanner Signature HW] *Heriot-Watt University is the Sunday > Times > > Scottish University of the Year 2011-2012 > * > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/8a0a143d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/8a0a143d/attachment-0002.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/8a0a143d/attachment-0003.jpe From sdame at 207me.com Thu Nov 17 16:49:01 2011 From: sdame at 207me.com (Stephen Dame) Date: Thu, 17 Nov 2011 08:49:01 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: Message-ID: <005501cca52f$aa0286c0$fe079440$@com> I'm running freeswitch on about 40 different m1.small c1.medium's in AWS regions, us-east, us-west, eu-west, and asia.. They are used in videoconf component of BigBlueButton.org . For most part they work great. There are occasional issues with voip quality but the app is 100% voip, with the BBB client all browser based. So the conference is subject to every ones local network connections and most issues "blamed" on the internet instead of freeswitch J We also tie in skype and DID direct to improve latency for some clients. But expectations are set so we meet them. You probably have tougher business conferencing clients that want perfect audio. But these are production deployed and generate revenue. All these are on Ubuntu 10.04 official amis with no mods to kernels. I'm sure there are tweaks that can be made, and bare metal solutions that would work a little better. Does anyone have any ideas how to optimize a Ubuntu instance. I would engage in a few hours of consulting is so. The tests below are interesting but above my paygrade to understand what they mean? I'm running them but don't have a clue how to interpret. If you want a simple of real running data, would be glad to run sample tests on these distributed servers and provide back for analysis. Regards, Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Picher Sent: Thursday, November 17, 2011 8:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Timing issues in AWS? Not having anywhere near the same trouble in a Xen Server in my lab... Although test_time is skewed 180 degrees from what it was in AWS.... freeswitch at internal> timer_test 120 10 Avg: 119.945ms Total Time: 1199.727ms 2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120993 2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 118914 2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119919 2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120032 2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 119802 2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 119999 2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119975 2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 120003 2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 119849 2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119971 freeswitch at internal> time_test 600 10 test 1 sleep 600 1 test 2 sleep 600 1 test 3 sleep 600 0 test 4 sleep 600 1 test 5 sleep 600 0 test 6 sleep 600 1 test 7 sleep 600 1 test 8 sleep 600 0 test 9 sleep 600 0 test 10 sleep 600 1 avg 0 [root at openuc bin]# uname -r 2.6.18-274.7.1.el5 [root at openuc bin]# grep CONFIG_HZ /boot/config-* /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000 Thoughts as to why AWS results are so different from XenServer? Other than not knowing who else is on the AWS box? Thanks, Mike On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher wrote: m1.large I have a c1.xlarge queued up and ready to test... On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen wrote: Just a simple question, what kind of AWS instance are you running your FreeSWITCH? It makes huge difference. Thanks, Chris On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher wrote: Hi guys, Trying to get to the bottom of some conference bridge issues I'm having with running the system in AWS. We're hearing a bunch of snap-crackle-pops in conference bridges and when I tcpdum on the server itself I see them in the RTP and see RTP timestamp problems. I've run the following: timer_test freeswitch at 127.0.0.1@internal> timer_test 120 10 Avg: 120.004ms Total Time: 1200.315ms 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120006 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 120040 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119976 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120007 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 120001 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 120031 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119996 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 119994 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 120005 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119991 test_time freeswitch at 127.0.0.1@internal> time_test 600 10 test 1 sleep 600 1592 test 2 sleep 600 986 test 3 sleep 600 1018 test 4 sleep 600 980 test 5 sleep 600 1005 test 6 sleep 600 1000 test 7 sleep 600 972 test 8 sleep 600 990 test 9 sleep 600 1006 test 10 sleep 600 994 avg 1054 For kernel: [root at openuc bin]# uname -r 2.6.21.7-2.fc8xen CONFIG_HZ: [root at openuc bin]# grep CONFIG_HZ /boot/config-* /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 are the xenU kernel settings screwing me here? Thanks, Mike -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/426aefed/attachment-0001.html From SPapineni at enghouse.com Thu Nov 17 17:41:40 2011 From: SPapineni at enghouse.com (Papineni, Suneel) Date: Thu, 17 Nov 2011 14:41:40 +0000 Subject: [Freeswitch-users] Issue with mod_diangling during compilation In-Reply-To: <1321470446167-7001513.post@n2.nabble.com> References: <9438D04074E0DE45A49CD76099821272F86BDF@CORP-MAIL-002.edge.local> <1321470446167-7001513.post@n2.nabble.com> Message-ID: <9438D04074E0DE45A49CD76099821272F86ED5@CORP-MAIL-002.edge.local> Hi, Thank you very much for pointing me in right direction. I tried with the new git version, and it worked fine to place a call from Gtalk to one of the extensions registered with FreeSwitch. I am getting other issues, but it is better to post as new thread. Thanks & Regards Suneel -----Original Message----- From: mazilo [mailto:Nabble_01394 at slickdeals.endjunk.com] Sent: 16 November 2011 19:07 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Issue with mod_diangling during compilation Papineni, Suneel wrote: > Unfortunately I am not finding the TLS DLLs' as mentioned in the link. > > I tried to compile whole solution and also compiled iksemel project > and mod_diangling project separately. Still I didn't find the TLS DLLs > in output (Release) folder.I thought starting from git commit f506e19e152e070f04574ad67e282af8691606f5 libdingaling no longer uses GNUTLS but rather OpenSSL. See this http://jira.freeswitch.com/browse/FS-3471 FS-3471 . ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-mod-diangling-during-compilation-tp7001164p7001513.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mpicher at gmail.com Thu Nov 17 18:19:45 2011 From: mpicher at gmail.com (Michael Picher) Date: Thu, 17 Nov 2011 10:19:45 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: <005501cca52f$aa0286c0$fe079440$@com> References: <005501cca52f$aa0286c0$fe079440$@com> Message-ID: Huh... maybe I need to get my own CentOS build up there instead of rely on Rightscale's 64 bit image... That's a whole other ball of wax I was trying to avoid having to figure out to do... Mike On Thu, Nov 17, 2011 at 8:49 AM, Stephen Dame wrote: > I?m running freeswitch on about 40 different m1.small c1.medium?s in AWS > regions, us-east, us-west, eu-west, and asia?. They are used in videoconf > component of BigBlueButton.org . For most part they work great? There are > occasional issues with voip quality but the app is 100% voip, with the BBB > client all browser based. So the conference is subject to every ones local > network connections and most issues ?blamed? on the internet instead of > freeswitch J We also tie in skype and DID direct to improve latency for > some clients. But expectations are set so we meet them.**** > > ** ** > > You probably have tougher business conferencing clients that want perfect > audio. But these are production deployed and generate revenue. All > these are on Ubuntu 10.04 official amis with no mods to kernels. **** > > ** ** > > I?m sure there are tweaks that can be made, and bare metal solutions that > would work a little better? Does anyone have any ideas how to optimize a > Ubuntu instance. I would engage in a few hours of consulting is so.**** > > ** ** > > The tests below are interesting but above my paygrade to understand what > they mean? **** > > ** ** > > I?m running them but don?t have a clue how to interpret. If you want a > simple of real running data, would be glad to run sample tests on these > distributed servers and provide back for analysis.**** > > ** ** > > Regards,**** > > Stephen**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Picher > *Sent:* Thursday, November 17, 2011 8:03 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Timing issues in AWS?**** > > ** ** > > Not having anywhere near the same trouble in a Xen Server in my lab... > > Although test_time is skewed 180 degrees from what it was in AWS.... > > freeswitch at internal> timer_test 120 10**** > > Avg: 119.945ms Total Time: 1199.727ms > > 2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: > samplecount after init: 1 > 2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: > samplecount after first step: 2 > 2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 > sleep 120 120993 > 2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 > sleep 120 118914 > 2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 > sleep 120 119919 > 2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 > sleep 120 120032 > 2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 > sleep 120 119802 > 2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 > sleep 120 119999 > 2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 > sleep 120 119975 > 2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 > sleep 120 120003 > 2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 > sleep 120 119849 > 2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 > sleep 120 119971**** > > > freeswitch at internal> time_test 600 10**** > > test 1 sleep 600 1 > test 2 sleep 600 1 > test 3 sleep 600 0 > test 4 sleep 600 1 > test 5 sleep 600 0 > test 6 sleep 600 1 > test 7 sleep 600 1 > test 8 sleep 600 0 > test 9 sleep 600 0 > test 10 sleep 600 1 > avg 0**** > > > [root at openuc bin]# uname -r**** > > 2.6.18-274.7.1.el5**** > > > [root at openuc bin]# grep CONFIG_HZ /boot/config-***** > > /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000**** > > > Thoughts as to why AWS results are so different from XenServer? Other > than not knowing who else is on the AWS box? > > Thanks, > Mike > > > **** > > On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher > wrote:**** > > m1.large > > I have a c1.xlarge queued up and ready to test...**** > > ** ** > > On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen > wrote:**** > > Just a simple question, what kind of AWS instance are you running your > FreeSWITCH?**** > > It makes huge difference.**** > > Thanks,**** > > Chris**** > > On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher > wrote:**** > > Hi guys, > > Trying to get to the bottom of some conference bridge issues I'm having > with running the system in AWS. > > We're hearing a bunch of snap-crackle-pops in conference bridges and when > I tcpdum on the server itself I see them in the RTP and see RTP timestamp > problems. > > I've run the following: > > > timer_test**** > > freeswitch at 127.0.0.1@internal> timer_test 120 10**** > > Avg: 120.004ms Total Time: 1200.315ms**** > > ** ** > > 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: > samplecount after init: 1 > freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] > mod_commands.c:466 Timer Test: samplecount after first step: 2 > 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 > sleep 120 120006 > 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 > sleep 120 120040 > 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 > sleep 120 119976 > 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 > sleep 120 120007 > 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 > sleep 120 120001 > 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 > sleep 120 120031 > 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 > sleep 120 119996 > 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 > sleep 120 119994 > 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 > sleep 120 120005 > 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 > sleep 120 119991**** > > > test_time**** > > freeswitch at 127.0.0.1@internal> time_test 600 10**** > > > test 1 sleep 600 1592 > test 2 sleep 600 986 > test 3 sleep 600 1018 > test 4 sleep 600 980 > test 5 sleep 600 1005 > test 6 sleep 600 1000 > test 7 sleep 600 972 > test 8 sleep 600 990 > test 9 sleep 600 1006 > test 10 sleep 600 994 > avg 1054**** > > > For kernel:**** > > [root at openuc bin]# uname -r > > 2.6.21.7-2.fc8xen**** > > > CONFIG_HZ:**** > > [root at openuc bin]# grep CONFIG_HZ /boot/config-* > > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000**** > > are the xenU kernel settings screwing me here? > > Thanks, > Mike > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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**** > > > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info**** > > > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/b0d67b5e/attachment-0001.html From jmoran at secureachsystems.com Thu Nov 17 18:26:52 2011 From: jmoran at secureachsystems.com (Jason Moran) Date: Thu, 17 Nov 2011 10:26:52 -0500 Subject: [Freeswitch-users] File playback delay References: <361E98F99D3CC3439EED59BC1924ED6958A158@SERVER2003.SecuReachSystems.local> Message-ID: <361E98F99D3CC3439EED59BC1924ED6958A21E@SERVER2003.SecuReachSystems.local> Anyone? Bueller? The files are already in mono ulaw wav format, so I don't think it's transcoding the files for phone playback. From: Jason Moran Sent: Monday, November 14, 2011 12:47 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] File playback delay I have found that the playback of a file can be delayed to the point that the callee will either hangup (thinking it MUST be over now since it has been quiet for 7-8 seconds) or a Voicemail's silence detection will trigger even though I hadn't finished playing back all of the audio files I had intended to play. Ideas: Does session.stream block (spidermonkey script)? Since I am doing several simultaneous calls if session.stream blocked file access that could account for it (although that would be a terrible calamity if true). Does anybody else have ideas? I have been using pcapsipdump to "record" the calls and I can confirm that the reports are true - in some cases the playback of the next audio file can delay for a long time. I have tested these calls to my mobile phone and office phone without any delays present, but it seems to happen to some people some of the time. Thanks, Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/38f76f98/attachment.html From x.liu at hw.ac.uk Thu Nov 17 18:30:59 2011 From: x.liu at hw.ac.uk (xl127) Date: Thu, 17 Nov 2011 15:30:59 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: References: <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> <4EC3D5AD.5070507@hw.ac.uk> <4EC3EA04.7030605@hw.ac.uk> <4EC50BF3.1070905@hw.ac.uk> Message-ID: <4EC528B3.5090306@hw.ac.uk> Yeah, you suggested to use a silence prompt. Sorry I didn't know the format to specify it and didn't realise there is an existing specific way to specify it. Now I know :-) Thanks! Xing On 17/11/11 13:37, Christopher Rienzo wrote: > > > > After some more tests, I found followings: > > 1. Testing from dialplan, the log output is the string of "CRIT > ${speech_detect_result}" rather than the recognition results. > > > oops > > > 2. Regarding to non barge in, I first send "speak" command then > send play_and_detect_speech with parameter: > detect:unimrcp:nuance5-mrcp1-1 > {start-input-timers=false,no-input-timeout=25000,recognition-timeout=25000}dudeYNNC_Nuance > > in which I removed the "say:" part. > > But very soon I received event CHANNEL_EXECUTE_COMPLETE for > play_and_detect_speech before the text playing finishes, > and speech_detect_result is null (actually the event header > does not contain this variable) > > I tried with "say:" part with empty text like > "say:unimrcp:en-GB: " but it doesn't work, see issue 3 below. > > > Try with silence as I originally suggested: > > silence_stream://1000 detect:unimrcp:nuance5-mrcp1-1 > {start-input-timers=false,no-input-timeout=25000, > recognition-timeout=25000}dudeYNNC_Nuance > > > 3. It seems that I can not send "speak" command twice before the > first one finishes. In my case I run one port TTS server on one > machine. > > If I send the command twice FS will give me Synthesizer > Error/Invalid TTS Module. > I thought the TTS request would be queued rather than it > immediately looks for the TTS resource. > > If I send the second command to another TTS machine, no error > occurs but I can only hear one utterance being spoken, > it looks like one utteraance was dropped somehow. > > > Wait for speak to finish before starting a new one. > > > On 16/11/11 16:51, xl127 wrote: >> Hi Christopher, >> >> The questions are cleared to me now. Many thanks for your >> explanations! >> >> Best regards, >> >> Xing >> >> >> On 16/11/11 15:52, Christopher Rienzo wrote: >>> >>> Responses inline >>> >>> >>> Now it works in my ESL app though I am just able to do one >>> dialogue ( I need to add the event catching for furthur >>> dialgoues). >>> >>> I have a couple of questions here: >>> >>> 1. In the first try, my Nuance server was able to be >>> accessed somehow (FS says the MRCP is not responding in 5000ms, >>> something like that), then FS says: [WARNING] >>> rtsp_client.c:386 () Failed to Connect to RTSP Server >>> *MailScanner has detected a possible fraud attempt from >>> "99.185.85.31:554" claiming to be* *MailScanner warning: >>> numerical links are often malicious:* 99.185.85.31:554 >>> , >>> >>> later FS says: >>> [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER channel error! >>> [ERR] switch_ivr_play_say.c:2439 Invalid TTS module! >>> >>> The SYNTHESIZER channel error and Invalid TTS module >>> error are obvious. >>> >>> What I don't understand is why it went to this stange >>> address: *MailScanner has detected a possible fraud attempt >>> from "99.185.85.31:554" claiming to be* *MailScanner >>> warning: numerical links are often malicious:* >>> 99.185.85.31:554 ? >>> >>> >>> check your unimrcp configuration. Make sure the default TTS and >>> ASR profiles are set to actual servers. >>> >>> 2. I specified TTS engine in play_and_detect_speech as >>> "say:unimrcp:nuance5-mrcp1-1: the text to speak" >>> It works though I didn't specify the TTS voice. >>> >>> How do I specify the TTS voice? In the mrcp profile >>> (how?)? or something like: >>> "say:unimrcp:nuance5-mrcp1-1:Serena: the text to >>> speak" (this seems not right.) >>> >>> >>> That won't work. Set the tts_engine variable as I explained >>> previously, or use say:unimrcp:voice:text to speak with the >>> desired voice and the correct default TTS profile defined in >>> unimrcp.conf.xml. This is a limitation of the say: notation. >>> Alternatively, the voice can be defined with the tts_voice >>> channel variable. >>> >>> 3. The barge-in works well, thanks!. Is the barge-in >>> configurable? In some scenarios, we might not allow barge-in. >>> >>> >>> If you don't want to barge in, just do "playback (or speak)" >>> first, then "play_and_detect_speech" with a silence prompt. >>> >>> >>> 4. How could I get the text which has spoken to the user >>> when barge-in occurs? >>> Or Could I get the time when barge-in occurs? If I know >>> the barge-in time and rough totale time for the whole text >>> to be spoken I can figure out the spoken text by >>> manually checking the recorded audio file later, which would >>> be painful. >>> >>> >>> If this is necessary, you might want to use the lower-level >>> functions instead to watch for the begin-speaking event. >>> >>> >>> 5. when I use "speak" and "detect_speech" apps in ESL, I >>> can catch event: DETECTED_SPEECH and speech-type: begin-speaking >>> and "detected-speech", then I do the recognition >>> results processing. >>> >>> The new app play_and_detect_speech seems not generate >>> these events any more. The way that I can think of to get >>> the results >>> is to catch event:CHANNEL_EXECUTE_COMPLETE then check if >>> variable_current_application=play_and_detect_speech, then get >>> the results from variable_detect_speech_result. >>> >>> Is this the proper way to get the results in ESL app? Or >>> will play_and_detect_speech later on be consistent with >>> detect_speech >>> in term of ASR events? >>> >>> >>> play_and_detect_speech is a higher level abstraction to simplify >>> things. If you want to have more control, go back to using the >>> ESL events. Reading the code in mod_dptools and >>> switch_ivr_async will give you hints about how to do it correctly. >>> >>> >>> 6. I'd like to set start-input-timers=false in the initial >>> request then start the recognition timers >>> (start-input-timers=true) >>> after the TTS finishes. >>> How possibly could I do this? >>> >>> >>> This is automatically done in the >>> switch_ivr_play_and_detect_speech() function. You just need to >>> specify start-input-timers=false in the beginning. >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> >> >> ------------------------------------------------------------------------ >> >> Scottish University of the Year 2011-12 *Heriot-Watt University >> is the Sunday Times >> Scottish University of the Year 2011-2012* >> >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > ------------------------------------------------------------------------ > > MailScanner Signature HW *Heriot-Watt University is the Sunday Times > > Scottish University of the Year 2011-2012 > * > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/8b4917cb/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/8b4917cb/attachment-0002.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/8b4917cb/attachment-0003.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/8b4917cb/attachment-0001.jpg From cmrienzo at gmail.com Thu Nov 17 18:48:02 2011 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Thu, 17 Nov 2011 10:48:02 -0500 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper> <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> <4EC3D5AD.5070507@hw.ac.uk> <4EC3EA04.7030605@hw.ac.uk> <4EC50BF3.1070905@hw.ac.uk> Message-ID: Actually in that silence prompt suggestion I gave, use {start-input-timers=true} On Thu, Nov 17, 2011 at 8:37 AM, Christopher Rienzo wrote: > > > >> After some more tests, I found followings: >> >> 1. Testing from dialplan, the log output is the string of "CRIT >> ${speech_detect_result}" rather than the recognition results. >> > > oops > > >> >> 2. Regarding to non barge in, I first send "speak" command then send >> play_and_detect_speech with parameter: >> detect:unimrcp:nuance5-mrcp1-1 >> {start-input-timers=false,no-input-timeout=25000,recognition-timeout=25000}dudeYNNC_Nuance >> >> in which I removed the "say:" part. >> >> But very soon I received event CHANNEL_EXECUTE_COMPLETE for >> play_and_detect_speech before the text playing finishes, >> and speech_detect_result is null (actually the event header does >> not contain this variable) >> >> I tried with "say:" part with empty text like "say:unimrcp:en-GB: " >> but it doesn't work, see issue 3 below. >> > > Try with silence as I originally suggested: > > silence_stream://1000 detect:unimrcp:nuance5-mrcp1-1 > {start-input-timers=false,no-input-timeout=25000, > recognition-timeout=25000}dudeYNNC_Nuance > >> >> 3. It seems that I can not send "speak" command twice before the first >> one finishes. In my case I run one port TTS server on one machine. >> >> If I send the command twice FS will give me Synthesizer >> Error/Invalid TTS Module. >> I thought the TTS request would be queued rather than it immediately >> looks for the TTS resource. >> >> If I send the second command to another TTS machine, no error occurs >> but I can only hear one utterance being spoken, >> it looks like one utteraance was dropped somehow. >> > > Wait for speak to finish before starting a new one. > > >> >> On 16/11/11 16:51, xl127 wrote: >> >> Hi Christopher, >> >> The questions are cleared to me now. Many thanks for your explanations! >> >> Best regards, >> >> Xing >> >> >> On 16/11/11 15:52, Christopher Rienzo wrote: >> >> >> Responses inline >> >> >> Now it works in my ESL app though I am just able to do one dialogue ( >>> I need to add the event catching for furthur dialgoues). >>> >>> I have a couple of questions here: >>> >>> 1. In the first try, my Nuance server was able to be accessed somehow >>> (FS says the MRCP is not responding in 5000ms, >>> something like that), then FS says: [WARNING] rtsp_client.c:386 () >>> Failed to Connect to RTSP Server *MailScanner has detected a possible >>> fraud attempt from "99.185.85.31:554" claiming to be* *MailScanner >>> warning: numerical links are often malicious:* 99.185.85.31:554 >>> , >>> >>> later FS says: >>> [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER channel error! >>> [ERR] switch_ivr_play_say.c:2439 Invalid TTS module! >>> >>> The SYNTHESIZER channel error and Invalid TTS module error are >>> obvious. >>> >>> What I don't understand is why it went to this stange address: *MailScanner >>> has detected a possible fraud attempt from "99.185.85.31:554" claiming to be >>> * *MailScanner warning: numerical links are often malicious:*99.185.85.31:554 >>> ? >>> >> >> check your unimrcp configuration. Make sure the default TTS and ASR >> profiles are set to actual servers. >> >> >>> 2. I specified TTS engine in play_and_detect_speech as >>> "say:unimrcp:nuance5-mrcp1-1: the text to speak" >>> It works though I didn't specify the TTS voice. >>> >>> How do I specify the TTS voice? In the mrcp profile (how?)? or >>> something like: >>> "say:unimrcp:nuance5-mrcp1-1:Serena: the text to speak" (this >>> seems not right.) >>> >> >> That won't work. Set the tts_engine variable as I explained previously, >> or use say:unimrcp:voice:text to speak with the desired voice and the >> correct default TTS profile defined in unimrcp.conf.xml. This is a >> limitation of the say: notation. Alternatively, the voice can be defined >> with the tts_voice channel variable. >> >> >> >>> 3. The barge-in works well, thanks!. Is the barge-in configurable? In >>> some scenarios, we might not allow barge-in. >>> >> >> If you don't want to barge in, just do "playback (or speak)" first, then >> "play_and_detect_speech" with a silence prompt. >> >> >>> >>> 4. How could I get the text which has spoken to the user when barge-in >>> occurs? >>> Or Could I get the time when barge-in occurs? If I know the >>> barge-in time and rough totale time for the whole text >>> to be spoken I can figure out the spoken text by manually checking >>> the recorded audio file later, which would be painful. >>> >> >> If this is necessary, you might want to use the lower-level functions >> instead to watch for the begin-speaking event. >> >> >>> >>> 5. when I use "speak" and "detect_speech" apps in ESL, I can catch >>> event: DETECTED_SPEECH and speech-type: begin-speaking >>> and "detected-speech", then I do the recognition results processing. >>> >>> The new app play_and_detect_speech seems not generate these events >>> any more. The way that I can think of to get the results >>> is to catch event:CHANNEL_EXECUTE_COMPLETE then check if >>> variable_current_application=play_and_detect_speech, then get >>> the results from variable_detect_speech_result. >>> >>> Is this the proper way to get the results in ESL app? Or will >>> play_and_detect_speech later on be consistent with detect_speech >>> in term of ASR events? >>> >> >> play_and_detect_speech is a higher level abstraction to simplify things. >> If you want to have more control, go back to using the ESL events. Reading >> the code in mod_dptools and switch_ivr_async will give you hints about how >> to do it correctly. >> >> >>> >>> 6. I'd like to set start-input-timers=false in the initial request >>> then start the recognition timers (start-input-timers=true) >>> after the TTS finishes. >>> How possibly could I do this? >>> >> >> This is automatically done in the switch_ivr_play_and_detect_speech() >> function. You just need to specify start-input-timers=false in the >> beginning. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> ------------------------------ >> >> [image: Scottish University of the Year 2011-12] *Heriot-Watt >> University is the Sunday Times >> Scottish University of the Year 2011-2012* >> >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> ------------------------------ >> >> [image: MailScanner Signature HW] *Heriot-Watt University is the >> Sunday Times >> >> Scottish University of the Year 2011-2012 >> * >> >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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... 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Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/031b09bd/attachment-0003.jpe From sdame at 207me.com Thu Nov 17 19:03:13 2011 From: sdame at 207me.com (Stephen Dame) Date: Thu, 17 Nov 2011 11:03:13 -0500 Subject: [Freeswitch-users] Speex Codec Settings Message-ID: <00dc01cca542$69a8da80$3cfa8f80$@com> I'm working on a radio station integration project, using speex 16000 into a 16000 conference with x-lite/bria and flash/red5 app. Trying to figure out how the to change the speex codec, and change Complexity.,Quality settings,VBR, CBR etc. I want to run high/quality very small <5 participant, talk show conferences. I have a client app sending qualityencode=10 to conference, but assume its converted to PCM, mixed with other streams and sent back out to other participants at the freeswitch default settings. I looked into speex source code and see some values Complexity 5, Quality 5, VBRQuality 4, AGClevel 8000 Question: Can I somehow pass these values into the call to setup the highest and force CBR, or do I need to change in codec source and recompile? It a dedicated server used for just this application. Thanks in advance Stephen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/7978d936/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 145 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/7978d936/attachment.gif From SPapineni at enghouse.com Thu Nov 17 19:15:28 2011 From: SPapineni at enghouse.com (Papineni, Suneel) Date: Thu, 17 Nov 2011 16:15:28 +0000 Subject: [Freeswitch-users] Joining a call from Gtalk to a conference on FreeSwitch Message-ID: <9438D04074E0DE45A49CD76099821272F86F86@CORP-MAIL-002.edge.local> Hi, I got the new GIT version and enabled mod_dingling and compiled. Everything went through and able to establish call to an extension if I configure that extension number in "client" profile. What I am trying to do is, I want to bridge or join a call coming from GTalk to an existing conference in FreeSwitch. For this purpose I configured a different number on "client profile" and created a dial-plan for this number to 'park' the call first before trying to join to the conference. Then using eventSockets I am trying to join this call to conference and issued following command. (tried with "uuid_bridge" command as well) "api uuid_transfer [Unique-ID] conference:xyz at default inline" Command is successful and also I can hear a sound that someone joined in the conference, but I didn't hear any voice at either side. I couldn't see any RTP flow as well (checked wireshark traces at FS). After sometime like 30 seconds call at GTalk is disconnected automatically. I am not sure why nothing is heard at both sides and why call got disconnected. Also tried answering the call first (after Park) and then bridging to conference, still got the same issue. Could someone please let me know if I am missing anything or need to configure in a different way for conferencing. Thanks & Regards Suneel Client.xml Dial-plan.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/c4e75b67/attachment.html From Hector.Geraldino at ip-soft.net Thu Nov 17 19:19:25 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Thu, 17 Nov 2011 11:19:25 -0500 Subject: [Freeswitch-users] calling limit_usage via ESL In-Reply-To: References: <6A6B4C284AD15042B429EB9D904544AD0224D44767@NY1-EXMB-01.ip-soft.net> Message-ID: <6A6B4C284AD15042B429EB9D904544AD0224D44826@NY1-EXMB-01.ip-soft.net> What about: "execute" + command + " " + arguments From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Wednesday, November 16, 2011 6:20 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] calling limit_usage via ESL The wiki says it's an API also. I have frequently executed it directly in fs_cli. When I try it without "api" at the start, it hangs. -Avi On Thu, Nov 17, 2011 at 1:00 AM, Hector Geraldino > wrote: I think that, if it's an application (i.e. can be executed directly from the dialplan), you shouldn't use the 'api' prefix. I'm not familiar with the php esl module, but in the Java ESL there's a difference when you try to call an api command vs and application: API: "api " + command + " " + arguments APP: SendMsg message = new SendMsg(uuid); message.addCallCommand("execute"); message.addExecuteAppName(appName); message.addExecuteAppArg(arguments); Also, you can always inspect the execution result if you're executing the command in sync mode. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Wednesday, November 16, 2011 5:41 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] calling limit_usage via ESL I'm using the php esl implementation in fusionpbx.. I'm able to pass "api show channels count" and "api show calls count" and the like, but when I do: "api limit_usage hash origination usa_pd" it's just blank. Pasting "limit_usage hash origination usa_pd" into fs_cli yields 0 (or higher as the case may be). There DOES seem to be a \n returned, but nothing else. I've got no 'friggin clue. Help, please? Thanks, Avi _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111117/10cce0f7/attachment-0001.html From x.liu at hw.ac.uk Thu Nov 17 19:21:30 2011 From: x.liu at hw.ac.uk (xl127) Date: Thu, 17 Nov 2011 16:21:30 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: References: <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com> <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> <4EC3D5AD.5070507@hw.ac.uk> <4EC3EA04.7030605@hw.ac.uk> <4EC50BF3.1070905@hw.ac.uk> Message-ID: <4EC5348A.6030409@hw.ac.uk> Yeah, I see, it should be set to true there. Thanks for reminding me this. On 17/11/11 15:48, Christopher Rienzo wrote: > Actually in that silence prompt suggestion I gave, use > {start-input-timers=true} > > > On Thu, Nov 17, 2011 at 8:37 AM, Christopher Rienzo > > wrote: > > > > > After some more tests, I found followings: > > 1. Testing from dialplan, the log output is the string of > "CRIT ${speech_detect_result}" rather than the recognition > results. > > > oops > > > 2. Regarding to non barge in, I first send "speak" command > then send play_and_detect_speech with parameter: > detect:unimrcp:nuance5-mrcp1-1 > {start-input-timers=false,no-input-timeout=25000,recognition-timeout=25000}dudeYNNC_Nuance > > in which I removed the "say:" part. > > But very soon I received event CHANNEL_EXECUTE_COMPLETE > for play_and_detect_speech before the text playing finishes, > and speech_detect_result is null (actually the event > header does not contain this variable) > > I tried with "say:" part with empty text like > "say:unimrcp:en-GB: " but it doesn't work, see issue 3 below. > > > Try with silence as I originally suggested: > > silence_stream://1000 detect:unimrcp:nuance5-mrcp1-1 > {start-input-timers=false,no-input-timeout=25000, > recognition-timeout=25000}dudeYNNC_Nuance > > > 3. It seems that I can not send "speak" command twice before > the first one finishes. In my case I run one port TTS server > on one machine. > > If I send the command twice FS will give me Synthesizer > Error/Invalid TTS Module. > I thought the TTS request would be queued rather than it > immediately looks for the TTS resource. > > If I send the second command to another TTS machine, no > error occurs but I can only hear one utterance being spoken, > it looks like one utteraance was dropped somehow. > > > Wait for speak to finish before starting a new one. > > > On 16/11/11 16:51, xl127 wrote: >> Hi Christopher, >> >> The questions are cleared to me now. Many thanks for your >> explanations! >> >> Best regards, >> >> Xing >> >> >> On 16/11/11 15:52, Christopher Rienzo wrote: >>> >>> Responses inline >>> >>> >>> Now it works in my ESL app though I am just able to do >>> one dialogue ( I need to add the event catching for >>> furthur dialgoues). >>> >>> I have a couple of questions here: >>> >>> 1. In the first try, my Nuance server was able to be >>> accessed somehow (FS says the MRCP is not responding in >>> 5000ms, >>> something like that), then FS says: [WARNING] >>> rtsp_client.c:386 () Failed to Connect to RTSP Server >>> *MailScanner has detected a possible fraud attempt from >>> "99.185.85.31:554" claiming to be* *MailScanner warning: >>> numerical links are often malicious:* 99.185.85.31:554 >>> , >>> >>> later FS says: >>> [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER >>> channel error! >>> [ERR] switch_ivr_play_say.c:2439 Invalid TTS module! >>> >>> The SYNTHESIZER channel error and Invalid TTS module >>> error are obvious. >>> >>> What I don't understand is why it went to this >>> stange address: *MailScanner has detected a possible >>> fraud attempt from "99.185.85.31:554" claiming to be* >>> *MailScanner warning: numerical links are often >>> malicious:* 99.185.85.31:554 ? >>> >>> >>> check your unimrcp configuration. Make sure the default TTS >>> and ASR profiles are set to actual servers. >>> >>> 2. I specified TTS engine in play_and_detect_speech as >>> "say:unimrcp:nuance5-mrcp1-1: the text to speak" >>> It works though I didn't specify the TTS voice. >>> >>> How do I specify the TTS voice? In the mrcp profile >>> (how?)? or something like: >>> "say:unimrcp:nuance5-mrcp1-1:Serena: the text >>> to speak" (this seems not right.) >>> >>> >>> That won't work. Set the tts_engine variable as I explained >>> previously, or use say:unimrcp:voice:text to speak with the >>> desired voice and the correct default TTS profile defined in >>> unimrcp.conf.xml. This is a limitation of the say: >>> notation. Alternatively, the voice can be defined with the >>> tts_voice channel variable. >>> >>> 3. The barge-in works well, thanks!. Is the barge-in >>> configurable? In some scenarios, we might not allow >>> barge-in. >>> >>> >>> If you don't want to barge in, just do "playback (or speak)" >>> first, then "play_and_detect_speech" with a silence prompt. >>> >>> >>> 4. How could I get the text which has spoken to the >>> user when barge-in occurs? >>> Or Could I get the time when barge-in occurs? If I >>> know the barge-in time and rough totale time for the >>> whole text >>> to be spoken I can figure out the spoken text by >>> manually checking the recorded audio file later, which >>> would be painful. >>> >>> >>> If this is necessary, you might want to use the lower-level >>> functions instead to watch for the begin-speaking event. >>> >>> >>> 5. when I use "speak" and "detect_speech" apps in ESL, >>> I can catch event: DETECTED_SPEECH and speech-type: >>> begin-speaking >>> and "detected-speech", then I do the recognition >>> results processing. >>> >>> The new app play_and_detect_speech seems not >>> generate these events any more. The way that I can think >>> of to get the results >>> is to catch event:CHANNEL_EXECUTE_COMPLETE then >>> check if >>> variable_current_application=play_and_detect_speech, >>> then get >>> the results from variable_detect_speech_result. >>> >>> Is this the proper way to get the results in ESL >>> app? Or will play_and_detect_speech later on be >>> consistent with detect_speech >>> in term of ASR events? >>> >>> >>> play_and_detect_speech is a higher level abstraction to >>> simplify things. If you want to have more control, go back >>> to using the ESL events. Reading the code in mod_dptools >>> and switch_ivr_async will give you hints about how to do it >>> correctly. >>> >>> >>> 6. I'd like to set start-input-timers=false in the >>> initial request then start the recognition timers >>> (start-input-timers=true) >>> after the TTS finishes. >>> How possibly could I do this? >>> >>> >>> This is automatically done in the >>> switch_ivr_play_and_detect_speech() function. You just need >>> to specify start-input-timers=false in the beginning. >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> >> >> ------------------------------------------------------------------------ >> >> Scottish University of the Year 2011-12 *Heriot-Watt >> University is the Sunday Times >> Scottish University of the Year 2011-2012* >> >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > ------------------------------------------------------------------------ > > MailScanner Signature HW *Heriot-Watt University is the Sunday > Times > > Scottish University of the Year 2011-2012 > * > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/f7888131/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/f7888131/attachment-0002.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/f7888131/attachment-0003.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/f7888131/attachment-0001.jpg From cjbujold at accra.ca Thu Nov 17 20:29:26 2011 From: cjbujold at accra.ca (Charles Bujold) Date: Thu, 17 Nov 2011 13:29:26 -0400 Subject: [Freeswitch-users] How to Update external IP address on Ubuntu? Message-ID: <000c01cca54e$74db1a10$5e914e30$@accra.ca> Hi, We are newbies to Ubuntu and Freeswitch. We have Freeswitch installed on Ubuntu with Auto-Nat and it works great. The issue we have is that our ISP only provides Dynamic IP's which do not change that often but when they do everyone goes into panic mode until we remember that by simply re-booting the server everything comes back to normal. My question: Is there a method we can put in place that would automate the external-IP update without having to reboot the server. Can Freeswitch notice that the external IP has change and reset itself without manual intervention? And if we need to reboot the server is there a program that would do this automatically when the external IP changes? Thanks cjb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/3ae8c923/attachment.html From brad at tech21.com Thu Nov 17 20:51:44 2011 From: brad at tech21.com (Brad Mina) Date: Thu, 17 Nov 2011 09:51:44 -0800 Subject: [Freeswitch-users] How to Update external IP address on Ubuntu? In-Reply-To: <000c01cca54e$74db1a10$5e914e30$@accra.ca> References: <000c01cca54e$74db1a10$5e914e30$@accra.ca> Message-ID: You can set your external RTP/SIP IP addresses in your sip profile configurations to a STUN server, that way the lookup happens whenever a SIP/RTP transaction takes place. On Thu, Nov 17, 2011 at 9:29 AM, Charles Bujold wrote: > Hi,**** > > ** ** > > We are newbies to Ubuntu and Freeswitch. We have Freeswitch installed on > Ubuntu with Auto-Nat and it works great. The issue we have is that our ISP > only provides Dynamic IP?s which do not change that often but when they do > everyone goes into panic mode until we remember that by simply re-booting > the server everything comes back to normal.**** > > ** ** > > My question: Is there a method we can put in place that would automate the > external-IP update without having to reboot the server. Can Freeswitch > notice that the external IP has change and reset itself without manual > intervention? And if we need to reboot the server is there a program that > would do this automatically when the external IP changes?**** > > ** ** > > Thanks**** > > cjb**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/8ad62f6a/attachment.html From cjbujold at accra.ca Thu Nov 17 21:04:29 2011 From: cjbujold at accra.ca (Charles Bujold) Date: Thu, 17 Nov 2011 14:04:29 -0400 Subject: [Freeswitch-users] How to Update external IP address on Ubuntu? In-Reply-To: References: <000c01cca54e$74db1a10$5e914e30$@accra.ca> Message-ID: <002301cca553$59dbafe0$0d930fa0$@accra.ca> Can you suggest a STUN server to use? Thanks From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brad Mina Sent: November-17-11 1:52 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Update external IP address on Ubuntu? You can set your external RTP/SIP IP addresses in your sip profile configurations to a STUN server, that way the lookup happens whenever a SIP/RTP transaction takes place. On Thu, Nov 17, 2011 at 9:29 AM, Charles Bujold wrote: Hi, We are newbies to Ubuntu and Freeswitch. We have Freeswitch installed on Ubuntu with Auto-Nat and it works great. The issue we have is that our ISP only provides Dynamic IP?s which do not change that often but when they do everyone goes into panic mode until we remember that by simply re-booting the server everything comes back to normal. My question: Is there a method we can put in place that would automate the external-IP update without having to reboot the server. Can Freeswitch notice that the external IP has change and reset itself without manual intervention? And if we need to reboot the server is there a program that would do this automatically when the external IP changes? Thanks cjb _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111117/53647d04/attachment.html From brad at tech21.com Thu Nov 17 21:10:00 2011 From: brad at tech21.com (Brad Mina) Date: Thu, 17 Nov 2011 10:10:00 -0800 Subject: [Freeswitch-users] How to Update external IP address on Ubuntu? In-Reply-To: <002301cca553$59dbafe0$0d930fa0$@accra.ca> References: <000c01cca54e$74db1a10$5e914e30$@accra.ca> <002301cca553$59dbafe0$0d930fa0$@accra.ca> Message-ID: stun.counterpath.com I find counterpath's STUN server works well enough, they have the setting on by default with most of the software, and I haven't had a time where that was ever down. There might be better choices, but this is what I use. On Thu, Nov 17, 2011 at 10:04 AM, Charles Bujold wrote: > Can you suggest a STUN server to use?**** > > ** ** > > Thanks**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brad Mina > *Sent:* November-17-11 1:52 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Update external IP address on > Ubuntu?**** > > ** ** > > You can set your external RTP/SIP IP addresses in your sip profile > configurations to a STUN server, that way the lookup happens whenever a > SIP/RTP transaction takes place.**** > > On Thu, Nov 17, 2011 at 9:29 AM, Charles Bujold wrote: > **** > > Hi,**** > > **** > > We are newbies to Ubuntu and Freeswitch. We have Freeswitch installed on > Ubuntu with Auto-Nat and it works great. The issue we have is that our ISP > only provides Dynamic IP?s which do not change that often but when they do > everyone goes into panic mode until we remember that by simply re-booting > the server everything comes back to normal.**** > > **** > > My question: Is there a method we can put in place that would automate the > external-IP update without having to reboot the server. Can Freeswitch > notice that the external IP has change and reset itself without manual > intervention? And if we need to reboot the server is there a program that > would do this automatically when the external IP changes?**** > > **** > > Thanks**** > > cjb**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/a780036b/attachment-0001.html From pete at privateconnect.com Thu Nov 17 21:18:18 2011 From: pete at privateconnect.com (Pete Mueller) Date: Thu, 17 Nov 2011 11:18:18 -0700 Subject: [Freeswitch-users] =?utf-8?q?mod=5Frtmp_and_a_security_camera?= Message-ID: <20111117111818.2ad02225396a31c9de30536f2e338977.01c4aeeac2.wbe@email13.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/4e4b7987/attachment.html From elliott at zoogmedia.com Thu Nov 17 08:14:39 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Thu, 17 Nov 2011 05:14:39 +0000 Subject: [Freeswitch-users] Polycom phone registration problem Message-ID: Hello, I'm trying to setup freeswitch and I'm having a problem with a polycom phone registering. The phone is behind nat but the freeswitch is on a public network can anyone tell me if I don't have something configured correctly or where should look for the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/4443a529/attachment-0001.html From ahe.sanath at gmail.com Thu Nov 17 11:17:29 2011 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Thu, 17 Nov 2011 13:47:29 +0530 Subject: [Freeswitch-users] Voice mail Call flow change Message-ID: Hi all, I am new to Freeswitch & want to do change voicemail call flow order. For example, when user not available, I need to ask from calling party as "Press1 for leave message or Press 2 for exit."If calling party press 1, then I want to go voicemail call path. I have some idea regarding lang/en/vm/sounds.xml file but no deep understanding how to change call flow. Pls give some example for doing above requirement. Br, Sanath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/d05865fd/attachment.html From ahe.sanath at gmail.com Thu Nov 17 12:50:28 2011 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Thu, 17 Nov 2011 15:20:28 +0530 Subject: [Freeswitch-users] Create User xml file automatically (directory/default/called_no.xml) Message-ID: Hi all, I need to do following in voicemail module of FS. when called number is not available, call is routed to voicemail. I did that using dial plan.But system error coming as "user not exist". When I create directory/default/called_no.xml file manually, it is not coming. I need to create that file automatically for each called number, which is not available. How to do that ? Br, Sanath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/a7a78ccd/attachment.html From ctroncoso at redvoiss.net Thu Nov 17 21:52:27 2011 From: ctroncoso at redvoiss.net (Camila Troncoso) Date: Thu, 17 Nov 2011 15:52:27 -0300 Subject: [Freeswitch-users] a=fmtp missing on FS answer Message-ID: Hello, I?m having some problems with c?dec negotiation and one way audio . I take some Sip captures and I notice that Freeswitch is not inserting * fmtp* parameters in SDP in his 200OK response. I notice that this is a common behavior. I really need that FS generates the complete SDP (with fmtp) because some devices only accept for example G729 annexb=no. I tried with the function sip_append_audio_sdp but it only accepts one a=fmtp line. ? @RFC3264: SDP Offer/Answer Negotiation Generating and processing offers or answers. - "a=fmtp" parameters are not taken into account when generating or processing answer ? Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso at redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/a898e2f9/attachment.html From brian at freeswitch.org Thu Nov 17 22:17:32 2011 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Nov 2011 13:17:32 -0600 Subject: [Freeswitch-users] a=fmtp missing on FS answer In-Reply-To: References: Message-ID: what exactly are you doing and what are the sip traces? What firmware rev? /b On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: > sip_append_audio_sdp -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/62ab733b/attachment.html From anthony.minessale at gmail.com Thu Nov 17 22:29:20 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Nov 2011 13:29:20 -0600 Subject: [Freeswitch-users] a=fmtp missing on FS answer In-Reply-To: References: Message-ID: you probably want to set verbose_sdp=true On Thu, Nov 17, 2011 at 1:17 PM, Brian West wrote: > what exactly are you doing and what are the sip traces? What firmware rev? > > /b > > On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: > > sip_append_audio_sdp > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/8119275d/attachment.html From victor.chukalovskiy at utoronto.ca Thu Nov 17 22:33:19 2011 From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy) Date: Thu, 17 Nov 2011 14:33:19 -0500 Subject: [Freeswitch-users] Issue with: answer confirmation + custom ringback + mod_freetdm In-Reply-To: <4EC152BF.9090201@utoronto.ca> References: <4EC152BF.9090201@utoronto.ca> Message-ID: <4EC5617F.4050804@utoronto.ca> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/5028cfa6/attachment.html From msc at freeswitch.org Thu Nov 17 22:42:14 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Nov 2011 11:42:14 -0800 Subject: [Freeswitch-users] Issue with: answer confirmation + custom ringback + mod_freetdm In-Reply-To: <4EC5617F.4050804@utoronto.ca> References: <4EC152BF.9090201@utoronto.ca> <4EC5617F.4050804@utoronto.ca> Message-ID: Can you pastebin the configuration you are using? Also, get a console log of working and non-working calls. Use "FreeSWITCH Log" as the syntax highlight. (use pastebin.freeswitch.org) Thanks, MC On Thu, Nov 17, 2011 at 11:33 AM, Victor Chukalovskiy < victor.chukalovskiy at utoronto.ca> wrote: > ** > Really looking for advise from someone on this. > > Thank you, > Victor > > On 11/14/2011 12:41 PM, Victor Chukalovskiy wrote: > > Hello, > > I'm looking for some help with the following: an answer confirmation + > custom ringback used to work Ok when placing call out via SIP. Now, when I > place a call out using analog line and mod_freetdm (dahdi mode), the > calling party hears no ringback (dead air) where I expect custom ringback > to be played. > > My successful call scenario: incoming call is answered and custom > ring-back is played all the time while call-hunt tries to find someone who > picks-up the line and presses group_confirm_key. What I like is that my > ring-back is played regardless of what signalling state outbound SIP calls > take all the way until one of them is answered by group_confirm key. > > My failed call scenario: The same, but whenever call goes out through > FreeTDM ringback is no longer played as soon as FreeTDM marks channels as > answered. Dead air is played instead. I'd like to send a custom ringback > instead of this silence regardless of what FreeTDM thinks all the way until > group_confirm key is detected. Instant ringback makes the trick, but > occasionally it triggers fast busy which makes it unusable. > > Thanks for looking, > Victor > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/45238fd0/attachment-0001.html From msc at freeswitch.org Thu Nov 17 22:48:32 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Nov 2011 11:48:32 -0800 Subject: [Freeswitch-users] Polycom phone registration problem In-Reply-To: References: Message-ID: Hi Elliot, IIRC, the scenario where the Polycom is behind NAT and the FS is on a public IP is one where Polycom fails because they are too stubborn to add rport to their SIP handling. (We've been barking at them for over 5 years on this issue.) There is a workaround in FS but it breaks non-Polycom phones that are on the same profile, so you'll need to have a separate profile for your Polycoms if you are in a mixed environment. In any case, for the sofia profile that has the Polycoms you should check out these two parameters: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#NDLB-force-rport http://wiki.freeswitch.org/wiki/NAT_Traversal#NDLB-connectile-dysfunction You might also want to hop into #freeswitch on irc.freenode.net and talk to some of our resident Polycom veterans. -MC On Wed, Nov 16, 2011 at 9:14 PM, Elliott Vogel wrote: > Hello, I?m trying to setup freeswitch and I?m having a problem with a > polycom phone registering. The phone is behind nat but the freeswitch is on > a public network can anyone tell me if I don?t have something configured > correctly or where should look for the problem?**** > > ** ** > > **** > > **** > > **** > > **** > > **** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > *** > * > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > **** > > **** > > ** ** > > ** ** > > ** ** > > **** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > **** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > **** > > **** > > **** > > **** > > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > **** > > **** > > **** > > ** ** > > **** > > ** ** > > **** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > ** ** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/824d5d93/attachment-0001.html From msc at freeswitch.org Thu Nov 17 22:55:38 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Nov 2011 11:55:38 -0800 Subject: [Freeswitch-users] Comfort noise or silence rtp packets during recording In-Reply-To: <4EC33D20.3060604@ezuce.com> References: <4EC33D20.3060604@ezuce.com> Message-ID: Try this: http://wiki.freeswitch.org/wiki/Variable_record_waste_resources -MC On Tue, Nov 15, 2011 at 8:33 PM, Joegen Baclor wrote: > I tried asking in the freeswitch irc channel but got no response. I'm > trying my luck here. > > I have recently received a packet capture from the sipx community that > seems to indicate that when FreeSwitch is recording a voice mail message > via our custom IVR, that it does not send either comfort noise or > silence packets back to the caller. This results to the call getting > dropped by the ITSP after 30 seconds because of RTP time out. Is there a > configurable parameter to make the IVR send some voice activity back? > > Joegen > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/e09a5acb/attachment.html From msc at freeswitch.org Thu Nov 17 22:57:46 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Nov 2011 11:57:46 -0800 Subject: [Freeswitch-users] File playback delay In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6958A21E@SERVER2003.SecuReachSystems.local> References: <361E98F99D3CC3439EED59BC1924ED6958A158@SERVER2003.SecuReachSystems.local> <361E98F99D3CC3439EED59BC1924ED6958A21E@SERVER2003.SecuReachSystems.local> Message-ID: Have you caught this behavior in the act and seen what's up in the logs? I understand it can be challenging to sift through all the log data. You may need to enable uuid logging so that it is easier to filter out the call you want to investigate. -MC On Thu, Nov 17, 2011 at 7:26 AM, Jason Moran wrote: > Anyone? Bueller? The files are already in mono ulaw wav format, so I > don?t think it?s transcoding the files for phone playback.**** > > ** ** > > *From:* Jason Moran > *Sent:* Monday, November 14, 2011 12:47 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] File playback delay**** > > ** ** > > I have found that the playback of a file can be delayed to the point that > the callee will either hangup (thinking it MUST be over now since it has > been quiet for 7-8 seconds) or a Voicemail?s silence detection will trigger > even though I hadn?t finished playing back all of the audio files I had > intended to play.**** > > ** ** > > Ideas: Does session.stream block (spidermonkey script)? Since I am doing > several simultaneous calls if session.stream blocked file access that could > account for it (although that would be a terrible calamity if true).**** > > ** ** > > Does anybody else have ideas? I have been using pcapsipdump to ?record? > the calls and I can confirm that the reports are true ? in some cases the > playback of the next audio file can delay for a long time. I have tested > these calls to my mobile phone and office phone without any delays present, > but it seems to happen to some people some of the time.**** > > ** ** > > Thanks,**** > > Jason**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/246be930/attachment.html From ctroncoso at redvoiss.net Thu Nov 17 23:24:42 2011 From: ctroncoso at redvoiss.net (Camila Troncoso) Date: Thu, 17 Nov 2011 17:24:42 -0300 Subject: [Freeswitch-users] a=fmtp missing on FS answer In-Reply-To: References: Message-ID: <24767b5c4ead7d1bcc6eee30753006f3@mail.gmail.com> I?m trying to make a call trough FS with G729 annexb=no in one side and G729 annexb=no ( the same) in the other. I have a sangoma transcoding card installed and configured, all is working fine. I use this card because in some cases I need transcoding . So I have late-negotiation=true and disable-transcoding=false. The problem is that FS answer without a=fmtp as show in the picture. Freeswitch version is 1.0 [image: sdp-problem.jpg] *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* jueves, 17 de noviembre de 2011 16:18 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer what exactly are you doing and what are the sip traces? What firmware rev? /b On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: sip_append_audio_sdp -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/77f763d4/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2182 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/77f763d4/attachment-0001.png -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 41048 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/77f763d4/attachment-0001.jpe From notlikeme75 at yahoo.com Thu Nov 17 23:48:46 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 17 Nov 2011 12:48:46 -0800 (PST) Subject: [Freeswitch-users] call count bind digit on static conference Message-ID: <1321562926.58272.YahooMailNeo@web65310.mail.ac2.yahoo.com> windows server 2008 32bit; freeswitch windows msi 11/4/2011 thank you to everyone who helped me fix my mod_flite tts issue. I went and? installed licensed cepstral and loaded mod_cepstral and all is fine now but led me to another issue :) I have followed the instructions at http://wiki.freeswitch.org/wiki/Conference_Announce_Count_Inline with the exception that I did not create a? a new"plain" profile i just added the action : to the default profile when i go into a static conference room it speaks tts callie and tells me I am the only caller, which is correct but unfortunately if second callers joins they don't get tts and when i press the 8 digit when two or more callers it shows 2011-11-17 15:23:27.284724 {INFO} mod_dptools.c:1316 Conference Conference not found 2011-11-17 15:23:27.305231 {NOTICE} switch_core_session.c:2367 Execute say(en number pronounced $(conf_count)) 2011-11-17 15:23:27.325737 {ERR} mod_say_en.c:130 Parse Error! I have read this (http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075160.html) but am still stuck, not knowing 1/ how to list correct rooms and 2/ tell my system how to find them using the variable "conference_name" in ?http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075160.html also concerning same issue is that when I do a command "conference list" the output is: Conference conference room 1-127.0.0.1 (1 member rate:8000)13; sofia/external/2064200990 at 66.54.140.446;b11d94a6-94ff-4955-bc87-13e332a7443;unknown;2064200990;hear:speak:floor;0;0;0;50 I try to "conference conference room 1-127.0.0.1" I get response "conference not found" I am sure that all of these issues are fixed with the same method, I just need help in the right direction. your help is appreciated; thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/656c8682/attachment.html From avi at avimarcus.net Fri Nov 18 00:00:45 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 17 Nov 2011 23:00:45 +0200 Subject: [Freeswitch-users] a=fmtp missing on FS answer In-Reply-To: <24767b5c4ead7d1bcc6eee30753006f3@mail.gmail.com> References: <24767b5c4ead7d1bcc6eee30753006f3@mail.gmail.com> Message-ID: Anthony said you probably want this: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp -Avi On Thu, Nov 17, 2011 at 10:24 PM, Camila Troncoso wrote: > I?m trying to make a call trough FS with G729 annexb=no in one side and > G729 annexb=no ( the same) in the other. I have a sangoma transcoding card > installed and configured, all is working fine. I use this card because in > some cases I need transcoding . So I have late-negotiation=true and > disable-transcoding=false. The problem is that FS answer without a=fmtp as > show in the picture. Freeswitch version is 1.0 > > > > [image: sdp-problem.jpg] > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* jueves, 17 de noviembre de 2011 16:18 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer > > > > what exactly are you doing and what are the sip traces? What firmware rev? > > > > /b > > > > On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: > > > > sip_append_audio_sdp > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/79605629/attachment.html From ctroncoso at redvoiss.net Fri Nov 18 00:05:29 2011 From: ctroncoso at redvoiss.net (Camila Troncoso) Date: Thu, 17 Nov 2011 18:05:29 -0300 Subject: [Freeswitch-users] a=fmtp missing on FS answer In-Reply-To: References: <24767b5c4ead7d1bcc6eee30753006f3@mail.gmail.com> Message-ID: <3407b5c8e286e2ae1b9f531520db2a8e@mail.gmail.com> This isnt the solution because it only appends rtpmap not fmtp, I tried it. *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* jueves, 17 de noviembre de 2011 18:01 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer Anthony said you probably want this: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp -Avi On Thu, Nov 17, 2011 at 10:24 PM, Camila Troncoso wrote: I?m trying to make a call trough FS with G729 annexb=no in one side and G729 annexb=no ( the same) in the other. I have a sangoma transcoding card installed and configured, all is working fine. I use this card because in some cases I need transcoding . So I have late-negotiation=true and disable-transcoding=false. The problem is that FS answer without a=fmtp as show in the picture. Freeswitch version is 1.0 *Error! Filename not specified.**Error! Filename not specified.* *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* jueves, 17 de noviembre de 2011 16:18 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer what exactly are you doing and what are the sip traces? What firmware rev? /b On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: sip_append_audio_sdp _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111117/fc7511d3/attachment-0001.html From msc at freeswitch.org Fri Nov 18 00:14:48 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Nov 2011 13:14:48 -0800 Subject: [Freeswitch-users] call count bind digit on static conference In-Reply-To: <1321562926.58272.YahooMailNeo@web65310.mail.ac2.yahoo.com> References: <1321562926.58272.YahooMailNeo@web65310.mail.ac2.yahoo.com> Message-ID: > Conference > conference room 1-127.0.0.1 (1 member rate:8000)13; sofia/external/2064200990 at 66.54.140.446;b11d94a6-94ff-4955-bc87-13e332a7443;unknown;2064200990;hear:speak:floor;0;0;0;50 > > I try to "conference conference room 1-127.0.0.1" > > I get response "conference not found" > > only one "conference" - like this: conference list or conference 1-127.0.0.1 list -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/499bbe09/attachment.html From brian at freeswitch.org Fri Nov 18 00:33:43 2011 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Nov 2011 15:33:43 -0600 Subject: [Freeswitch-users] a=fmtp missing on FS answer In-Reply-To: <3407b5c8e286e2ae1b9f531520db2a8e@mail.gmail.com> References: <24767b5c4ead7d1bcc6eee30753006f3@mail.gmail.com> <3407b5c8e286e2ae1b9f531520db2a8e@mail.gmail.com> Message-ID: <3F2BAC20-F525-4749-BA37-2B6BE7BED67D@freeswitch.org> {sip_append_audio_sdp=a=fmtp:18 annexb=no}sofia/internal/foo at bar.com Or some variation of that? /b On Nov 17, 2011, at 3:05 PM, Camila Troncoso wrote: > This isnt the solution because it only appends rtpmap not fmtp, I tried it. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* jueves, 17 de noviembre de 2011 18:01 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer > > > > Anthony said you probably want this: > > http://wiki.freeswitch.org/wiki/Variable_verbose_sdp > > > -Avi > > > > On Thu, Nov 17, 2011 at 10:24 PM, Camila Troncoso > wrote: > > I?m trying to make a call trough FS with G729 annexb=no in one side and > G729 annexb=no ( the same) in the other. I have a sangoma transcoding card > installed and configured, all is working fine. I use this card because in > some cases I need transcoding . So I have late-negotiation=true and > disable-transcoding=false. The problem is that FS answer without a=fmtp as > show in the picture. Freeswitch version is 1.0 > > > > *Error! Filename not specified.**Error! Filename not specified.* > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* jueves, 17 de noviembre de 2011 16:18 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer > > > > what exactly are you doing and what are the sip traces? What firmware rev? > > > > /b > > > > On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: > > > > sip_append_audio_sdp > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/6f345a1f/attachment.html From avi at avimarcus.net Fri Nov 18 00:54:34 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 17 Nov 2011 23:54:34 +0200 Subject: [Freeswitch-users] calling limit_usage via ESL In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0224D44826@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD0224D44767@NY1-EXMB-01.ip-soft.net> <6A6B4C284AD15042B429EB9D904544AD0224D44826@NY1-EXMB-01.ip-soft.net> Message-ID: no using "execute" it hangs. Can I craft/view ESL stuff raw somewhere other than telnet? I recall hitting ctrl-C and killing freeswitch in telnet.. -Avi On Thu, Nov 17, 2011 at 6:19 PM, Hector Geraldino wrote: > What about: ?"execute" + command + " " + arguments > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi > Marcus > Sent: Wednesday, November 16, 2011 6:20 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] calling limit_usage via ESL > > > > The wiki says it's an API also. I have frequently executed it directly in > fs_cli. > > When I try it without "api" at the start, it hangs. > > -Avi > > > > On Thu, Nov 17, 2011 at 1:00 AM, Hector Geraldino > wrote: > > I think that, if it?s an application (i.e. can be executed directly from the > dialplan), you shouldn?t use the ?api? prefix. > > > > I?m not familiar with the php esl module, but in the Java ESL there?s a > difference when you try to call an api command vs and application: > > > > API: > > ??????????????? "api " + command + " " + arguments > > > > APP: > > SendMsg message = new SendMsg(uuid); > > message.addCallCommand("execute"); > > ??????? ??????? message.addExecuteAppName(appName); > > ??????????????? message.addExecuteAppArg(arguments); > > > > Also, you can always inspect the execution result if you?re executing the > command in sync mode. > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi > Marcus > Sent: Wednesday, November 16, 2011 5:41 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] calling limit_usage via ESL > > > > I'm using the php esl implementation in fusionpbx.. I'm able to pass "api > show channels count" and "api show calls count" and the like, but when I do: > > "api?limit_usage hash origination usa_pd" it's just blank. > > Pasting "limit_usage hash origination usa_pd" into fs_cli yields 0 (or > higher as the case may be). > > > > There DOES seem to be a \n returned, but nothing else. I've got no 'friggin > clue. > > Help, please? > > Thanks, > > Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 Fri Nov 18 01:35:49 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Nov 2011 16:35:49 -0600 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: <005501cca52f$aa0286c0$fe079440$@com> Message-ID: the "amazon linux" works well with modern GIT revisions. older ones need a centos kernel to work well. On Thu, Nov 17, 2011 at 9:19 AM, Michael Picher wrote: > Huh... maybe I need to get my own CentOS build up there instead of rely > on Rightscale's 64 bit image... That's a whole other ball of wax I was > trying to avoid having to figure out to do... > > Mike > > > On Thu, Nov 17, 2011 at 8:49 AM, Stephen Dame wrote: > >> I?m running freeswitch on about 40 different m1.small c1.medium?s in AWS >> regions, us-east, us-west, eu-west, and asia?. They are used in videoconf >> component of BigBlueButton.org . For most part they work great? There are >> occasional issues with voip quality but the app is 100% voip, with the BBB >> client all browser based. So the conference is subject to every ones local >> network connections and most issues ?blamed? on the internet instead of >> freeswitch J We also tie in skype and DID direct to improve latency for >> some clients. But expectations are set so we meet them.**** >> >> ** ** >> >> You probably have tougher business conferencing clients that want perfect >> audio. But these are production deployed and generate revenue. All >> these are on Ubuntu 10.04 official amis with no mods to kernels. **** >> >> ** ** >> >> I?m sure there are tweaks that can be made, and bare metal solutions that >> would work a little better? Does anyone have any ideas how to optimize a >> Ubuntu instance. I would engage in a few hours of consulting is so.**** >> >> ** ** >> >> The tests below are interesting but above my paygrade to understand what >> they mean? **** >> >> ** ** >> >> I?m running them but don?t have a clue how to interpret. If you want a >> simple of real running data, would be glad to run sample tests on these >> distributed servers and provide back for analysis.**** >> >> ** ** >> >> Regards,**** >> >> Stephen**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael >> Picher >> *Sent:* Thursday, November 17, 2011 8:03 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Timing issues in AWS?**** >> >> ** ** >> >> Not having anywhere near the same trouble in a Xen Server in my lab... >> >> Although test_time is skewed 180 degrees from what it was in AWS.... >> >> freeswitch at internal> timer_test 120 10**** >> >> Avg: 119.945ms Total Time: 1199.727ms >> >> 2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: >> samplecount after init: 1 >> 2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: >> samplecount after first step: 2 >> 2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 >> sleep 120 120993 >> 2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 >> sleep 120 118914 >> 2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 >> sleep 120 119919 >> 2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 >> sleep 120 120032 >> 2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 >> sleep 120 119802 >> 2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 >> sleep 120 119999 >> 2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 >> sleep 120 119975 >> 2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 >> sleep 120 120003 >> 2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 >> sleep 120 119849 >> 2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 >> sleep 120 119971**** >> >> >> freeswitch at internal> time_test 600 10**** >> >> test 1 sleep 600 1 >> test 2 sleep 600 1 >> test 3 sleep 600 0 >> test 4 sleep 600 1 >> test 5 sleep 600 0 >> test 6 sleep 600 1 >> test 7 sleep 600 1 >> test 8 sleep 600 0 >> test 9 sleep 600 0 >> test 10 sleep 600 1 >> avg 0**** >> >> >> [root at openuc bin]# uname -r**** >> >> 2.6.18-274.7.1.el5**** >> >> >> [root at openuc bin]# grep CONFIG_HZ /boot/config-***** >> >> /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y >> /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 >> /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y >> /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 >> /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y >> /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 >> /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y >> /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 >> /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y >> /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000**** >> >> >> Thoughts as to why AWS results are so different from XenServer? Other >> than not knowing who else is on the AWS box? >> >> Thanks, >> Mike >> >> >> **** >> >> On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher >> wrote:**** >> >> m1.large >> >> I have a c1.xlarge queued up and ready to test...**** >> >> ** ** >> >> On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen >> wrote:**** >> >> Just a simple question, what kind of AWS instance are you running your >> FreeSWITCH?**** >> >> It makes huge difference.**** >> >> Thanks,**** >> >> Chris**** >> >> On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher >> wrote:**** >> >> Hi guys, >> >> Trying to get to the bottom of some conference bridge issues I'm having >> with running the system in AWS. >> >> We're hearing a bunch of snap-crackle-pops in conference bridges and when >> I tcpdum on the server itself I see them in the RTP and see RTP timestamp >> problems. >> >> I've run the following: >> >> >> timer_test**** >> >> freeswitch at 127.0.0.1@internal> timer_test 120 10**** >> >> Avg: 120.004ms Total Time: 1200.315ms**** >> >> ** ** >> >> 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: >> samplecount after init: 1 >> freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] >> mod_commands.c:466 Timer Test: samplecount after first step: 2 >> 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 >> sleep 120 120006 >> 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 >> sleep 120 120040 >> 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 >> sleep 120 119976 >> 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 >> sleep 120 120007 >> 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 >> sleep 120 120001 >> 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 >> sleep 120 120031 >> 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 >> sleep 120 119996 >> 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 >> sleep 120 119994 >> 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 >> sleep 120 120005 >> 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 >> sleep 120 119991**** >> >> >> test_time**** >> >> freeswitch at 127.0.0.1@internal> time_test 600 10**** >> >> >> test 1 sleep 600 1592 >> test 2 sleep 600 986 >> test 3 sleep 600 1018 >> test 4 sleep 600 980 >> test 5 sleep 600 1005 >> test 6 sleep 600 1000 >> test 7 sleep 600 972 >> test 8 sleep 600 990 >> test 9 sleep 600 1006 >> test 10 sleep 600 994 >> avg 1054**** >> >> >> For kernel:**** >> >> [root at openuc bin]# uname -r >> >> 2.6.21.7-2.fc8xen**** >> >> >> CONFIG_HZ:**** >> >> [root at openuc bin]# grep CONFIG_HZ /boot/config-* >> >> /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y >> /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set >> /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set >> /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 >> /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y >> /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 >> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set >> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set >> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set >> /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y >> /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000**** >> >> are the xenU kernel settings screwing me here? >> >> Thanks, >> Mike >> >> >> -- >> There are 10 kinds of people in this world, those who understand binary >> and those who don't. >> >> mpicher at gmail.com >> blog: http://www.sipxecs.info >> call: sip:mpicher at sipxecs.info**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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**** >> >> ** ** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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**** >> >> >> >> >> -- >> There are 10 kinds of people in this world, those who understand binary >> and those who don't. >> >> mpicher at gmail.com >> blog: http://www.sipxecs.info >> call: sip:mpicher at sipxecs.info**** >> >> >> >> >> -- >> There are 10 kinds of people in this world, those who understand binary >> and those who don't. >> >> mpicher at gmail.com >> blog: http://www.sipxecs.info >> call: sip:mpicher at sipxecs.info**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/0b11349e/attachment-0001.html From spencer at 5ninesolutions.com Fri Nov 18 02:05:34 2011 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 17 Nov 2011 15:05:34 -0800 Subject: [Freeswitch-users] Failed DTMF payload check error Message-ID: Hello all, I'm receiving a few intermittent [ERR] switch_rtp.c:314 Failed DTMF payload check. errors in my logs. Does anyone know what could be causing this and is it something to worry about? I've had no complaints of DTMF not working. Thanks, Spencer From freeswitch-list at puzzled.xs4all.nl Fri Nov 18 02:24:06 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 18 Nov 2011 00:24:06 +0100 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: <005501cca52f$aa0286c0$fe079440$@com> Message-ID: <4EC59796.5040402@puzzled.xs4all.nl> On 11/17/2011 04:19 PM, Michael Picher wrote: > Huh... maybe I need to get my own CentOS build up there instead of rely > on Rightscale's 64 bit image... That's a whole other ball of wax I was > trying to avoid having to figure out to do... Check out boxgrinder.org if you want to put together your own CentOS image. Regards, Patrick From matthew at corp.crocker.com Fri Nov 18 02:39:00 2011 From: matthew at corp.crocker.com (Matthew S. Crocker) Date: Thu, 17 Nov 2011 18:39:00 -0500 (EST) Subject: [Freeswitch-users] Polycom phone registration problem In-Reply-To: Message-ID: <41619b09-5666-468f-a34d-372f10c0063a@zimbra1.crocker.com> Couldn't you do something like if $contact.address != ip.src_addr && sip_agent contains Polycom THEN NDLB-connectile-dysfunction -Matt ----- Original Message ----- > From: "Michael Collins" > To: "FreeSWITCH Users Help" > Sent: Thursday, November 17, 2011 2:48:32 PM > Subject: Re: [Freeswitch-users] Polycom phone registration problem > Hi Elliot, > IIRC, the scenario where the Polycom is behind NAT and the FS is on a > public IP is one where Polycom fails because they are too stubborn > to add rport to their SIP handling. (We've been barking at them for > over 5 years on this issue.) > There is a workaround in FS but it breaks non-Polycom phones that are > on the same profile, so you'll need to have a separate profile for > your Polycoms if you are in a mixed environment. In any case, for > the sofia profile that has the Polycoms you should check out these > two parameters: > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#NDLB-force-rport > http://wiki.freeswitch.org/wiki/NAT_Traversal#NDLB-connectile-dysfunction > You might also want to hop into #freeswitch on irc.freenode.net and > talk to some of our resident Polycom veterans. > -MC > On Wed, Nov 16, 2011 at 9:14 PM, Elliott Vogel < > elliott at zoogmedia.com > wrote: > > Hello, I?m trying to setup freeswitch and I?m having a problem with > > a > > polycom phone registering. The phone is behind nat but the > > freeswitch is on a public network can anyone tell me if I don?t > > have > > something configured correctly or where should look for the > > problem? > > > < include > > > > < domain name = " zoogmedia.com " > > > > < params > > > > < param name = " dial-string " value = " > > {sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})} > > " /> > > > > > > < variables > > > > < variable name = " record_stereo " value = " true " /> > > > < variable name = " default_gateway " value = " > > $${default_provider} > > " /> > > > < variable name = " default_areacode " value = " 312 " /> > > > < variable name = " transfer_fallback_extension " value = " > > operator > > " /> > > > > > > < groups > > > > < group name = " default " > > > > < users > > > > < user id = " 14149823263 " > > > > < params > > > > < param name = " password " value = " 123456 " /> > > > > > > < variables > > > > < variable name = " sip-force-contact " value = " > > NDLB-connectile-dysfunction " /> > > > < variable name = " toll_allow " value = " > > domestic,international,local " /> > > > < variable name = " accountcode " value = " 1000 " /> > > > < variable name = " user_context " value = " default " /> > > > < variable name = " outbound_caller_id_name " value = " Sales " /> > > > < variable name = " outbound_caller_id_number " value = " > > +14149823263 " /> > > > > > > > > > > > > > > > > > > > > > > > > < configuration name = " sofia.conf " description = " sofia > > Endpoint > > " > > > > < global_settings > > > > < param name = " log-level " value = " 0 " /> > > > > > > < param name = " debug-presence " value = " 0 " /> > > > > > > > > > < profiles > > > > < profile name = " endpoints " > > > > < gateways > > > > < gateway name = " TNG " > > > > < param name = " username " value = " user " /> > > > < param name = " password " value = " password " /> > > > < param name = " proxy " value = " 69.25.128.195:5060 " /> > > > < param name = " from-domain " value = " $${local_ip_v4}:5060 " /> > > > < param name = " dtmf-type " value = " rfc2833 " /> > > > < param name = " extension-in-contact " value = " true " /> > > > < param name = " caller-id-in-from " value = " true " /> > > > < param name = " register " value = " false " /> > > > > > > > > > < aliases > > > > > > > < domains > > > > < domain name = " all " alias = " false " parse = " true " /> > > > > > > < settings > > > > < param name = " debug " value = " 0 " /> > > > < param name = " sip-trace " value = " no " /> > > > < param name = " sip-capture " value = " no " /> > > > < param name = " rtp-ip " value = " $${external_rtp_ip} " /> > > > < param name = " sip-ip " value = " $${external_sip_ip} " /> > > > < param name = " ext-rtp-ip " value = " $${external_rtp_ip} " /> > > > < param name = " ext-sip-ip " value = " $${external_sip_ip} " /> > > > < param name = " sip-port " value = " 5070 " /> > > > < param name = " tls " value = " $${external_ssl_enable} " /> > > > < param name = " tls-bind-params " value = " transport=tls " /> > > > < param name = " tls-sip-port " value = " $${external_tls_port} " > > /> > > > < param name = " tls-cert-dir " value = " $${external_ssl_dir} " /> > > > < param name = " tls-version " value = " $${sip_tls_version} " /> > > > < param name = " rfc2833-pt " value = " 101 " /> > > > < param name = " dialplan " value = " XML " /> > > > < param name = " context " value = " default " /> > > > < param name = " dtmf-duration " value = " 2000 " /> > > > < param name = " inbound-codec-prefs " value = " > > $${global_codec_prefs} " /> > > > < param name = " outbound-codec-prefs " value = " > > $${outbound_codec_prefs} " /> > > > < param name = " hold-music " value = " $${hold_music} " /> > > > < param name = " rtp-timer-name " value = " soft " /> > > > < param name = " local-network-acl " value = " localnet.auto " /> > > > < param name = " manage-presence " value = " false " /> > > > < param name = " inbound-codec-negotiation " value = " greedy " /> > > > < param name = " nonce-ttl " value = " 60 " /> > > > < param name = " auth-calls " value = " false " /> > > > < param name = " rtp-timeout-sec " value = " 300 " /> > > > < param name = " rtp-hold-timeout-sec " value = " 1800 " /> > > > < param name = " rtp-rewrite-timestamps " value = " true " /> > > > < param name = " track-calls " value = " true " /> > > > > > > > > > > > > > > > > > > > > > < include > > > > < context name = " default " > > > > < extension name = " unloop " > > > > < condition field = " ${unroll_loops} " expression = " ^true$ " /> > > > < condition field = " ${sip_looped_call} " expression = " ^true$ " > > > > > > < action application = " deflect " data = " ${destination_number} " > > /> > > > > > > > > > < extension name = " check_auth " continue = " true " > > > > < condition field = " ${sip_authorized} " expression = " ^true$ " > > break = " never " > > > > < anti-action application = " respond " data = " 407 " /> > > > > > > > > > < extension name = " local " > > > > < condition field = " ${toll_allow} " expression = " local " /> > > > < condition field = " destination_number " expression = " > > ^([0-9]{7})$ " > > > > < action application = " set " data = " > > effective_caller_id_name=${outbound_caller_id_name} " /> > > > < action application = " set " data = " > > effective_caller_id_number=${outbound_caller_id_number} " /> > > > < action application = " bridge " data = " > > sofia/gateway/TNG/+1${default_areacode}$1 " /> > > > > > > > > > < extension name = " domestic " > > > > < condition field = " ${toll_allow} " expression = " domestic " /> > > > < condition field = " destination_number " expression = " > > ^(1{0,1}\d{10})$ " > > > > < action application = " set " data = " > > effective_caller_id_name=${outbound_caller_id_name} " /> > > > < action application = " set " data = " > > effective_caller_id_number=${outbound_caller_id_number} " /> > > > < action application = " bridge " data = " sofia/gateway/TNG/+$1 " > > /> > > > > > > > > > < extension name = " international " > > > > < condition field = " ${toll_allow} " expression = " international > > " > > /> > > > < condition field = " destination_number " expression = " > > ^011?(\d+)$ > > " > > > > < action application = " set " data = " > > effective_caller_id_name=${outbound_caller_id_name} " /> > > > < action application = " set " data = " > > effective_caller_id_number=${outbound_caller_id_number} " /> > > > < action application = " bridge " data = " sofia/gateway/TNG/+$1 " > > /> > > > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111117/0e6e6581/attachment-0001.html From jbaclor at ezuce.com Fri Nov 18 03:17:08 2011 From: jbaclor at ezuce.com (Joegen Baclor) Date: Fri, 18 Nov 2011 08:17:08 +0800 Subject: [Freeswitch-users] Comfort noise or silence rtp packets during recording In-Reply-To: References: <4EC33D20.3060604@ezuce.com> Message-ID: <4EC5A404.1010609@ezuce.com> Thanks Michael. Exactly what I am looking for. On 11/18/2011 03:55 AM, Michael Collins wrote: > Try this: > http://wiki.freeswitch.org/wiki/Variable_record_waste_resources > > -MC > > On Tue, Nov 15, 2011 at 8:33 PM, Joegen Baclor > wrote: > > I tried asking in the freeswitch irc channel but got no response. I'm > trying my luck here. > > I have recently received a packet capture from the sipx community that > seems to indicate that when FreeSwitch is recording a voice mail > message > via our custom IVR, that it does not send either comfort noise or > silence packets back to the caller. This results to the call getting > dropped by the ITSP after 30 seconds because of RTP time out. Is > there a > configurable parameter to make the IVR send some voice activity back? > > Joegen > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111118/3a96e279/attachment.html From henrikaagaardsorensen at gmail.com Fri Nov 18 03:33:41 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Fri, 18 Nov 2011 01:33:41 +0100 Subject: [Freeswitch-users] Open Source MSC. Message-ID: Hi everyone. I do know that this is the mailing list for FreeSwitch. But as I'm struggling finding anything (on Google etc.) about any open source MSC (HLR/AUC/SMSC/MMSC etc.) I'm trying here. I want to use FreeSwitch in a MVNO setup with a operator to make a proof-of-concept idea. But I need the MSC handling SIGTRAN and acts as HLR/AUC etc. I've looked at http://www.openimscore.org/ and http://www.mobicents.org, but I'm struggling figuring out there place in such setup. This is just for the proof-of-concept, so scale-ability, stability etc. is not as big a priority. Can anyone help me in the right direction? All help is appreciated! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/7da0ef45/attachment.html From brian at freeswitch.org Fri Nov 18 04:15:01 2011 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Nov 2011 19:15:01 -0600 Subject: [Freeswitch-users] Polycom phone registration problem In-Reply-To: <41619b09-5666-468f-a34d-372f10c0063a@zimbra1.crocker.com> References: <41619b09-5666-468f-a34d-372f10c0063a@zimbra1.crocker.com> Message-ID: <3CCC7F73-9C9E-49F2-AC6D-EDECF47E69DB@freeswitch.org> or just set force-rport=safe /b On Nov 17, 2011, at 5:39 PM, Matthew S. Crocker wrote: > Couldn't you do something like > > if $contact.address != ip.src_addr && sip_agent contains Polycom THEN NDLB-connectile-dysfunction > > -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/a7c73609/attachment.html From help at pdscc.com Fri Nov 18 04:19:40 2011 From: help at pdscc.com (Harondel J. Sibble) Date: Thu, 17 Nov 2011 17:19:40 -0800 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: References: Message-ID: Yes but you still can't use outlook. -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com Blog: http://www.pdscc.com/blog (604) 739-3709 (voice) "Henrik Aagaard S?rensen" wrote: Hi everyone. I do know that this is the mailing list for FreeSwitch. But as I'm struggling finding anything (on Google etc.) about any open source MSC (HLR/AUC/SMSC/MMSC etc.) I'm trying here. I want to use FreeSwitch in a MVNO setup with a operator to make a proof-of-concept idea. But I need the MSC handling SIGTRAN and acts as HLR/AUC etc. I've looked at http://www.openimscore.org/ and http://www.mobicents.org, but I'm struggling figuring out there place in such setup. This is just for the proof-of-concept, so scale-ability, stability etc. is not as big a priority. Can anyone help me in the right direction? All help is appreciated! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/278dac32/attachment.html From elliott at zoogmedia.com Fri Nov 18 04:24:16 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Fri, 18 Nov 2011 01:24:16 +0000 Subject: [Freeswitch-users] Polycom phone registration problem In-Reply-To: <3CCC7F73-9C9E-49F2-AC6D-EDECF47E69DB@freeswitch.org> References: <41619b09-5666-468f-a34d-372f10c0063a@zimbra1.crocker.com> <3CCC7F73-9C9E-49F2-AC6D-EDECF47E69DB@freeswitch.org> Message-ID: Where would I set this in config? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, November 17, 2011 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Polycom phone registration problem or just set force-rport=safe /b On Nov 17, 2011, at 5:39 PM, Matthew S. Crocker wrote: Couldn't you do something like if $contact.address != ip.src_addr && sip_agent contains Polycom THEN NDLB-connectile-dysfunction -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/0a9e0b26/attachment-0001.html From henrikaagaardsorensen at gmail.com Fri Nov 18 04:26:59 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Fri, 18 Nov 2011 02:26:59 +0100 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: References: Message-ID: Hi Harondel. I'm sorry, but I'm a bit confused with your reply? On Fri, Nov 18, 2011 at 2:19 AM, Harondel J. Sibble wrote: > ** Yes but you still can't use outlook. > -- > Harondel J. Sibble > Sibble Computer Consulting > Creating Solutions for the small and medium business computer user. > help at pdscc.com (use pgp keyid 0x3AD5C11D) > http://www.pdscc.com > Blog: http://www.pdscc.com/blog > (604) 739-3709 (voice) > > > "Henrik Aagaard S?rensen" wrote: >> >> Hi everyone. >> >> I do know that this is the mailing list for FreeSwitch. But as I'm >> struggling finding anything (on Google etc.) about any open source MSC >> (HLR/AUC/SMSC/MMSC etc.) I'm trying here. >> >> I want to use FreeSwitch in a MVNO setup with a operator to make a >> proof-of-concept idea. >> >> But I need the MSC handling SIGTRAN and acts as HLR/AUC etc. >> >> I've looked at http://www.openimscore.org/ and http://www.mobicents.org, >> but I'm struggling figuring out there place in such setup. >> >> This is just for the proof-of-concept, so scale-ability, stability etc. >> is not as big a priority. >> >> Can anyone help me in the right direction? All help is appreciated! >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111118/c36cb9be/attachment.html From freeswitch-list at puzzled.xs4all.nl Fri Nov 18 04:33:42 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 18 Nov 2011 02:33:42 +0100 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: References: Message-ID: <4EC5B5F6.80600@puzzled.xs4all.nl> On 11/18/2011 01:33 AM, Henrik Aagaard S?rensen wrote: > Hi everyone. > > I do know that this is the mailing list for FreeSwitch. But as I'm > struggling finding anything (on Google etc.) about any open source MSC > (HLR/AUC/SMSC/MMSC etc.) I'm trying here. > > I want to use FreeSwitch in a MVNO setup with a operator to make a > proof-of-concept idea. > > But I need the MSC handling SIGTRAN and acts as HLR/AUC etc. > > I've looked at http://www.openimscore.org/ and http://www.mobicents.org > , but I'm struggling figuring out there place > in such setup. > > This is just for the proof-of-concept, so scale-ability, stability etc. > is not as big a priority. > > Can anyone help me in the right direction? All help is appreciated! I am not aware of one Open Source project that does all you are looking for. OpenBSC (http://openbsc.osmocom.org/trac/) seems to come close (MSC, BSC, AUC, HLR, VLR, EIR). Kannel (www.kannel.org) might be some piece of the SMSC puzzle and for SIGTRAN Mobicents-SS7 seems to fit the bill: http://www.mobicents.org/ss7/intro.html If you are going the TDM route too then Sangoma has excellent TDM cards (and hardware EC cards) and they are supported by Mobicents Have fun integrating all the parts. Seems like a fun job :) Regards, Patrick From notlikeme75 at yahoo.com Fri Nov 18 04:35:06 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 17 Nov 2011 17:35:06 -0800 (PST) Subject: [Freeswitch-users] conference room announce In-Reply-To: References: Message-ID: <1321580106.82528.YahooMailNeo@web65306.mail.ac2.yahoo.com> the conference name is "conference room1" in the dialplan. should i be naming it just a number? ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Thursday, November 17, 2011 5:36 PM Subject: FreeSWITCH-users Digest, Vol 65, Issue 133 ----- Forwarded Message ----- 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: call count bind digit on static conference (Michael Collins) ? 2. Re: a=fmtp missing on FS answer (Brian West) ? 3. Re: calling limit_usage via ESL (Avi Marcus) ? 4. Re: Timing issues in AWS? (Anthony Minessale) Conference conference room 1-127.0.0.1 (1 member rate:8000)13; sofia/external/2064200990 at 66.54.140.446;b11d94a6-94ff-4955-bc87-13e332a7443;unknown;2064200990;hear:speak:floor;0;0;0;50 > >I try to "conference conference room 1-127.0.0.1" > >I get response "conference not found" > > only one "conference" - like this: conference list or conference 1-127.0.0.1 list -MC {sip_append_audio_sdp=a=fmtp:18 annexb=no}sofia/internal/foo at bar.com Or some variation of that? /b On Nov 17, 2011, at 3:05 PM, Camila Troncoso wrote: This isnt the solution because it only appends rtpmap not fmtp, I tried it. > > > >*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus >*Sent:* jueves, 17 de noviembre de 2011 18:01 >*To:* FreeSWITCH Users Help >*Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer > > > >Anthony said you probably want this: > >http://wiki.freeswitch.org/wiki/Variable_verbose_sdp > > >-Avi > > > >On Thu, Nov 17, 2011 at 10:24 PM, Camila Troncoso >wrote: > >I?m trying to make a call trough FS with G729 annexb=no in one side and >G729 annexb=no ( the same) in the other. I have a sangoma transcoding card >installed and configured, all is working fine. I use this card because in >some cases I need transcoding . So I have late-negotiation=true ?and >disable-transcoding=false. The problem is that FS answer without a=fmtp as >show in the picture. Freeswitch version is 1.0 > > > >*Error! Filename not specified.**Error! Filename not specified.* > > > > > > > >*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West >*Sent:* jueves, 17 de noviembre de 2011 16:18 >*To:* FreeSWITCH Users Help >*Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer > > > >what exactly are you doing and what are the sip traces? ?What firmware rev? > > > >/b > > > >On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: > > > >sip_append_audio_sdp > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.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 >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.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 > no using "execute" it hangs. Can I craft/view ESL stuff raw somewhere other than telnet? I recall hitting ctrl-C and killing freeswitch in telnet.. -Avi On Thu, Nov 17, 2011 at 6:19 PM, Hector Geraldino wrote: > What about: ?"execute" + command + " " + arguments > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi > Marcus > Sent: Wednesday, November 16, 2011 6:20 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] calling limit_usage via ESL > > > > The wiki says it's an API also. I have frequently executed it directly in > fs_cli. > > When I try it without "api" at the start, it hangs. > > -Avi > > > > On Thu, Nov 17, 2011 at 1:00 AM, Hector Geraldino > wrote: > > I think that, if it?s an application (i.e. can be executed directly from the > dialplan), you shouldn?t use the ?api? prefix. > > > > I?m not familiar with the php esl module, but in the Java ESL there?s a > difference when you try to call an api command vs and application: > > > > API: > > ??????????????? "api " + command + " " + arguments > > > > APP: > > SendMsg message = new SendMsg(uuid); > > message.addCallCommand("execute"); > > ??????? ??????? message.addExecuteAppName(appName); > > ??????????????? message.addExecuteAppArg(arguments); > > > > Also, you can always inspect the execution result if you?re executing the > command in sync mode. > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi > Marcus > Sent: Wednesday, November 16, 2011 5:41 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] calling limit_usage via ESL > > > > I'm using the php esl implementation in fusionpbx.. I'm able to pass "api > show channels count" and "api show calls count" and the like, but when I do: > > "api?limit_usage hash origination usa_pd" it's just blank. > > Pasting "limit_usage hash origination usa_pd" into fs_cli yields 0 (or > higher as the case may be). > > > > There DOES seem to be a \n returned, but nothing else. I've got no 'friggin > clue. > > Help, please? > > Thanks, > > Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > the "amazon linux" works well with modern GIT revisions. older ones need a centos kernel to work well. On Thu, Nov 17, 2011 at 9:19 AM, Michael Picher wrote: Huh...? maybe I need to get my own CentOS build up there instead of rely on Rightscale's 64 bit image...? That's a whole other ball of wax I was trying to avoid having to figure out to do... > >Mike > > > >On Thu, Nov 17, 2011 at 8:49 AM, Stephen Dame wrote: > >I?m running freeswitch on about 40 ?different m1.small c1.medium?s in AWS regions, us-east, us-west, eu-west, and asia?.? They are used in videoconf component of BigBlueButton.org .?? For most part they work great? There are occasional issues with voip quality but the app is 100% voip, with the BBB client all browser based.? So the conference is subject to every ones local network connections and most issues ?blamed? on the internet instead of freeswitch J?We also tie in skype and DID direct to improve latency for some clients.? But expectations are set so we meet them. >>? >>You probably have tougher business conferencing clients that want perfect audio.? But these are production deployed and generate revenue.????? All these are on Ubuntu 10.04 official amis with no mods to kernels.? >>? >>I?m sure there are tweaks that can be made, and bare metal solutions that would work a little better?? Does anyone have any ideas how to optimize a Ubuntu instance.? I would engage in a few hours of consulting is so. >>? >>The tests below are interesting but above my paygrade to understand what they mean? >>? >>I?m running them but don?t have a clue how to interpret.? If you want a simple of real running data, would be glad to run sample tests on these distributed servers and provide back for analysis. >>? >>Regards, >>Stephen >>? >>From:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Picher >>Sent: Thursday, November 17, 2011 8:03 AM >>To: FreeSWITCH Users Help >>Subject: Re: [Freeswitch-users] Timing issues in AWS? >>? >>Not having anywhere near the same trouble in a Xen Server in my lab... >> >>Although test_time is skewed 180 degrees from what it was in AWS.... >> >>freeswitch at internal> timer_test 120 10 >>Avg: 119.945ms Total Time: 1199.727ms >> >>2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 >>2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 >>2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120993 >>2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 118914 >>2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119919 >>2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120032 >>2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 119802 >>2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 119999 >>2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119975 >>2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 120003 >>2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 119849 >>2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119971 >> >>freeswitch at internal> time_test 600 10 >>test 1 sleep 600 1 >>test 2 sleep 600 1 >>test 3 sleep 600 0 >>test 4 sleep 600 1 >>test 5 sleep 600 0 >>test 6 sleep 600 1 >>test 7 sleep 600 1 >>test 8 sleep 600 0 >>test 9 sleep 600 0 >>test 10 sleep 600 1 >>avg 0 >> >>[root at openuc bin]# uname -r >>2.6.18-274.7.1.el5 >> >>[root at openuc bin]# grep CONFIG_HZ /boot/config-* >>/boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set >>/boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set >>/boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y >>/boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 >>/boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set >>/boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set >>/boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y >>/boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 >>/boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set >>/boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set >>/boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y >>/boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 >>/boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set >>/boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set >>/boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y >>/boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 >>/boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set >>/boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set >>/boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y >>/boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000 >> >>Thoughts as to why AWS results are so different from XenServer?? Other than not knowing who else is on the AWS box? >> >>Thanks, >>? Mike >> >> >> >>On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher wrote: >>m1.large >> >>I have a c1.xlarge queued up and ready to test... >>? >>On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen wrote: >>Just a simple question, what kind of AWS instance are you running your FreeSWITCH? >>It makes huge difference. >>Thanks, >>Chris >>On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher wrote: >>Hi guys, >>> >>>Trying to get to the bottom of some conference bridge issues I'm having with running the system in AWS. >>> >>>We're hearing a bunch of snap-crackle-pops in conference bridges and when I tcpdum on the server itself I see them in the RTP and see RTP timestamp problems. >>> >>>I've run the following: >>> >>> >>>timer_test >>>freeswitch at 127.0.0.1@internal> timer_test 120 10 >>>Avg: 120.004ms Total Time: 1200.315ms >>>? >>>2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 >>>freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 >>>2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120006 >>>2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 120040 >>>2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119976 >>>2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120007 >>>2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 120001 >>>2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 120031 >>>2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119996 >>>2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 119994 >>>2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 120005 >>>2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119991 >>> >>>test_time >>>freeswitch at 127.0.0.1@internal> time_test 600 10 >>> >>>test 1 sleep 600 1592 >>>test 2 sleep 600 986 >>>test 3 sleep 600 1018 >>>test 4 sleep 600 980 >>>test 5 sleep 600 1005 >>>test 6 sleep 600 1000 >>>test 7 sleep 600 972 >>>test 8 sleep 600 990 >>>test 9 sleep 600 1006 >>>test 10 sleep 600 994 >>>avg 1054 >>> >>>For kernel: >>>[root at openuc bin]# uname -r >>> >>>2.6.21.7-2.fc8xen >>> >>>CONFIG_HZ: >>>[root at openuc bin]# grep CONFIG_HZ /boot/config-* >>> >>>/boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y >>>/boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set >>>/boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set >>>/boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 >>>/boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set >>>/boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set >>>/boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y >>>/boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 >>>/boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set >>>/boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set >>>/boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set >>>/boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y >>>/boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 >>>are the xenU kernel settings screwing me here? >>> >>>Thanks, >>>? Mike >>> >>> >>>-- >>>There are 10 kinds of people in this world, those who understand binary and those who don't. >>> >>>mpicher at gmail.com >>>blog: http://www.sipxecs.info >>>call: sip:mpicher at sipxecs.info >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>> >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.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 >>? >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.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 >> >> >> >>-- >>There are 10 kinds of people in this world, those who understand binary and those who don't. >> >>mpicher at gmail.com >>blog: http://www.sipxecs.info >>call: sip:mpicher at sipxecs.info >> >> >> >>-- >>There are 10 kinds of people in this world, those who understand binary and those who don't. >> >>mpicher at gmail.com >>blog: http://www.sipxecs.info >>call: sip:mpicher at sipxecs.info >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.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 >> >> > > >-- >There are 10 kinds of people in this world, those who understand binary and those who don't. > >mpicher at gmail.com >blog: http://www.sipxecs.info >call: sip:mpicher at sipxecs.info > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/3d9e3eca/attachment-0001.html From brian at freeswitch.org Fri Nov 18 04:42:23 2011 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Nov 2011 19:42:23 -0600 Subject: [Freeswitch-users] Polycom phone registration problem In-Reply-To: References: <41619b09-5666-468f-a34d-372f10c0063a@zimbra1.crocker.com> <3CCC7F73-9C9E-49F2-AC6D-EDECF47E69DB@freeswitch.org> Message-ID: <4D59F1E5-FC29-4FC2-87A1-BF0853C80F39@freeswitch.org> in the sofia profile. /b On Nov 17, 2011, at 7:24 PM, Elliott Vogel wrote: > Where would I set this in config? > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West > Sent: Thursday, November 17, 2011 7:15 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Polycom phone registration problem > > or just set force-rport=safe > > /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111117/69036178/attachment.html From elliott at zoogmedia.com Fri Nov 18 09:08:20 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Fri, 18 Nov 2011 06:08:20 +0000 Subject: [Freeswitch-users] dial plan help Message-ID: Does anyone has an example on how to setup a normalization rule in the dial plan? basically I want to transform 7 digit number into to 10 digit number and then continue processing the dial plan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/3f5cdd61/attachment.html From avi at avimarcus.net Fri Nov 18 12:04:27 2011 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 18 Nov 2011 11:04:27 +0200 Subject: [Freeswitch-users] dial plan help In-Reply-To: References: Message-ID: You either need to normalize all number via a second variable - e.g. ${destination_fixed} or something... and set that for everything. Hmm, using the new OR logic in the dialplan, you can actually do better things... but it still can have complication. Or, much simpler - but slightly higher on resource usage - is to transfer the call back to whichever context at the full number. e.g. This takes all 7 digit calls and restarts the dialplan as 1-212-$x. So I'd recommend putting this at the beginning of your dialplan.. -Avi Marcus On Fri, Nov 18, 2011 at 8:08 AM, Elliott Vogel wrote: > Does anyone has an example on how to setup a normalization rule in the dial > plan? basically I want to transform 7 digit number into to 10 digit number > and then continue processing the dial plan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 avi at avimarcus.net Fri Nov 18 12:12:48 2011 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 18 Nov 2011 11:12:48 +0200 Subject: [Freeswitch-users] iptables dropping SIP packets..? In-Reply-To: <4EC50CF6.8090103@chaschperli.ch> References: <4EC50CF6.8090103@chaschperli.ch> Message-ID: maybe.. the yealink does have some long packets. I tried just to see... # modprobe -r nf_ct_sip FATAL: Module nf_ct_sip not found. It's not it's own but part of nf_conntrack_sip? That doesn't sound as safe to remove.. (p.s. I have no idea what I'm mucking with.) -Avi On Thu, Nov 17, 2011 at 3:32 PM, Thomas Mueller wrote: > > > On 17.11.2011 14:06, Avi Marcus wrote: > > Service: sip (udp/5060) (nf_ct_sip: dropping packet) - 9 packets > I see this every few days - it's via an IP with a Yealink T20p phone. I've > tried looking this up in the past.. but I don't understand what rules would > be dropping this, and what is being dropped. > Any clues on how to investigate? > > Maybe this is related to this bugreport? > > nf_ct_sip dropping SIP messages larger then MTU > http://bugzilla.netfilter.org/show_bug.cgi?id=760 > > There is some possibility that you don't need the nf_ct_sip - just unload it > (modprobe -r) and if it works, do blacklist the module in (for exmple in > /etc/modprobe.d/00-my-local.conf ) > > - Thomas > > From raimund.sacherer at logitravel.com Fri Nov 18 13:02:20 2011 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Fri, 18 Nov 2011 11:02:20 +0100 (CET) Subject: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! In-Reply-To: <31d59404-8563-4920-b64a-f79a7ba95e0e@Raimund-ThinkPad-X61s> Message-ID: <6e1064d8-f3fa-4a86-adf3-94a74f22b497@Raimund-ThinkPad-X61s> Hello, I have a rather problematic provider with which I am stuck in Brasil. After wrestling around I am able to make calls, but when FreeSWITCH send's it's BYE, I do receive a 200 OK, but it seems that FreeSWITCH does not recognize what it receives and keeps sending BYE's with Answers from the remote party that the call no longer exists. Looking at the Sip trace and the debug log (attached as txt file) everything *seems* fine ... but I am not a FS developer either, so I would appreciate if someone with more experience can have a look at it. When connecting with a Eyebeam or X-Lite this problem does not exist, but I could not tell a real difference from the tcp-dumps other than X-Lite uses TCP instead of UPD. Trying to use TCP in FreeSWITCH times out after receiving a 100 Giving a try messages after the Invite, so not sure what to make of this either ..., but I would settle for a working BYE and move on ... Thank you, Best Ray IRC:Hatrix76, Hatrix, Hatrix|Away -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/40a05af5/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: surfix_trace_and_log.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/40a05af5/attachment-0001.txt From jbaclor at ezuce.com Fri Nov 18 14:46:28 2011 From: jbaclor at ezuce.com (Joegen Baclor) Date: Fri, 18 Nov 2011 19:46:28 +0800 Subject: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! In-Reply-To: <6e1064d8-f3fa-4a86-adf3-94a74f22b497@Raimund-ThinkPad-X61s> References: <31d59404-8563-4920-b64a-f79a7ba95e0e@Raimund-ThinkPad-X61s> <6e1064d8-f3fa-4a86-adf3-94a74f22b497@Raimund-ThinkPad-X61s> Message-ID: CSeq in the 200 Ok is wrong. It should match the CSeq of the bye request. Should be CSeq: 20466239 BYE On Fri, Nov 18, 2011 at 6:02 PM, Raimund Sacherer < raimund.sacherer at logitravel.com> wrote: > Hello, > > I have a rather problematic provider with which I am stuck in Brasil. > After wrestling around I am able to make calls, but when FreeSWITCH send's > it's BYE, I do receive a 200 OK, but it seems that FreeSWITCH does not > recognize what it receives and keeps sending BYE's with Answers from the > remote party that the call no longer exists. > > Looking at the Sip trace and the debug log (attached as txt file) > everything *seems* fine ... but I am not a FS developer either, so I would > appreciate if someone with more experience can have a look at it. > > When connecting with a Eyebeam or X-Lite this problem does not exist, but > I could not tell a real difference from the tcp-dumps other than X-Lite > uses TCP instead of UPD. > > Trying to use TCP in FreeSWITCH times out after receiving a 100 Giving a > try messages after the Invite, so not sure what to make of this either ..., > but I would settle for a working BYE and move on ... > > Thank you, > Best > Ray > > IRC:Hatrix76, Hatrix, Hatrix|Away > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111118/bcceea72/attachment.html From raimund.sacherer at logitravel.com Fri Nov 18 15:00:11 2011 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Fri, 18 Nov 2011 13:00:11 +0100 (CET) Subject: [Freeswitch-users] High load on database server In-Reply-To: <6120aaed-bd6b-46b1-a853-a17ab10244b5@Raimund-ThinkPad-X61s> Message-ID: Hello, I have had lot's of issues with 2.14 odbc as well, the Threadong parameter helped to reduce the crashes from approx. 4 a day to 1 a day, but the crashing was still there. I compiled my own debian packages for odbc with the 2.11 centos 5 sources (so I can install other debiacn packages that depend on odbc) and this works like a charm! So I would recommend using 2.11 instead of 2.14, I do not have expierence with newer odbc's though. using postgres 9.(0|1), 8.4 and debian squeeze with fairly recent Freeswitch boxes best ----- Original Message ----- From: "Cliff Wells" To: "FreeSWITCH Users Help" Sent: Thursday, October 27, 2011 8:02:46 AM Subject: Re: [Freeswitch-users] High load on database server On Wed, 2011-10-26 at 11:12 -0500, Anthony Minessale wrote: > if you have older unixodbc edit /etc/odbcinst.ini and add the param > Threading=0 to each db entry. > In general postgres to me appears to care more about multithreaded > environments and with the above threading=0 change works pretty well. I tried this (after googling a bit and coming across an older thread you responded to) and it didn't change anything. Crashes at ~330 concurrent calls like clockwork. This was with unixODBC 2.14 and PostgreSQL 8.4 odbc driver. Regards, Cliff FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 S?guenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para m?vil Logitravel.com Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por error, se ruega notif?quese inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formar?n parte de un fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que deber? ejercer mediante escrito a la direcci?n anteriormente citada. From raimund.sacherer at logitravel.com Fri Nov 18 15:03:41 2011 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Fri, 18 Nov 2011 13:03:41 +0100 (CET) Subject: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! In-Reply-To: <4e0283bb-426f-4a3b-87d3-a202e1267504@Raimund-ThinkPad-X61s> Message-ID: <6f3a5ec0-8cae-43cb-8f15-ed2097268bd9@Raimund-ThinkPad-X61s> I read up on the RFC regarding that and it states that only the METHOD (BYE) has to be the same, only for REGISTERS the CSeq has to be the same, so If I interpret the RFC correctly (which does not have to be the case) this should not be the issue. best Ray ----- Original Message ----- From: "Joegen Baclor" To: "FreeSWITCH Users Help" Sent: Friday, November 18, 2011 12:46:28 PM Subject: Re: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! CSeq in the 200 Ok is wrong. It should match the CSeq of the bye request. Should be CSeq: 20466239 BYE On Fri, Nov 18, 2011 at 6:02 PM, Raimund Sacherer < raimund.sacherer at logitravel.com > wrote: Hello, I have a rather problematic provider with which I am stuck in Brasil. After wrestling around I am able to make calls, but when FreeSWITCH send's it's BYE, I do receive a 200 OK, but it seems that FreeSWITCH does not recognize what it receives and keeps sending BYE's with Answers from the remote party that the call no longer exists. Looking at the Sip trace and the debug log (attached as txt file) everything *seems* fine ... but I am not a FS developer either, so I would appreciate if someone with more experience can have a look at it. When connecting with a Eyebeam or X-Lite this problem does not exist, but I could not tell a real difference from the tcp-dumps other than X-Lite uses TCP instead of UPD. Trying to use TCP in FreeSWITCH times out after receiving a 100 Giving a try messages after the Invite, so not sure what to make of this either ..., but I would settle for a working BYE and move on ... Thank you, Best Ray IRC:Hatrix76, Hatrix, Hatrix|Away _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 S?guenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para m?vil Logitravel.com Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por error, se ruega notif?quese inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formar?n parte de un fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que deber? ejercer mediante escrito a la direcci?n anteriormente citada. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/3ca3c688/attachment-0001.html From brian at freeswitch.org Fri Nov 18 15:16:38 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Nov 2011 06:16:38 -0600 Subject: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! In-Reply-To: <6e1064d8-f3fa-4a86-adf3-94a74f22b497@Raimund-ThinkPad-X61s> References: <6e1064d8-f3fa-4a86-adf3-94a74f22b497@Raimund-ThinkPad-X61s> Message-ID: That would indicate the dialog is not matching or the network path is not right! /b Sent from my iPhone On Nov 18, 2011, at 4:02 AM, Raimund Sacherer wrote: > Hello, > > I have a rather problematic provider with which I am stuck in Brasil. After wrestling around I am able to make calls, but when FreeSWITCH send's it's BYE, I do receive a 200 OK, but it seems that FreeSWITCH does not recognize what it receives and keeps sending BYE's with Answers from the remote party that the call no longer exists. > > Looking at the Sip trace and the debug log (attached as txt file) everything *seems* fine ... but I am not a FS developer either, so I would appreciate if someone with more experience can have a look at it. > > When connecting with a Eyebeam or X-Lite this problem does not exist, but I could not tell a real difference from the tcp-dumps other than X-Lite uses TCP instead of UPD. > > Trying to use TCP in FreeSWITCH times out after receiving a 100 Giving a try messages after the Invite, so not sure what to make of this either ..., but I would settle for a working BYE and move on ... > > Thank you, > Best > Ray > > IRC:Hatrix76, Hatrix, Hatrix|Away > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111118/0a3a2d2e/attachment.html From raimund.sacherer at logitravel.com Fri Nov 18 15:34:33 2011 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Fri, 18 Nov 2011 13:34:33 +0100 (CET) Subject: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! In-Reply-To: <6544fea3-9ec3-415c-a463-049850ae6f76@Raimund-ThinkPad-X61s> Message-ID: Hello Brian, AFAICT the tags and the Call-ID and the Cseq method are correct in the Trace, therefore, if someone with more indepth knowledge could have a look on the (in the first email) atteched fs+log with siptrace I would be glad to hear what's exactly the problem, I did not open a JIRA because I do not think it's an FS bug. In any case here is a PB link to the same log+trace: http://pastebin.freeswitch.org/17816 best Ray ----- Original Message ----- From: "Brian West" To: "FreeSWITCH Users Help" Cc: "freeswitch-users" Sent: Friday, November 18, 2011 1:16:38 PM Subject: Re: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! That would indicate the dialog is not matching or the network path is not right! /b Sent from my iPhone On Nov 18, 2011, at 4:02 AM, Raimund Sacherer < raimund.sacherer at logitravel.com > wrote: Hello, I have a rather problematic provider with which I am stuck in Brasil. After wrestling around I am able to make calls, but when FreeSWITCH send's it's BYE, I do receive a 200 OK, but it seems that FreeSWITCH does not recognize what it receives and keeps sending BYE's with Answers from the remote party that the call no longer exists. Looking at the Sip trace and the debug log (attached as txt file) everything *seems* fine ... but I am not a FS developer either, so I would appreciate if someone with more experience can have a look at it. When connecting with a Eyebeam or X-Lite this problem does not exist, but I could not tell a real difference from the tcp-dumps other than X-Lite uses TCP instead of UPD. Trying to use TCP in FreeSWITCH times out after receiving a 100 Giving a try messages after the Invite, so not sure what to make of this either ..., but I would settle for a working BYE and move on ... Thank you, Best Ray IRC:Hatrix76, Hatrix, Hatrix|Away
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111118/b133ab47/attachment.html From raimund.sacherer at logitravel.com Fri Nov 18 15:36:53 2011 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Fri, 18 Nov 2011 13:36:53 +0100 (CET) Subject: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! In-Reply-To: Message-ID: <3517c311-6732-4b93-97bf-101306284f80@Raimund-ThinkPad-X61s> Just to have it directly in the mail, the problematic BYE with subsequent received messages: 2011-11-18 05:23:47.648898 [DEBUG] mod_sofia.c:508 Sending BYE to sofia/external-test/01140039999 send 997 bytes to udp/[187.111.X.Y]:5060 at 08:23:47.650139: ------------------------------------------------------------------------ BYE sip:0231140039999 at 187.111.X.Z SIP/2.0 Via: SIP/2.0/UDP 187.103.My.IP:5085;rport;branch=z9hG4bK2j52DyB29UUFc Route: Max-Forwards: 70 From: "Heinrich III S" ;tag=NyvaUBt4440rF To: ;tag=as62aa2285 Call-ID: 6c126984-8c61-122f-7e82-782bcb2cab60 CSeq: 20466239 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10df279 2011-10-15 07-59-23 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:47 sofia/external-test/01140039999 Standard HANGUP, cause: NORMAL_CLEARING Supported: 100rel, timer, precondition, path, replaces 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:602 (sofia/external-test/01140039999) State HANGUP going to sleep Proxy-Authorization: Digest username="USERNAME", realm="airtel.net.br", nonce="4ec616180000a60f73db499833afd9f3ec0df8d64801dd98", algorithm=MD5, uri="sip:0231140039999 at 187.111.X.Z", response="4877c4af90e769ed87bc7c17c5ddfcf3" Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:393 (sofia/external-test/01140039999) State Change CS_HANGUP -> CS_REPORTING 2011-11-18 05:23:47.648898 [DEBUG] switch_core_session.c:1177 Send signal sofia/external-test/01140039999 [BREAK] 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:362 (sofia/external-test/01140039999) Running State Change CS_REPORTING 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:662 (sofia/external-test/01140039999) State REPORTING 2011-11-18 05:23:47.648898 [DEBUG] switch_ivr_bridge.c:1332 sofia/external-test/01140039999 skip receive message [UNBRIDGE] (channel is hungup already) 2011-11-18 05:23:47.648898 [DEBUG] switch_ivr_bridge.c:1335 sofia/external/freeswitch at pbx.toolfactory.net skip receive message [UNBRIDGE] (channel is hungup already) 2011-11-18 05:23:47.648898 [DEBUG] switch_core_session.c:2261 sofia/external/freeswitch at pbx.toolfactory.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:417 (sofia/external/freeswitch at pbx.toolfactory.net) State EXECUTE going to sleep 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:362 (sofia/external/freeswitch at pbx.toolfactory.net) Running State Change CS_HANGUP 2011-11-18 05:23:47.648898 [DEBUG] switch_ivr_async.c:978 Stop recording file /usr/local/freeswitch/recordings/queues/2011-11-18-05-23-23.126.01140039999.94b27bf8-11be-11e1-be76-bfe92509c5c7.wav 2011-11-18 05:23:47.648898 [DEBUG] switch_core_media_bug.c:480 Removing BUG from sofia/external/freeswitch at pbx.toolfactory.net 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:602 (sofia/external/freeswitch at pbx.toolfactory.net) State HANGUP 2011-11-18 05:23:47.648898 [DEBUG] mod_sofia.c:459 sofia/external/freeswitch at pbx.toolfactory.net Overriding SIP cause 480 with 200 from the other leg 2011-11-18 05:23:47.648898 [DEBUG] mod_sofia.c:465 Channel sofia/external/freeswitch at pbx.toolfactory.net hanging up, cause: NORMAL_CLEARING 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:47 sofia/external/freeswitch at pbx.toolfactory.net Standard HANGUP, cause: NORMAL_CLEARING 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:602 (sofia/external/freeswitch at pbx.toolfactory.net) State HANGUP going to sleep 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:393 (sofia/external/freeswitch at pbx.toolfactory.net) State Change CS_HANGUP -> CS_REPORTING 2011-11-18 05:23:47.648898 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/freeswitch at pbx.toolfactory.net [BREAK] 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:362 (sofia/external/freeswitch at pbx.toolfactory.net) Running State Change CS_REPORTING 2011-11-18 05:23:47.648898 [DEBUG] switch_core_state_machine.c:662 (sofia/external/freeswitch at pbx.toolfactory.net) State REPORTING recv 461 bytes from udp/[187.111.X.Y]:5060 at 08:23:47.653748: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.PB.X:5085;received=192.168.PB.X;rport=5085;branch=z9hG4bK2j52DyB29UUFc From: "Heinrich III S" ;tag=NyvaUBt4440rF To: ;tag=as62aa2285 Call-ID: 6c126984-8c61-122f-7e82-782bcb2cab60 CSeq: 2147483647 BYE Server: Airtel Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 ------------------------------------------------------------------------ send 997 bytes to udp/[187.111.X.Y]:5060 at 08:23:48.652955: ------------------------------------------------------------------------ BYE sip:0231140039999 at 187.111.X.Z SIP/2.0 Via: SIP/2.0/UDP 187.103.My.IP:5085;rport;branch=z9hG4bK2j52DyB29UUFc Route: Max-Forwards: 70 From: "Heinrich III S" ;tag=NyvaUBt4440rF To: ;tag=as62aa2285 Call-ID: 6c126984-8c61-122f-7e82-782bcb2cab60 CSeq: 20466239 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10df279 2011-10-15 07-59-23 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, timer, precondition, path, replaces Proxy-Authorization: Digest username="USERNAME", realm="airtel.net.br", nonce="4ec616180000a60f73db499833afd9f3ec0df8d64801dd98", algorithm=MD5, uri="sip:0231140039999 at 187.111.X.Z", response="4877c4af90e769ed87bc7c17c5ddfcf3" Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ recv 461 bytes from udp/[187.111.X.Y]:5060 at 08:23:48.656032: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.PB.X:5085;received=192.168.PB.X;rport=5085;branch=z9hG4bK2j52DyB29UUFc From: "Heinrich III S" ;tag=NyvaUBt4440rF To: ;tag=as62aa2285 Call-ID: 6c126984-8c61-122f-7e82-782bcb2cab60 CSeq: 2147483647 BYE Server: Airtel Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 ------------------------------------------------------------------------ 2011-11-18 05:23:49.328907 [DEBUG] switch_core_state_machine.c:79 sofia/external-test/01140039999 Standard REPORTING, cause: NORMAL_CLEARING 2011-11-18 05:23:49.328907 [DEBUG] switch_core_state_machine.c:662 (sofia/external-test/01140039999) State REPORTING going to sleep 2011-11-18 05:23:49.328907 [DEBUG] switch_core_state_machine.c:387 (sofia/external-test/01140039999) State Change CS_REPORTING -> CS_DESTROY 2011-11-18 05:23:49.328907 [DEBUG] switch_core_session.c:1177 Send signal sofia/external-test/01140039999 [BREAK] 2011-11-18 05:23:49.328907 [DEBUG] switch_core_session.c:1377 Session 11055 (sofia/external-test/01140039999) Locked, Waiting on external entities 2011-11-18 05:23:49.328907 [NOTICE] switch_core_session.c:1395 Session 11055 (sofia/external-test/01140039999) Ended 2011-11-18 05:23:49.328907 [NOTICE] switch_core_session.c:1397 Close Channel sofia/external-test/01140039999 [CS_DESTROY] 2011-11-18 05:23:49.328907 [DEBUG] switch_core_state_machine.c:491 (sofia/external-test/01140039999) Callstate Change HANGUP -> DOWN 2011-11-18 05:23:49.328907 [DEBUG] switch_core_state_machine.c:494 (sofia/external-test/01140039999) Running State Change CS_DESTROY 2011-11-18 05:23:49.328907 [DEBUG] switch_core_state_machine.c:504 (sofia/external-test/01140039999) State DESTROY 2011-11-18 05:23:49.328907 [DEBUG] mod_sofia.c:370 sofia/external-test/01140039999 SOFIA DESTROY 2011-11-18 05:23:49.328907 [DEBUG] switch_core_state_machine.c:86 sofia/external-test/01140039999 Standard DESTROY 2011-11-18 05:23:49.328907 [DEBUG] switch_core_state_machine.c:504 (sofia/external-test/01140039999) State DESTROY going to sleep 2011-11-18 05:23:49.508907 [DEBUG] switch_core_state_machine.c:79 sofia/external/freeswitch at pbx.toolfactory.net Standard REPORTING, cause: NORMAL_CLEARING 2011-11-18 05:23:49.508907 [DEBUG] switch_core_state_machine.c:662 (sofia/external/freeswitch at pbx.toolfactory.net) State REPORTING going to sleep 2011-11-18 05:23:49.508907 [DEBUG] switch_core_state_machine.c:387 (sofia/external/freeswitch at pbx.toolfactory.net) State Change CS_REPORTING -> CS_DESTROY 2011-11-18 05:23:49.508907 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/freeswitch at pbx.toolfactory.net [BREAK] 2011-11-18 05:23:49.508907 [DEBUG] switch_core_session.c:1377 Session 11054 (sofia/external/freeswitch at pbx.toolfactory.net) Locked, Waiting on external entities 2011-11-18 05:23:49.508907 [NOTICE] switch_core_session.c:1395 Session 11054 (sofia/external/freeswitch at pbx.toolfactory.net) Ended 2011-11-18 05:23:49.508907 [NOTICE] switch_core_session.c:1397 Close Channel sofia/external/freeswitch at pbx.toolfactory.net [CS_DESTROY] 2011-11-18 05:23:49.508907 [DEBUG] switch_core_state_machine.c:491 (sofia/external/freeswitch at pbx.toolfactory.net) Callstate Change HANGUP -> DOWN 2011-11-18 05:23:49.508907 [DEBUG] switch_core_state_machine.c:494 (sofia/external/freeswitch at pbx.toolfactory.net) Running State Change CS_DESTROY 2011-11-18 05:23:49.508907 [DEBUG] switch_core_state_machine.c:504 (sofia/external/freeswitch at pbx.toolfactory.net) State DESTROY 2011-11-18 05:23:49.508907 [DEBUG] mod_sofia.c:370 sofia/external/freeswitch at pbx.toolfactory.net SOFIA DESTROY 2011-11-18 05:23:49.508907 [DEBUG] switch_core_state_machine.c:86 sofia/external/freeswitch at pbx.toolfactory.net Standard DESTROY 2011-11-18 05:23:49.508907 [DEBUG] switch_core_state_machine.c:504 (sofia/external/freeswitch at pbx.toolfactory.net) State DESTROY going to sleep send 997 bytes to udp/[187.111.X.Y]:5060 at 08:23:50.653194: ------------------------------------------------------------------------ BYE sip:0231140039999 at 187.111.X.Z SIP/2.0 Via: SIP/2.0/UDP 187.103.My.IP:5085;rport;branch=z9hG4bK2j52DyB29UUFc Route: Max-Forwards: 70 From: "Heinrich III S" ;tag=NyvaUBt4440rF To: ;tag=as62aa2285 Call-ID: 6c126984-8c61-122f-7e82-782bcb2cab60 CSeq: 20466239 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10df279 2011-10-15 07-59-23 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, timer, precondition, path, replaces Proxy-Authorization: Digest username="USERNAME", realm="airtel.net.br", nonce="4ec616180000a60f73db499833afd9f3ec0df8d64801dd98", algorithm=MD5, uri="sip:0231140039999 at 187.111.X.Z", response="4877c4af90e769ed87bc7c17c5ddfcf3" Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ recv 461 bytes from udp/[187.111.X.Y]:5060 at 08:23:50.656440: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.PB.X:5085;received=192.168.PB.X;rport=5085;branch=z9hG4bK2j52DyB29UUFc From: "Heinrich III S" ;tag=NyvaUBt4440rF To: ;tag=as62aa2285 Call-ID: 6c126984-8c61-122f-7e82-782bcb2cab60 CSeq: 2147483647 BYE Server: Airtel Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 ------------------------------------------------------------------------ send 997 bytes to udp/[187.111.X.Y]:5060 at 08:23:54.653194: ------------------------------------------------------------------------ BYE sip:0231140039999 at 187.111.X.Z SIP/2.0 Via: SIP/2.0/UDP 187.103.My.IP:5085;rport;branch=z9hG4bK2j52DyB29UUFc Route: Max-Forwards: 70 From: "Heinrich III S" ;tag=NyvaUBt4440rF To: ;tag=as62aa2285 Call-ID: 6c126984-8c61-122f-7e82-782bcb2cab60 CSeq: 20466239 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10df279 2011-10-15 07-59-23 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, timer, precondition, path, replaces Proxy-Authorization: Digest username="USERNAME", realm="airtel.net.br", nonce="4ec616180000a60f73db499833afd9f3ec0df8d64801dd98", algorithm=MD5, uri="sip:0231140039999 at 187.111.X.Z", response="4877c4af90e769ed87bc7c17c5ddfcf3" Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ recv 494 bytes from udp/[187.111.X.Y]:5060 at 08:23:54.656920: ------------------------------------------------------------------------ SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.PB.X:5085;received=192.168.PB.X;rport=5085;branch=z9hG4bK2j52DyB29UUFc From: "Heinrich III S" ;tag=NyvaUBt4440rF To: ;tag=as62aa2285 Call-ID: 6c126984-8c61-122f-7e82-782bcb2cab60 CSeq: 2147483647 BYE Server: Airtel Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 ------------------------------------------------------------------------ ----- Original Message ----- From: "Brian West" To: "FreeSWITCH Users Help" Cc: "freeswitch-users" Sent: Friday, November 18, 2011 1:16:38 PM Subject: Re: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! That would indicate the dialog is not matching or the network path is not right! /b Sent from my iPhone On Nov 18, 2011, at 4:02 AM, Raimund Sacherer < raimund.sacherer at logitravel.com > wrote: Hello, I have a rather problematic provider with which I am stuck in Brasil. After wrestling around I am able to make calls, but when FreeSWITCH send's it's BYE, I do receive a 200 OK, but it seems that FreeSWITCH does not recognize what it receives and keeps sending BYE's with Answers from the remote party that the call no longer exists. Looking at the Sip trace and the debug log (attached as txt file) everything *seems* fine ... but I am not a FS developer either, so I would appreciate if someone with more experience can have a look at it. When connecting with a Eyebeam or X-Lite this problem does not exist, but I could not tell a real difference from the tcp-dumps other than X-Lite uses TCP instead of UPD. Trying to use TCP in FreeSWITCH times out after receiving a 100 Giving a try messages after the Invite, so not sure what to make of this either ..., but I would settle for a working BYE and move on ... Thank you, Best Ray IRC:Hatrix76, Hatrix, Hatrix|Away
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 notlikeme75 at yahoo.com Fri Nov 18 16:00:35 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Fri, 18 Nov 2011 05:00:35 -0800 (PST) Subject: [Freeswitch-users] voice bbs using voicemail mod Message-ID: <1321621235.26803.YahooMailNeo@web65311.mail.ac2.yahoo.com> windows server 2008 32 bit. I am trying to create a public voice bulletin board system that would let users on the ivr record messages and play the last 100 in a last in first out mode. I believe i can do this by using voicemail mod. is there a way to add "profiles" to voicemail mod like we do for conferences? if so then i would be able to have regular voicemail options and a separate voice bbs profile that would manage me creating a single or more voice bbs voicemail box that acts differently and doesn't allow public users to delete messages on the bbs box. if this isn't the case, then would it work to create a voice bbs mod based on voicemail mod and have it point to a voicebbs.conf.xml like? voicemail.conf.xml does? if so, how do i create/edit this dll file in windows to do such a thing? how do i get ahold of the creators of voicemail mod for maybe an edit? thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/ae146a26/attachment.html From ctroncoso at redvoiss.net Fri Nov 18 16:19:15 2011 From: ctroncoso at redvoiss.net (Camila Troncoso) Date: Fri, 18 Nov 2011 10:19:15 -0300 Subject: [Freeswitch-users] a=fmtp missing on FS answer In-Reply-To: <3F2BAC20-F525-4749-BA37-2B6BE7BED67D@freeswitch.org> References: <24767b5c4ead7d1bcc6eee30753006f3@mail.gmail.com> <3407b5c8e286e2ae1b9f531520db2a8e@mail.gmail.com> <3F2BAC20-F525-4749-BA37-2B6BE7BED67D@freeswitch.org> Message-ID: <48ccd6e8d7916baf23f894df22024d43@mail.gmail.com> Yes , i tried that solution, but I force it to be annexb=no, what if the calling party has annexb=yes ( and the called party also)]? I can?t know ?a priori? what codec will be used. I?m looking for a more drastic solution , where the answer form FS takes also into account the fmtp from the negotiated codec . *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* jueves, 17 de noviembre de 2011 18:34 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer {sip_append_audio_sdp=a=fmtp:18 annexb=no}sofia/internal/foo at bar.com Or some variation of that? /b On Nov 17, 2011, at 3:05 PM, Camila Troncoso wrote: This isnt the solution because it only appends rtpmap not fmtp, I tried it. *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* jueves, 17 de noviembre de 2011 18:01 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer Anthony said you probably want this: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp -Avi On Thu, Nov 17, 2011 at 10:24 PM, Camila Troncoso wrote: I?m trying to make a call trough FS with G729 annexb=no in one side and G729 annexb=no ( the same) in the other. I have a sangoma transcoding card installed and configured, all is working fine. I use this card because in some cases I need transcoding . So I have late-negotiation=true and disable-transcoding=false. The problem is that FS answer without a=fmtp as show in the picture. Freeswitch version is 1.0 *Error! Filename not specified.**Error! Filename not specified.* *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* jueves, 17 de noviembre de 2011 16:18 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer what exactly are you doing and what are the sip traces? What firmware rev? /b On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: sip_append_audio_sdp _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111118/082b0424/attachment.html From miha at softnet.si Fri Nov 18 17:10:04 2011 From: miha at softnet.si (Miha Zoubek) Date: Fri, 18 Nov 2011 15:10:04 +0100 Subject: [Freeswitch-users] A in B leg CDR still sending after value=fals Message-ID: <4EC6673C.2070000@softnet.si> HI, I have problem with modul mod_radius_cdr and dial plan. I have try to disable a and than b leg in my dial plan, but freeswitch is still sending values. Please tell me what I am doing wrong. Dial plan: Regards, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/53343179/attachment.html From brian at freeswitch.org Fri Nov 18 17:19:23 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Nov 2011 08:19:23 -0600 Subject: [Freeswitch-users] a=fmtp missing on FS answer In-Reply-To: <48ccd6e8d7916baf23f894df22024d43@mail.gmail.com> References: <24767b5c4ead7d1bcc6eee30753006f3@mail.gmail.com> <3407b5c8e286e2ae1b9f531520db2a8e@mail.gmail.com> <3F2BAC20-F525-4749-BA37-2B6BE7BED67D@freeswitch.org> <48ccd6e8d7916baf23f894df22024d43@mail.gmail.com> Message-ID: If its not doing what you want.. and you tried everything you know email consulting at freeswitch.org for a quote to get it to do what you want. /b On Nov 18, 2011, at 7:19 AM, Camila Troncoso wrote: > Yes , i tried that solution, but I force it to be annexb=no, what if the > calling party has annexb=yes ( and the called party also)]? I can?t know ?a > priori? what codec will be used. I?m looking for a more drastic solution , > where the answer form FS takes also into account the fmtp from the > negotiated codec . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/ed959d70/attachment-0001.html From acrow at integrafin.co.uk Fri Nov 18 17:42:37 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 18 Nov 2011 14:42:37 +0000 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> <4EC3B7A8.7010306@integrafin.co.uk> Message-ID: <4EC66EDD.80204@integrafin.co.uk> >> Hi all, >> >> I built a new machine for this. I know faxing via audio is not the done >> thing, however this is only to the local Mitel box then out via ISDN until >> we get the licenses and/or E1 cards for both machine. On the new machine I >> built a new kernel without tickless and with 1000Hz - and lo and behold ECM >> faxing now works! >> >> Cheers >> >> Alex >> > > That's really good to hear! Cheers, I'm just glad it *can* be done, even if it's not advised. I get frustrated when things prove impossible! Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From curriegrad2004 at gmail.com Fri Nov 18 18:20:54 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 18 Nov 2011 07:20:54 -0800 Subject: [Freeswitch-users] High load on database server In-Reply-To: References: <6120aaed-bd6b-46b1-a853-a17ab10244b5@Raimund-ThinkPad-X61s> Message-ID: FreeSWITCH at the moment compiles fine with iODBC and you might give that a try if you want to see if that does anything On Fri, Nov 18, 2011 at 4:00 AM, Raimund Sacherer wrote: > Hello, > > I have had lot's of issues with 2.14 odbc as well, the Threadong parameter helped to reduce the crashes from approx. 4 a day to 1 a day, but the crashing was still there. > > I compiled my own debian packages for odbc with the 2.11 centos 5 sources (so I can install other debiacn packages that depend on odbc) and this works like a charm! > > So I would recommend using 2.11 instead of 2.14, I do not have expierence with newer odbc's though. > > using postgres 9.(0|1), 8.4 and debian squeeze with fairly recent Freeswitch boxes > > > best > > > > ----- Original Message ----- > From: "Cliff Wells" > To: "FreeSWITCH Users Help" > Sent: Thursday, October 27, 2011 8:02:46 AM > Subject: Re: [Freeswitch-users] High load on database server > > On Wed, 2011-10-26 at 11:12 -0500, Anthony Minessale wrote: >> if you have older unixodbc edit /etc/odbcinst.ini and add the param >> Threading=0 to each db entry. > >> In general postgres to me appears to care more about multithreaded >> environments and with the above threading=0 change works pretty well. > > I tried this (after googling a bit and coming across an older thread you > responded to) and it didn't change anything. ?Crashes at ~330 concurrent > calls like clockwork. ?This was with unixODBC 2.14 and PostgreSQL 8.4 > odbc driver. > > Regards, > Cliff > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > Logitravel.com > Raimund Sacherer > Sistemas > Agencia de Viajes Online > www.logitravel.com > Edificio Logitravel, Parcela 3B (Parc Bit) > Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca > Tel 902 366 847 | Fax 971 213 495 > S?guenos en: ? ?Facebook de Logitravel ?Twitter de Logitravel ? Blog de Logitravel ? ? ?Logitravel en Youtube ? Logitravel en Foursquare ? ? ? ?Descarga nuestras aplicaciones para m?vil ? ? ? Logitravel.com > Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por error, se ruega notif?quese inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formar?n parte de un fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que deber? ejercer mediante escrito a la direcci?n anteriormente citada. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 elliott at zoogmedia.com Fri Nov 18 18:33:00 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Fri, 18 Nov 2011 15:33:00 +0000 Subject: [Freeswitch-users] User Directory Message-ID: I have been trying to put a user outside of a group like how it's shown on http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide but when I put a user outside of a group the user cant authenticate. Can anyone see what I'm doing wrong or is there error in the documentation? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/55168b0f/attachment-0001.html From steveu at coppice.org Fri Nov 18 18:42:20 2011 From: steveu at coppice.org (Steve Underwood) Date: Fri, 18 Nov 2011 23:42:20 +0800 Subject: [Freeswitch-users] {Solved} Anyone using t38modem with FS and Hylafax. In-Reply-To: References: <4EBAC3D5.5020407@integrafin.co.uk> <2B91BC36-0533-46CB-9E7F-811E038FE654@freeswitch.org> <4EBAF0B7.6030703@integrafin.co.uk> <16658018-A64C-425F-99B8-1261D3E0272D@freeswitch.org> <4EBB7D17.9080208@integrafin.co.uk> <4EBBDB3B.2070509@coppice.org> <4EBCF3BE.3030009@integrafin.co.uk> <4EBD1A83.2020807@integrafin.co.uk> <4EBD27B0.4000504@integrafin.co.uk> <4EBD4012.7040802@coppice.org> Message-ID: <4EC67CDC.2040202@coppice.org> On 11/14/2011 07:24 PM, Tihomir Culjaga wrote: > > > On Fri, Nov 11, 2011 at 4:32 PM, Steve Underwood > wrote: > > Turning off ECM is equivalent to saying "I use FAX, but I couldn't > give > a damn if they are received". > > > and it always work if the network is stable (you might loose a pixel > here and there...)! If you are running it via open internet, sorry but > try something else (T.38 or switch to e-mail). You cannot expect > pass-through to work reliably in such scenarios. Was that intended to be a joke, or did it just turn out that way? Steve From mstockton at harqen.com Fri Nov 18 18:46:28 2011 From: mstockton at harqen.com (Matt Stockton) Date: Fri, 18 Nov 2011 09:46:28 -0600 Subject: [Freeswitch-users] rtp-timer-name issue and question In-Reply-To: References: Message-ID: This is through Sonus media device by the way. Have been looking at this doc, but still to no avail: http://wiki.freeswitch.org/wiki/RTP_Issues It looks like rtp-timer-name = none is causing me more issues than I thought, so at this point I am trying to set it back to soft and fix the DTMF delay issue. I have a call out to my provider to find out if they have any details. Some media servers they send me to do not have the DTMF delay, and some do....ugh On Wed, Nov 16, 2011 at 9:20 AM, Matt Stockton wrote: > Hi all, > > I am battling the following issue: > > My applications are primarily lua scripts that play files, accept DTMF , > and make db calls and url callbacks to post information. At one point, I > was having issues with an ever increasing delay for DTMF recognition as the > application progressed (e.g. I would use playAndGetDigits, and the DTMF > wouldn't be 'recognized' for a delayed amount of time. This time would > increase and it appeared to be dependent on the number of db calls and url > callbacks I made in the app -- as I made more, the delay would get worse). > I asked around, and ended up changing this setting in my external sip > profile: rtp-timer-name = none (it was soft before). This fixed my issue. > > Now, I am battling a separate issue, and I seem to have isolated it's > cause to setting rtp-timer-name to none instead of soft. The problem now is > that when I want to play an mp3 file, and I use streamFile with shout or > playback with http_get, the beginning of the file is not played at all > (playing starts probably about 200-400ms into the file. This is very > repeatable for me - it's not happening to every streamed file, but it seems > to be very deterministic on while line of lua script it's happening (the > streamFile is happening in a loop, and it seems to always be happening to > the first streamed file). If I change rtp-timer-name to soft, everything > is fine, but then I'll have the first problem I described above. > > So, to fix one problem I need rtp-timer-name = none, for the other > problem, I need rtp-timer-name = soft. I must be doing something wrong > here, any suggestions? > > Thanks, > Matt > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/803cbf62/attachment.html From x.liu at hw.ac.uk Fri Nov 18 20:33:20 2011 From: x.liu at hw.ac.uk (xl127) Date: Fri, 18 Nov 2011 17:33:20 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <4EC5348A.6030409@hw.ac.uk> References: <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> <4EC3D5AD.5070507@hw.ac.uk> <4EC3EA04.7030605@hw.ac.uk> <4EC50BF3.1070905@hw.ac.uk> <4EC5348A.6030409@hw.ac.uk> Message-ID: <4EC696E0.40000@hw.ac.uk> I want to specify for play_and_detect_speech using URI format. I am running a Tomcat server that hosts the grammars and I am sure it can be accessed from local machine or from a different machine. If I use http://localhost:8084/grammars/FS_Nuance/myGrammar.gram or http://127.0.0.1:8084/grammars/FS_Nuance/myGrammar.gram FS got grammar loading failure (MRCP/1.0 1 407 COMPLETE) If I change localhost to my concrete external IP, it works for Nuance server, but not for Voxeo Prophecy server. I also have difficulties to specify URI grammars for detect_speech. Any suggestions please? Thanks, Xing On 17/11/11 16:21, xl127 wrote: > Yeah, I see, it should be set to true there. Thanks for reminding me this. > > > On 17/11/11 15:48, Christopher Rienzo wrote: >> Actually in that silence prompt suggestion I gave, use >> {start-input-timers=true} >> >> >> On Thu, Nov 17, 2011 at 8:37 AM, Christopher Rienzo >> > wrote: >> >> >> >> >> After some more tests, I found followings: >> >> 1. Testing from dialplan, the log output is the string of >> "CRIT ${speech_detect_result}" rather than the recognition >> results. >> >> >> oops >> >> >> 2. Regarding to non barge in, I first send "speak" command >> then send play_and_detect_speech with parameter: >> detect:unimrcp:nuance5-mrcp1-1 >> {start-input-timers=false,no-input-timeout=25000,recognition-timeout=25000}dudeYNNC_Nuance >> >> in which I removed the "say:" part. >> >> But very soon I received event CHANNEL_EXECUTE_COMPLETE >> for play_and_detect_speech before the text playing finishes, >> and speech_detect_result is null (actually the event >> header does not contain this variable) >> >> I tried with "say:" part with empty text like >> "say:unimrcp:en-GB: " but it doesn't work, see issue 3 below. >> >> >> Try with silence as I originally suggested: >> >> silence_stream://1000 detect:unimrcp:nuance5-mrcp1-1 >> {start-input-timers=false,no-input-timeout=25000, >> recognition-timeout=25000}dudeYNNC_Nuance >> >> >> 3. It seems that I can not send "speak" command twice >> before the first one finishes. In my case I run one port TTS >> server on one machine. >> >> If I send the command twice FS will give me Synthesizer >> Error/Invalid TTS Module. >> I thought the TTS request would be queued rather than it >> immediately looks for the TTS resource. >> >> If I send the second command to another TTS machine, no >> error occurs but I can only hear one utterance being spoken, >> it looks like one utteraance was dropped somehow. >> >> >> Wait for speak to finish before starting a new one. >> >> >> On 16/11/11 16:51, xl127 wrote: >>> Hi Christopher, >>> >>> The questions are cleared to me now. Many thanks for your >>> explanations! >>> >>> Best regards, >>> >>> Xing >>> >>> >>> On 16/11/11 15:52, Christopher Rienzo wrote: >>>> >>>> Responses inline >>>> >>>> >>>> Now it works in my ESL app though I am just able to do >>>> one dialogue ( I need to add the event catching for >>>> furthur dialgoues). >>>> >>>> I have a couple of questions here: >>>> >>>> 1. In the first try, my Nuance server was able to be >>>> accessed somehow (FS says the MRCP is not responding in >>>> 5000ms, >>>> something like that), then FS says: [WARNING] >>>> rtsp_client.c:386 () Failed to Connect to RTSP Server >>>> *MailScanner has detected a possible fraud attempt from >>>> "99.185.85.31:554" claiming to be* *MailScanner >>>> warning: numerical links are often malicious:* >>>> 99.185.85.31:554 , >>>> >>>> later FS says: >>>> [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER >>>> channel error! >>>> [ERR] switch_ivr_play_say.c:2439 Invalid TTS module! >>>> >>>> The SYNTHESIZER channel error and Invalid TTS module >>>> error are obvious. >>>> >>>> What I don't understand is why it went to this >>>> stange address: *MailScanner has detected a possible >>>> fraud attempt from "99.185.85.31:554" claiming to be* >>>> *MailScanner warning: numerical links are often >>>> malicious:* 99.185.85.31:554 ? >>>> >>>> >>>> check your unimrcp configuration. Make sure the default >>>> TTS and ASR profiles are set to actual servers. >>>> >>>> 2. I specified TTS engine in play_and_detect_speech as >>>> "say:unimrcp:nuance5-mrcp1-1: the text to speak" >>>> It works though I didn't specify the TTS voice. >>>> >>>> How do I specify the TTS voice? In the mrcp >>>> profile (how?)? or something like: >>>> "say:unimrcp:nuance5-mrcp1-1:Serena: the text >>>> to speak" (this seems not right.) >>>> >>>> >>>> That won't work. Set the tts_engine variable as I >>>> explained previously, or use say:unimrcp:voice:text to >>>> speak with the desired voice and the correct default TTS >>>> profile defined in unimrcp.conf.xml. This is a limitation >>>> of the say: notation. Alternatively, the voice can be >>>> defined with the tts_voice channel variable. >>>> >>>> 3. The barge-in works well, thanks!. Is the barge-in >>>> configurable? In some scenarios, we might not allow >>>> barge-in. >>>> >>>> >>>> If you don't want to barge in, just do "playback (or >>>> speak)" first, then "play_and_detect_speech" with a silence >>>> prompt. >>>> >>>> >>>> 4. How could I get the text which has spoken to the >>>> user when barge-in occurs? >>>> Or Could I get the time when barge-in occurs? If I >>>> know the barge-in time and rough totale time for the >>>> whole text >>>> to be spoken I can figure out the spoken text by >>>> manually checking the recorded audio file later, which >>>> would be painful. >>>> >>>> >>>> If this is necessary, you might want to use the lower-level >>>> functions instead to watch for the begin-speaking event. >>>> >>>> >>>> 5. when I use "speak" and "detect_speech" apps in >>>> ESL, I can catch event: DETECTED_SPEECH and >>>> speech-type: begin-speaking >>>> and "detected-speech", then I do the recognition >>>> results processing. >>>> >>>> The new app play_and_detect_speech seems not >>>> generate these events any more. The way that I can >>>> think of to get the results >>>> is to catch event:CHANNEL_EXECUTE_COMPLETE then >>>> check if >>>> variable_current_application=play_and_detect_speech, >>>> then get >>>> the results from variable_detect_speech_result. >>>> >>>> Is this the proper way to get the results in ESL >>>> app? Or will play_and_detect_speech later on be >>>> consistent with detect_speech >>>> in term of ASR events? >>>> >>>> >>>> play_and_detect_speech is a higher level abstraction to >>>> simplify things. If you want to have more control, go back >>>> to using the ESL events. Reading the code in mod_dptools >>>> and switch_ivr_async will give you hints about how to do it >>>> correctly. >>>> >>>> >>>> 6. I'd like to set start-input-timers=false in the >>>> initial request then start the recognition timers >>>> (start-input-timers=true) >>>> after the TTS finishes. >>>> How possibly could I do this? >>>> >>>> >>>> This is automatically done in the >>>> switch_ivr_play_and_detect_speech() function. You just >>>> need to specify start-input-timers=false in the beginning. >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>> >>> >>> ------------------------------------------------------------------------ >>> >>> Scottish University of the Year 2011-12 *Heriot-Watt >>> University is the Sunday Times >>> Scottish University of the Year 2011-2012* >>> >>> Heriot-Watt University is a Scottish charity >>> registered under charity number SC000278. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> >> >> ------------------------------------------------------------------------ >> >> MailScanner Signature HW *Heriot-Watt University is the >> Sunday Times >> >> Scottish University of the Year 2011-2012 >> * >> >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > ------------------------------------------------------------------------ > > MailScanner Signature HW *Heriot-Watt University is the Sunday Times > Scottish University of the Year 2011-2012* > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/0ea4952d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/0ea4952d/attachment-0003.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/0ea4952d/attachment-0004.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/0ea4952d/attachment-0005.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/0ea4952d/attachment-0001.jpg From mario at ims.bg Fri Nov 18 12:02:20 2011 From: mario at ims.bg (Mario Karakanovski) Date: Fri, 18 Nov 2011 11:02:20 +0200 Subject: [Freeswitch-users] Authenticate text messages Message-ID: Hi everyone, Does somebody know how to do authentication for text messages? I try to set auth-all-packets=true in the sofia profile, but without success. All help is appreciated! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/aac8b748/attachment.html From rml at tollfreeforwarding.com Fri Nov 18 12:58:07 2011 From: rml at tollfreeforwarding.com (RaviRaj Mulasa) Date: Fri, 18 Nov 2011 09:58:07 +0000 Subject: [Freeswitch-users] Loading an IVR menu on the fly via URL Message-ID: <8B94625BC339264DBA61E314BE9EC2CF230EC8A6@EXCH125.IFN.com> Hi FreeSWITCH Enthusiastics/Gurus Requirement 1. Customers can build a custom IVR using a UI. 2. Back end code persists the IVR details in a DB. 3. A URL will be exposed and will return the a XML file in the format as follows 4. Load and start executing the IVR 'CUSTOMER_ID_ivr' from events socket on the fly via HTTP URI. Gave it a shot with current setup ivr.conf.xml The ivr.conf.xml is auto loaded at startup of FreeSWITCH. To load any new IVR menus , we need to call API command reloadxml which reloads the whole XML configs in autload_configs folder. The reloadxml might take long time based on the number of XML files present in the folder "/mnt/<>/ivr_menus" Good /Hope to have for the IVR menu 1. Load /Unload IVR menus on the fly using API command(s) freeswitch at localhost> loadivr << URL to fetch the XML describing the IVR menu(s)>> freeswitch at localhost> unloadivr <> 2. From the event socket SendMsg call-command: execute execute-app-name: ivr execute-app-arg: <>SPACE <> Educated/Wild Guess We might be able to play IVR menu on the fly by combining the mod_xml_curl and dialplan tool - ivr. Please let us know how can we achieve the functionality , we are more interested in the second approach(Event Socket) under Good /Hope to have for the IVR menu. Thanks RaviRaj. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/efbde91e/attachment.html From george.d at telesoftlabs.com Fri Nov 18 16:38:03 2011 From: george.d at telesoftlabs.com (George D'mithrov) Date: Fri, 18 Nov 2011 19:08:03 +0530 Subject: [Freeswitch-users] Call transferring from analog phone is not working (OpenZap) Message-ID: <81BCC027A3DF104B89991D9E3621F9B42EF9FC@tslsrv.TSL.local> Hi all, Am new to freeswitch. I configured freeswitch with OpenZap (analog lines). Now am able to make call between SIP phone and Analog phone. But call transferring is not working from analog phone (am using hook flash, blind transferring), but the same is working from SIP phone. when a call is transferring from the analog line the actual call is get disconnected. Any suggestion is well appreciated, Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/010dc803/attachment.html From frankjr at mcpeekdodge.com Fri Nov 18 19:30:09 2011 From: frankjr at mcpeekdodge.com (Frank Busalacchi) Date: Fri, 18 Nov 2011 16:30:09 +0000 Subject: [Freeswitch-users] Custom Call reporting Message-ID: Hello, and thanks for a great piece of software! I currently have FreeSwitch recording all inbound calls. I'm envisioning writing an intranet webpage that would allow managers to be able to generate a list of hyperlinks to call recordings based on ad hoc criteria such as an extension that was on the call path , CLID of the inbound call, date etc. I'm figuring the best way to do this is to name the inbound call recording by its UUID, and by parsing the logs and extracting the UUID of all calls that match the criteria. The system is currently configured to log through CDR-CSV with default installation settings. When I peruse the logs, I see only the beginning of the phone call (Initial inbound, or initial outbound)....What I don't see is the entire "call path". How do I get the log to show every transfer of the call etc? My thought is that if I could do that, then I could basically grab the UUID of the call, grab all log entries that have the UUID in it, and see every extension that was involved with that UUID. Am I on the right path? I'm tinkering with a production system, so I don't want to mess with settings and interrupt the business during business hours. My guess is that I need to set the CDR-CSV log setting to log both AB instead of the default to get the info I am looking for to show up in the log? Thanks, -Frank From hynek.cihlar at gmail.com Fri Nov 18 20:46:39 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Fri, 18 Nov 2011 18:46:39 +0100 Subject: [Freeswitch-users] Playback race condition Message-ID: Hello all, I got on shaky grounds issuing commands to Freeswitch over ESL. For example, issuing the following commands close enough (on my system at around 100 ms and less) causes troubles. *bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 playback::tone_stream://%(150,4000,425);loops=-1* *<100ms appart* *bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all* *previous playback is not stopped, the following is queued (not played)* *bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 playback::tone_stream://%(1000,4000,425);loops=-1* It looks like the first tone playback is not properly initialized when the break arrives. Waiting for the start of the first tone playback (waiting for the right event) is not a solution, I don't want the tone to be played at all if the events come this close. Any ideas? Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/4864e936/attachment-0001.html From raimund.sacherer at logitravel.com Fri Nov 18 21:21:38 2011 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Fri, 18 Nov 2011 19:21:38 +0100 (CET) Subject: [Freeswitch-users] How to tell Leg A to still go on with it's executable action lists after an transfer/REFER? Message-ID: <6db359ae-fca7-4449-9eb3-62c183b39683@Raimund-ThinkPad-X61s> Hello to all, Is it true that a call looses the rest of the dialplan actions it has currently after I do a transfer/REFER? It seems so. How can I force the execution of more dialplan actions after the transfer/REFER? What it does now is: What works: A -> call in -> B Accepts -> B Talks -> B Hangs up -> A is transferred to IVR What does not work: A -> call in -> B Accepts -> B Talks -> B Transfers/REFER to C -> C Talks -> C Hangs Up -> A should be transferred to IVR It seems that A looses everything it should do after B (or C) hangs up, which had been defined in the dialplan, after it got transfer/REFERED to C. How do I rectify this? exports? variables? ?? Thank you, best regards Ray -- From x.liu at hw.ac.uk Fri Nov 18 22:09:44 2011 From: x.liu at hw.ac.uk (xl127) Date: Fri, 18 Nov 2011 19:09:44 +0000 Subject: [Freeswitch-users] How to implement TTS barge-in using FS ESL In-Reply-To: <4EC696E0.40000@hw.ac.uk> References: <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com> <4EBD70A9.6090505@hw.ac.uk> <4EC1127F.2000504@hw.ac.uk> <4EC1236E.3090402@hw.ac.uk> <4EC14A92.1040704@hw.ac.uk> <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1CB6@ex6.mail.win.hw.ac.uk> <4EC3D5AD.5070507@hw.ac.uk> <4EC3EA04.7030605@hw.ac.uk> <4EC50BF3.1070905@hw.ac.uk> <4EC5348A.6030409@hw.ac.uk> <4EC696E0.40000@hw.ac.uk> Message-ID: <4EC6AD78.7090300@hw.ac.uk> Now I see the reason for localhost not working with MRCP server is that the URI address is sent to the MRCP server rather than used to fetch the grammars in FS side. (if just using grammar name, it seems the grammar is fetched from FS side and sent to MRCP server) so the localhost means the MRCP server, not my FS (with Tomcat server running) machine. The problem for Voxeo Prophecy: From the Prophecy log, it has http:\\x.x.x.x:8084\Iced_FS_AsrSluTts\grammars\FS_Nuance\pizza_order_voxeo.gram MRCPServerAPP/Couldn't process new grammar - Content-Id header is missing. MRCPServerAPP/Failed to start speech recognition. Could not process the grammar specified. It looks like the http URI was translated into a format of a file directory. Is this because of FS or because of Voxeo Prophecy MRCP Server? On 18/11/11 17:33, xl127 wrote: > I want to specify for play_and_detect_speech using URI format. > > I am running a Tomcat server that hosts the grammars and I am sure it > can be accessed from local machine or from a different machine. > > If I use http://localhost:8084/grammars/FS_Nuance/myGrammar.gram > or http://127.0.0.1:8084/grammars/FS_Nuance/myGrammar.gram > > FS got grammar loading failure (MRCP/1.0 1 407 COMPLETE) > > If I change localhost to my concrete external IP, it works for Nuance > server, but not for Voxeo Prophecy server. > > I also have difficulties to specify URI grammars for detect_speech. > > Any suggestions please? > > Thanks, > Xing > > > On 17/11/11 16:21, xl127 wrote: >> Yeah, I see, it should be set to true there. Thanks for reminding me >> this. >> >> >> On 17/11/11 15:48, Christopher Rienzo wrote: >>> Actually in that silence prompt suggestion I gave, use >>> {start-input-timers=true} >>> >>> >>> On Thu, Nov 17, 2011 at 8:37 AM, Christopher Rienzo >>> > wrote: >>> >>> >>> >>> >>> After some more tests, I found followings: >>> >>> 1. Testing from dialplan, the log output is the string of >>> "CRIT ${speech_detect_result}" rather than the recognition >>> results. >>> >>> >>> oops >>> >>> >>> 2. Regarding to non barge in, I first send "speak" command >>> then send play_and_detect_speech with parameter: >>> detect:unimrcp:nuance5-mrcp1-1 >>> {start-input-timers=false,no-input-timeout=25000,recognition-timeout=25000}dudeYNNC_Nuance >>> >>> in which I removed the "say:" part. >>> >>> But very soon I received event >>> CHANNEL_EXECUTE_COMPLETE for play_and_detect_speech before >>> the text playing finishes, >>> and speech_detect_result is null (actually the event >>> header does not contain this variable) >>> >>> I tried with "say:" part with empty text like >>> "say:unimrcp:en-GB: " but it doesn't work, see issue 3 below. >>> >>> >>> Try with silence as I originally suggested: >>> >>> silence_stream://1000 detect:unimrcp:nuance5-mrcp1-1 >>> {start-input-timers=false,no-input-timeout=25000, >>> recognition-timeout=25000}dudeYNNC_Nuance >>> >>> >>> 3. It seems that I can not send "speak" command twice >>> before the first one finishes. In my case I run one port TTS >>> server on one machine. >>> >>> If I send the command twice FS will give me Synthesizer >>> Error/Invalid TTS Module. >>> I thought the TTS request would be queued rather than >>> it immediately looks for the TTS resource. >>> >>> If I send the second command to another TTS machine, no >>> error occurs but I can only hear one utterance being spoken, >>> it looks like one utteraance was dropped somehow. >>> >>> >>> Wait for speak to finish before starting a new one. >>> >>> >>> On 16/11/11 16:51, xl127 wrote: >>>> Hi Christopher, >>>> >>>> The questions are cleared to me now. Many thanks for your >>>> explanations! >>>> >>>> Best regards, >>>> >>>> Xing >>>> >>>> >>>> On 16/11/11 15:52, Christopher Rienzo wrote: >>>>> >>>>> Responses inline >>>>> >>>>> >>>>> Now it works in my ESL app though I am just able to do >>>>> one dialogue ( I need to add the event catching for >>>>> furthur dialgoues). >>>>> >>>>> I have a couple of questions here: >>>>> >>>>> 1. In the first try, my Nuance server was able to be >>>>> accessed somehow (FS says the MRCP is not responding >>>>> in 5000ms, >>>>> something like that), then FS says: [WARNING] >>>>> rtsp_client.c:386 () Failed to Connect to RTSP Server >>>>> *MailScanner has detected a possible fraud attempt >>>>> from "99.185.85.31:554" claiming to be* *MailScanner >>>>> warning: numerical links are often malicious:* >>>>> 99.185.85.31:554 , >>>>> >>>>> later FS says: >>>>> [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER >>>>> channel error! >>>>> [ERR] switch_ivr_play_say.c:2439 Invalid TTS module! >>>>> >>>>> The SYNTHESIZER channel error and Invalid TTS >>>>> module error are obvious. >>>>> >>>>> What I don't understand is why it went to this >>>>> stange address: *MailScanner has detected a possible >>>>> fraud attempt from "99.185.85.31:554" claiming to be* >>>>> *MailScanner warning: numerical links are often >>>>> malicious:* 99.185.85.31:554 ? >>>>> >>>>> >>>>> check your unimrcp configuration. Make sure the default >>>>> TTS and ASR profiles are set to actual servers. >>>>> >>>>> 2. I specified TTS engine in play_and_detect_speech as >>>>> "say:unimrcp:nuance5-mrcp1-1: the text to speak" >>>>> It works though I didn't specify the TTS voice. >>>>> >>>>> How do I specify the TTS voice? In the mrcp >>>>> profile (how?)? or something like: >>>>> "say:unimrcp:nuance5-mrcp1-1:Serena: the text >>>>> to speak" (this seems not right.) >>>>> >>>>> >>>>> That won't work. Set the tts_engine variable as I >>>>> explained previously, or use say:unimrcp:voice:text to >>>>> speak with the desired voice and the correct default TTS >>>>> profile defined in unimrcp.conf.xml. This is a limitation >>>>> of the say: notation. Alternatively, the voice can be >>>>> defined with the tts_voice channel variable. >>>>> >>>>> 3. The barge-in works well, thanks!. Is the barge-in >>>>> configurable? In some scenarios, we might not allow >>>>> barge-in. >>>>> >>>>> >>>>> If you don't want to barge in, just do "playback (or >>>>> speak)" first, then "play_and_detect_speech" with a >>>>> silence prompt. >>>>> >>>>> >>>>> 4. How could I get the text which has spoken to the >>>>> user when barge-in occurs? >>>>> Or Could I get the time when barge-in occurs? If >>>>> I know the barge-in time and rough totale time for the >>>>> whole text >>>>> to be spoken I can figure out the spoken text by >>>>> manually checking the recorded audio file later, which >>>>> would be painful. >>>>> >>>>> >>>>> If this is necessary, you might want to use the >>>>> lower-level functions instead to watch for the >>>>> begin-speaking event. >>>>> >>>>> >>>>> 5. when I use "speak" and "detect_speech" apps in >>>>> ESL, I can catch event: DETECTED_SPEECH and >>>>> speech-type: begin-speaking >>>>> and "detected-speech", then I do the recognition >>>>> results processing. >>>>> >>>>> The new app play_and_detect_speech seems not >>>>> generate these events any more. The way that I can >>>>> think of to get the results >>>>> is to catch event:CHANNEL_EXECUTE_COMPLETE then >>>>> check if >>>>> variable_current_application=play_and_detect_speech, >>>>> then get >>>>> the results from variable_detect_speech_result. >>>>> >>>>> Is this the proper way to get the results in ESL >>>>> app? Or will play_and_detect_speech later on be >>>>> consistent with detect_speech >>>>> in term of ASR events? >>>>> >>>>> >>>>> play_and_detect_speech is a higher level abstraction to >>>>> simplify things. If you want to have more control, go >>>>> back to using the ESL events. Reading the code in >>>>> mod_dptools and switch_ivr_async will give you hints about >>>>> how to do it correctly. >>>>> >>>>> >>>>> 6. I'd like to set start-input-timers=false in the >>>>> initial request then start the recognition timers >>>>> (start-input-timers=true) >>>>> after the TTS finishes. >>>>> How possibly could I do this? >>>>> >>>>> >>>>> This is automatically done in the >>>>> switch_ivr_play_and_detect_speech() function. You just >>>>> need to specify start-input-timers=false in the beginning. >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> Scottish University of the Year 2011-12 *Heriot-Watt >>>> University is the Sunday Times >>>> Scottish University of the Year 2011-2012* >>>> >>>> Heriot-Watt University is a Scottish charity >>>> registered under charity number SC000278. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>> >>> >>> ------------------------------------------------------------------------ >>> >>> MailScanner Signature HW *Heriot-Watt University is the >>> Sunday Times >>> >>> Scottish University of the Year 2011-2012 >>> * >>> >>> Heriot-Watt University is a Scottish charity >>> registered under charity number SC000278. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> >> >> ------------------------------------------------------------------------ >> >> MailScanner Signature HW *Heriot-Watt University is the Sunday Times >> Scottish University of the Year 2011-2012* >> >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > ------------------------------------------------------------------------ > > MailScanner Signature HW *Heriot-Watt University is the Sunday Times > Scottish University of the Year 2011-2012* > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/8dfa726c/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/8dfa726c/attachment-0004.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/8dfa726c/attachment-0005.jpe -------------- next part -------------- A non-text attachment was scrubbed... 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Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/8dfa726c/attachment-0001.jpg From adam at sherman.ca Fri Nov 18 22:31:00 2011 From: adam at sherman.ca (Adam Sherman) Date: Fri, 18 Nov 2011 14:31:00 -0500 Subject: [Freeswitch-users] Transcoding and TLS Handling Proxy Message-ID: <053D3A9A-7C6B-4283-9CA8-F22E73D90BD3@sherman.ca> Good Morning, I have an existing SIP-based hosted PBX service running on a commercial platform. To handle some edge cases, I would like to setup a proxy to achieve two purposes: transcoding from iLBC, SILK & G.722 to G.711 and offering SIP over TLS and SRTP to customers that require it. I was originally looking to see if OpenSIPS could do this, but I now believe this is better achieved by a B2BUA and thus FreeSWITCH (please comment on that logic). Some key points I have found, so far: 1. Since the proxy will be transcoding and/or handling SRTP streams, it will also need to handle NAT traversal; 2. It should be completely stateless, I think; 3. UACs will be registering through it (they will be configured with it as an outbound proxy); 4. If one UAC ends up being connected to another UAC (e.g. extension to extension dialling), we want the SRTP stream to simply be relayed; 5. This proxy will simply relay all signalling and decrypted/transcoded media to the existing switch; 6. If I run into scaling issues, I can use Sangoma's hardware transcoding solution with FreeSWITCH; It would be great to hear all your comments and suggestions of what components and modules to use and any other tidbits you can provide. Also, if anyone has done something like this already and wants to offer commercial services to replicate it for me, send me an email. Thanks, A. -- Adam Sherman Technologist Deputy SARCOM, SAR Global 1 Coordonnateur de L'AQBRS, r?gion 07 Outaouais +1 613 797 6819 From msc at freeswitch.org Sat Nov 19 00:15:29 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Nov 2011 13:15:29 -0800 Subject: [Freeswitch-users] Create User xml file automatically (directory/default/called_no.xml) In-Reply-To: References: Message-ID: I would use mod_xml_curl. See the wiki for more information - there is a lot of it. See also the freeswitch-contrib repo, specifically intralanman's stuff. He has examples. -MC On Thu, Nov 17, 2011 at 1:50 AM, Sanath Prasanna wrote: > Hi all, > I need to do following in voicemail module of FS. > when called number is not available, call is routed to voicemail. I did > that using dial plan.But system error coming as "user not exist". When I > create directory/default/called_no.xml file manually, it is not coming. I > need to create that file automatically for each called number, which is > not available. How to do that ? > Br, > Sanath > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111118/acccaaea/attachment.html From brian at freeswitch.org Sat Nov 19 00:56:14 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Nov 2011 15:56:14 -0600 Subject: [Freeswitch-users] Transcoding and TLS Handling Proxy In-Reply-To: <053D3A9A-7C6B-4283-9CA8-F22E73D90BD3@sherman.ca> References: <053D3A9A-7C6B-4283-9CA8-F22E73D90BD3@sherman.ca> Message-ID: <08C8D265-9462-4BB5-BA99-7004E01AC1A6@freeswitch.org> All you need is Client -> SRTP/TLS -> FS -> (Decrypted) PROXY what ever. FS will be your registrar, transcoding.. don't over complicate it. :P /b From adam at sherman.ca Sat Nov 19 01:00:11 2011 From: adam at sherman.ca (Adam Sherman) Date: Fri, 18 Nov 2011 17:00:11 -0500 Subject: [Freeswitch-users] Transcoding and TLS Handling Proxy In-Reply-To: <08C8D265-9462-4BB5-BA99-7004E01AC1A6@freeswitch.org> References: <053D3A9A-7C6B-4283-9CA8-F22E73D90BD3@sherman.ca> <08C8D265-9462-4BB5-BA99-7004E01AC1A6@freeswitch.org> Message-ID: <247586D5-BC68-4431-A6D4-F70F9443C09A@sherman.ca> On 2011-11-18, at 4:56 PM, Brian West wrote: > > Client -> SRTP/TLS -> FS -> (Decrypted) PROXY what ever. > > FS will be your registrar, transcoding.. don't over complicate it. :P I don't want to manage another user list, I want the registrations to be proxy'd, this won't work? A. -- Adam Sherman Technologist Deputy SARCOM, SAR Global 1 Coordonnateur de L'AQBRS, r?gion 07 Outaouais +1 613 797 6819 From brian at freeswitch.org Sat Nov 19 01:05:22 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Nov 2011 16:05:22 -0600 Subject: [Freeswitch-users] Transcoding and TLS Handling Proxy In-Reply-To: <247586D5-BC68-4431-A6D4-F70F9443C09A@sherman.ca> References: <053D3A9A-7C6B-4283-9CA8-F22E73D90BD3@sherman.ca> <08C8D265-9462-4BB5-BA99-7004E01AC1A6@freeswitch.org> <247586D5-BC68-4431-A6D4-F70F9443C09A@sherman.ca> Message-ID: Nope you can email consulting at freeswitch.org if you want to have something like that added but currently we are NOT a proxy. We are in some ways quasi proxy / b2bua in some functions... but not registration. /b On Nov 18, 2011, at 4:00 PM, Adam Sherman wrote: > I don't want to manage another user list, I want the registrations to be proxy'd, this won't work? > > A. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/35f46194/attachment.html From kris at kriskinc.com Sat Nov 19 02:19:09 2011 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 18 Nov 2011 18:19:09 -0500 Subject: [Freeswitch-users] Transcoding and TLS Handling Proxy In-Reply-To: <053D3A9A-7C6B-4283-9CA8-F22E73D90BD3@sherman.ca> References: <053D3A9A-7C6B-4283-9CA8-F22E73D90BD3@sherman.ca> Message-ID: Adam, A few (random) comments: - If you don't want to maintain two separate user lists you could frontend a FreeSWITCH instance (or multiple FreeSWITCH instances) and your commercial system with OpenSIPS or Kamailio and use the path module for registrations to your existing commercial system: http://www.opensips.org/html/docs/modules/1.7.x/path.html http://www.ietf.org/rfc/rfc3327.txt Your existing commercial system will need to support path (RFC 3327). This config will also end up getting pretty complicated as you will need to configure OpenSIPS to relay traffic between the systems supporting various encryption schemes and codecs to make sure endpoints are compatible: Endpoint (REGISTER TLS) -> OpenSIPS -> Commercial system (UDP/TCP) Endpoint (INVITE TLS w/ G.722/SILK/iLBC) -> OpenSIPS -> FreeSWITCH -> Commercial system (UDP/TCP G711) Commercial system (INVITE UDP/TCP G711) -> OpenSIPS/FreeSWITCH -> Endpoint (TLS w/ G.722/SILK/iLBC) Make no mistake: this is very complex. - The Sangoma hardware doesn't currently support SILK and last time I checked there is no planned support for it. SILK will most likely be supplanted by an IETF standard Opus once it's finalized anyway. I think only then will we start to see more "SILK" (really Opus) implementations. What are you looking for from SILK? - Depending on what kind of API and/or provisioning system is available on your commercial system you may be able to use mod_xml_curl and/or one of the OpenSIPS DB backends to access subscriber information on your commercial system. On Fri, Nov 18, 2011 at 2:31 PM, Adam Sherman wrote: > Good Morning, > > I have an existing SIP-based hosted PBX service running on a commercial platform. To handle some edge cases, I would like to setup a proxy to achieve two purposes: transcoding from iLBC, SILK & G.722 to G.711 and offering SIP over TLS and SRTP to customers that require it. I was originally looking to see if OpenSIPS could do this, but I now believe this is better achieved by a B2BUA and thus FreeSWITCH (please comment on that logic). > > Some key points I have found, so far: > > 1. Since the proxy will be transcoding and/or handling SRTP streams, it will also need to handle NAT traversal; > > 2. It should be completely stateless, I think; > > 3. UACs will be registering through it (they will be configured with it as an outbound proxy); > > 4. If one UAC ends up being connected to another UAC (e.g. extension to extension dialling), we want the SRTP stream to simply be relayed; > > 5. This proxy will simply relay all signalling and decrypted/transcoded media to the existing switch; > > 6. If I run into scaling issues, I can use Sangoma's hardware transcoding solution with FreeSWITCH; > > It would be great to hear all your comments and suggestions of what components and modules to use and any other tidbits you can provide. > > Also, if anyone has done something like this already and wants to offer commercial services to replicate it for me, send me an email. > > Thanks, > > A. > > -- > Adam Sherman > Technologist > Deputy SARCOM, SAR Global 1 > Coordonnateur de L'AQBRS, r?gion 07 Outaouais > +1 613 797 6819 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 From msc at freeswitch.org Sat Nov 19 03:50:44 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Nov 2011 16:50:44 -0800 Subject: [Freeswitch-users] Custom Call reporting In-Reply-To: References: Message-ID: Well, this is an interesting question. If it's a simple call A > B then you only have one uuid to worry about and you're good. But what if a transfer is involved? Now it gets hairy. Add in b leg uuid logging and it gets even hairier. Throw in blind vs. attended transfer and now you're talking 80's rock band hairy. One option is to use xml cdrs - they give you the whole picture. I personally like them a lot. Another is to attempt to write some logic that would take the main a-leg uuid and find all related uuid's. It's not nearly as easy as you'd think. Where are you storing your CDRs? -MC On Fri, Nov 18, 2011 at 8:30 AM, Frank Busalacchi wrote: > Hello, and thanks for a great piece of software! > > I currently have FreeSwitch recording all inbound calls. I'm envisioning > writing an intranet webpage that would allow managers to be able to > generate a list of hyperlinks to call recordings based on ad hoc criteria > such as an extension that was on the call path , CLID of the inbound call, > date etc. > > I'm figuring the best way to do this is to name the inbound call recording > by its UUID, and by parsing the logs and extracting the UUID of all calls > that match the criteria. > > The system is currently configured to log through CDR-CSV with default > installation settings. When I peruse the logs, I see only the beginning of > the phone call (Initial inbound, or initial outbound)....What I don't see > is the entire "call path". How do I get the log to show every transfer of > the call etc? My thought is that if I could do that, then I could > basically grab the UUID of the call, grab all log entries that have the > UUID in it, and see every extension that was involved with that UUID. Am I > on the right path? I'm tinkering with a production system, so I don't want > to mess with settings and interrupt the business during business hours. My > guess is that I need to set the CDR-CSV log setting to log both AB instead > of the default to get the info I am looking for to show up in the log? > > Thanks, > > -Frank > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111118/8403b0f1/attachment-0001.html From david.villasmil.work at gmail.com Sat Nov 19 06:12:47 2011 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 19 Nov 2011 04:12:47 +0100 Subject: [Freeswitch-users] Authorization User Message-ID: Hello Guys, When I create a GW it MUST have a username and password in the sofia config, so i set a generic one like "123" Problem is, i need to send out in the FROM the callerid of the actual caller... I've tried setting (lua) on the outgoing call: session:execute("set","sip_from_user=" .. caller_id .. "") session:execute("set","sip_auth_username=" .. caller_id .. "") session:execute("set","sip_req_user=" .. caller_id .. "") But still the only thing that is correctly set is: Remote-Party-ID: "David" ;party=calling;screen=yes;privacy=off [Expert Info (Note/Undecoded): Unrecognised SIP header (Remote-Party-ID)] [Message: Unrecognised SIP header (Remote-Party-ID)] [Severity level: Note] [Group: Undecoded] From: "David" ;tag=jtr1yQHFUUp3H SIP Display info: "David" SIP from address: sip:123 at 1.2.3.4:5060 SIP from address User Part: 123 SIP from address Host Part: 1.2.3.4 SIP from address Host Port: 5060 SIP tag: jtr1yQHFUUp3H I would need to set the FROM correctly... Any thoughts? Thanks David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111119/e7ad57ea/attachment.html From mstockton at harqen.com Sat Nov 19 08:59:14 2011 From: mstockton at harqen.com (Matt Stockton) Date: Fri, 18 Nov 2011 23:59:14 -0600 Subject: [Freeswitch-users] rtp-timer-name issue and question In-Reply-To: References: Message-ID: Just as an FYI, using s session:sleep(0,1) seems to solve my issues, it removes the delays. I can keep the rtp-timer-name as soft. Found info about it here: http://jira.freeswitch.org/browse/FS-1360 On Fri, Nov 18, 2011 at 9:46 AM, Matt Stockton wrote: > This is through Sonus media device by the way. Have been looking at this > doc, but still to no avail: http://wiki.freeswitch.org/wiki/RTP_Issues > > It looks like rtp-timer-name = none is causing me more issues than I > thought, so at this point I am trying to set it back to soft and fix the > DTMF delay issue. I have a call out to my provider to find out if they have > any details. Some media servers they send me to do not have the DTMF delay, > and some do....ugh > > On Wed, Nov 16, 2011 at 9:20 AM, Matt Stockton wrote: > >> Hi all, >> >> I am battling the following issue: >> >> My applications are primarily lua scripts that play files, accept DTMF , >> and make db calls and url callbacks to post information. At one point, I >> was having issues with an ever increasing delay for DTMF recognition as the >> application progressed (e.g. I would use playAndGetDigits, and the DTMF >> wouldn't be 'recognized' for a delayed amount of time. This time would >> increase and it appeared to be dependent on the number of db calls and url >> callbacks I made in the app -- as I made more, the delay would get worse). >> I asked around, and ended up changing this setting in my external sip >> profile: rtp-timer-name = none (it was soft before). This fixed my issue. >> >> Now, I am battling a separate issue, and I seem to have isolated it's >> cause to setting rtp-timer-name to none instead of soft. The problem now is >> that when I want to play an mp3 file, and I use streamFile with shout or >> playback with http_get, the beginning of the file is not played at all >> (playing starts probably about 200-400ms into the file. This is very >> repeatable for me - it's not happening to every streamed file, but it seems >> to be very deterministic on while line of lua script it's happening (the >> streamFile is happening in a loop, and it seems to always be happening to >> the first streamed file). If I change rtp-timer-name to soft, everything >> is fine, but then I'll have the first problem I described above. >> >> So, to fix one problem I need rtp-timer-name = none, for the other >> problem, I need rtp-timer-name = soft. I must be doing something wrong >> here, any suggestions? >> >> Thanks, >> Matt >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111118/9cdf8bb1/attachment.html From hynek.cihlar at gmail.com Sat Nov 19 10:31:53 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sat, 19 Nov 2011 08:31:53 +0100 Subject: [Freeswitch-users] Playback race condition In-Reply-To: References: Message-ID: As a workaround I put a synthetic a delay between the start and stop of the playback. I wouldn't want to keep this solution however since it blocks one of the worker threads (decreasing overall system throughput) and second, I really don't know what the synthetic delay should be or whether it could differ during different conditions like during a high load. Was anybody facing similar problem? How did you solve it? Am I using the API wrong and is there another way to invoke playback/stop? Thanks! Hynek On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar wrote: > Hello all, > > I got on shaky grounds issuing commands to Freeswitch over ESL. > > For example, issuing the following commands close enough (on my system at > around 100 ms and less) causes troubles. > > *bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 > playback::tone_stream://%(150,4000,425);loops=-1* > *<100ms appart* > *bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all* > *previous playback is not stopped, the following is queued (not played)* > *bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 > playback::tone_stream://%(1000,4000,425);loops=-1* > > It looks like the first tone playback is not properly initialized when the > break arrives. Waiting for the start of the first tone playback (waiting > for the right event) is not a solution, I don't want the tone to be played > at all if the events come this close. > > Any ideas? > > > Hynek > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111119/3ed25e41/attachment.html From peter.olsson at visionutveckling.se Sat Nov 19 10:48:17 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 19 Nov 2011 08:48:17 +0100 Subject: [Freeswitch-users] Playback race condition In-Reply-To: References: , Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> Since you're using bgapi, it's impossible to guarantee that uuid_broadcast has actually executed before you exeute uuid_break (they will be executed in two different threads). You will have to do this in a more controlled way (wait for events to show up etc.). And also, do you really need to use bgapi all the time? I'm not sure how uuid_broadcast is handled, but uuid_break is totally safe to use without bgapi, since it will just update a flag and then return immediately. I think uuid_broadcast will do the same thing - so getting rid of bgapi should be a good ebnough solution. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [hynek.cihlar at gmail.com] Skickat: den 19 november 2011 08:31 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Playback race condition As a workaround I put a synthetic a delay between the start and stop of the playback. I wouldn't want to keep this solution however since it blocks one of the worker threads (decreasing overall system throughput) and second, I really don't know what the synthetic delay should be or whether it could differ during different conditions like during a high load. Was anybody facing similar problem? How did you solve it? Am I using the API wrong and is there another way to invoke playback/stop? Thanks! Hynek On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar > wrote: Hello all, I got on shaky grounds issuing commands to Freeswitch over ESL. For example, issuing the following commands close enough (on my system at around 100 ms and less) causes troubles. bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 playback::tone_stream://%(150,4000,425);loops=-1 <100ms appart bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all previous playback is not stopped, the following is queued (not played) bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 playback::tone_stream://%(1000,4000,425);loops=-1 It looks like the first tone playback is not properly initialized when the break arrives. Waiting for the start of the first tone playback (waiting for the right event) is not a solution, I don't want the tone to be played at all if the events come this close. Any ideas? Hynek !DSPAM:4ec75b5732763454765634! From hynek.cihlar at gmail.com Sat Nov 19 11:12:41 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sat, 19 Nov 2011 09:12:41 +0100 Subject: [Freeswitch-users] Playback race condition In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> Message-ID: Two reasons I am not using api in this case. 1. Performance 2. Doesn't seem to work 1. Blocking issuing the command will eventually reduce the system throughput. Agree this is only a potential theoretical problem and I haven't gotten into production where I could get the actual results. 2. When issuing api uuid_playback and api uuid_break right after, I'm always getting an error response from the uuid_break. The ESL message body is "-ERR no reply". Nothing in the logs (no idea which log type/level will could generate other useful info). Hynek On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Since you're using bgapi, it's impossible to guarantee that uuid_broadcast > has actually executed before you exeute uuid_break (they will be executed > in two different threads). You will have to do this in a more controlled > way (wait for events to show up etc.). And also, do you really need to use > bgapi all the time? I'm not sure how uuid_broadcast is handled, but > uuid_break is totally safe to use without bgapi, since it will just update > a flag and then return immediately. I think uuid_broadcast will do the same > thing - so getting rid of bgapi should be a good ebnough solution. > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [ > hynek.cihlar at gmail.com] > Skickat: den 19 november 2011 08:31 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Playback race condition > > As a workaround I put a synthetic a delay between the start and stop of > the playback. I wouldn't want to keep this solution however since it blocks > one of the worker threads (decreasing overall system throughput) and > second, I really don't know what the synthetic delay should be or whether > it could differ during different conditions like during a high load. > > Was anybody facing similar problem? How did you solve it? Am I using the > API wrong and is there another way to invoke playback/stop? > > Thanks! > Hynek > > > > On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar > wrote: > Hello all, > > I got on shaky grounds issuing commands to Freeswitch over ESL. > > For example, issuing the following commands close enough (on my system at > around 100 ms and less) causes troubles. > > bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 > playback::tone_stream://%(150,4000,425);loops=-1 > <100ms appart > bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all > previous playback is not stopped, the following is queued (not played) > bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 > playback::tone_stream://%(1000,4000,425);loops=-1 > > It looks like the first tone playback is not properly initialized when the > break arrives. Waiting for the start of the first tone playback (waiting > for the right event) is not a solution, I don't want the tone to be played > at all if the events come this close. > > Any ideas? > > > Hynek > > > !DSPAM:4ec75b5732763454765634! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111119/02936817/attachment-0001.html From peter.olsson at visionutveckling.se Sat Nov 19 11:35:47 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 19 Nov 2011 09:35:47 +0100 Subject: [Freeswitch-users] Playback race condition In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper>, Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> If you think about performance I would say it's better to handle this without spawning lots of extra threads. First of all, most common commands can be queued into the the channel thread using the execute method (http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes it possible to queue commands directly to the channel's thread, and the ESL session will never wait for the result, it returns immediately. I've never used uuid_broadcast myself, so I'm not really sure how it's handled, but by looking quickly in the code it looks like it shouldn't block either, since it's actually just queuing a playback command to the channel's thread. So basically, keep away from uuid_X-commands, and use execute method as much as possible, and when using uuid_x-methods, make sure that won't block - but if they do, make sure to handle handle the events for the bgapi correctly. I've built a complete IVR system using these methods, with a single ESL connection, and I've never had any performace issues. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [hynek.cihlar at gmail.com] Skickat: den 19 november 2011 09:12 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Playback race condition Two reasons I am not using api in this case. 1. Performance 2. Doesn't seem to work 1. Blocking issuing the command will eventually reduce the system throughput. Agree this is only a potential theoretical problem and I haven't gotten into production where I could get the actual results. 2. When issuing api uuid_playback and api uuid_break right after, I'm always getting an error response from the uuid_break. The ESL message body is "-ERR no reply". Nothing in the logs (no idea which log type/level will could generate other useful info). Hynek On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson > wrote: Since you're using bgapi, it's impossible to guarantee that uuid_broadcast has actually executed before you exeute uuid_break (they will be executed in two different threads). You will have to do this in a more controlled way (wait for events to show up etc.). And also, do you really need to use bgapi all the time? I'm not sure how uuid_broadcast is handled, but uuid_break is totally safe to use without bgapi, since it will just update a flag and then return immediately. I think uuid_broadcast will do the same thing - so getting rid of bgapi should be a good ebnough solution. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [hynek.cihlar at gmail.com] Skickat: den 19 november 2011 08:31 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Playback race condition As a workaround I put a synthetic a delay between the start and stop of the playback. I wouldn't want to keep this solution however since it blocks one of the worker threads (decreasing overall system throughput) and second, I really don't know what the synthetic delay should be or whether it could differ during different conditions like during a high load. Was anybody facing similar problem? How did you solve it? Am I using the API wrong and is there another way to invoke playback/stop? Thanks! Hynek On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar >> wrote: Hello all, I got on shaky grounds issuing commands to Freeswitch over ESL. For example, issuing the following commands close enough (on my system at around 100 ms and less) causes troubles. bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 playback::tone_stream://%(150,4000,425);loops=-1 <100ms appart bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all previous playback is not stopped, the following is queued (not played) bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 playback::tone_stream://%(1000,4000,425);loops=-1 It looks like the first tone playback is not properly initialized when the break arrives. Waiting for the start of the first tone playback (waiting for the right event) is not a solution, I don't want the tone to be played at all if the events come this close. Any ideas? Hynek _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 !DSPAM:4ec764af32761901519715! From hynek.cihlar at gmail.com Sat Nov 19 12:04:52 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sat, 19 Nov 2011 10:04:52 +0100 Subject: [Freeswitch-users] Playback race condition In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> Message-ID: Peter, thanks for pointing me to the right direction. I will try the execute command, since queueing is exactly what I would need to preserve consistency across commands (I had an impression that bgapi will give me this consistency). Also I like to hear that it is possible to use a single connection since I have designed the communication with one connection also. May I ask how many concurrent sessions can your system handle? Hynek On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > If you think about performance I would say it's better to handle this > without spawning lots of extra threads. > > First of all, most common commands can be queued into the the channel > thread using the execute method ( > http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes it > possible to queue commands directly to the channel's thread, and the ESL > session will never wait for the result, it returns immediately. > > I've never used uuid_broadcast myself, so I'm not really sure how it's > handled, but by looking quickly in the code it looks like it shouldn't > block either, since it's actually just queuing a playback command to the > channel's thread. > > So basically, keep away from uuid_X-commands, and use execute method as > much as possible, and when using uuid_x-methods, make sure that won't block > - but if they do, make sure to handle handle the events for the bgapi > correctly. > > I've built a complete IVR system using these methods, with a single ESL > connection, and I've never had any performace issues. > > /Peter > > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [ > hynek.cihlar at gmail.com] > Skickat: den 19 november 2011 09:12 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Playback race condition > > Two reasons I am not using api in this case. > > 1. Performance > 2. Doesn't seem to work > > 1. Blocking issuing the command will eventually reduce the system > throughput. Agree this is only a potential theoretical problem and I > haven't gotten into production where I could get the actual results. > > 2. When issuing api uuid_playback and api uuid_break right after, I'm > always getting an error response from the uuid_break. The ESL message body > is "-ERR no reply". Nothing in the logs (no idea which log type/level will > could generate other useful info). > > Hynek > > > > On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < > peter.olsson at visionutveckling.se> > wrote: > Since you're using bgapi, it's impossible to guarantee that uuid_broadcast > has actually executed before you exeute uuid_break (they will be executed > in two different threads). You will have to do this in a more controlled > way (wait for events to show up etc.). And also, do you really need to use > bgapi all the time? I'm not sure how uuid_broadcast is handled, but > uuid_break is totally safe to use without bgapi, since it will just update > a flag and then return immediately. I think uuid_broadcast will do the same > thing - so getting rid of bgapi should be a good ebnough solution. > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> [ > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] för Hynek Cihlar [ > hynek.cihlar at gmail.com] > Skickat: den 19 november 2011 08:31 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Playback race condition > > As a workaround I put a synthetic a delay between the start and stop of > the playback. I wouldn't want to keep this solution however since it blocks > one of the worker threads (decreasing overall system throughput) and > second, I really don't know what the synthetic delay should be or whether > it could differ during different conditions like during a high load. > > Was anybody facing similar problem? How did you solve it? Am I using the > API wrong and is there another way to invoke playback/stop? > > Thanks! > Hynek > > > > On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar hynek.cihlar at gmail.com>>> wrote: > Hello all, > > I got on shaky grounds issuing commands to Freeswitch over ESL. > > For example, issuing the following commands close enough (on my system at > around 100 ms and less) causes troubles. > > bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 > playback::tone_stream://%(150,4000,425);loops=-1 > <100ms appart > bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all > previous playback is not stopped, the following is queued (not played) > bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 > playback::tone_stream://%(1000,4000,425);loops=-1 > > It looks like the first tone playback is not properly initialized when the > break arrives. Waiting for the start of the first tone playback (waiting > for the right event) is not a solution, I don't want the tone to be played > at all if the events come this close. > > Any ideas? > > > Hynek > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4ec764af32761901519715! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111119/c94472a0/attachment.html From peter.olsson at visionutveckling.se Sat Nov 19 12:28:05 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 19 Nov 2011 10:28:05 +0100 Subject: [Freeswitch-users] Playback race condition In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper>, Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> bgapi can of course be used also, and in some cases you might still need it. But the important thing to keep in mind when doing bgapi is to handle the events correctly. When executing bgapi you will get a job-id back, this job id must then be stored internally, so you can link this to the background job event. I try to use the sendMsg/execute as much as possible, you will get CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE events, so you will always know what's going on. I've only tested my system for about 100 concurrent calls, but that was handled without any problems at all. What's really important is to make really sure you never execute something that will block - since that will cause everything to "hang". /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [hynek.cihlar at gmail.com] Skickat: den 19 november 2011 10:04 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Playback race condition Peter, thanks for pointing me to the right direction. I will try the execute command, since queueing is exactly what I would need to preserve consistency across commands (I had an impression that bgapi will give me this consistency). Also I like to hear that it is possible to use a single connection since I have designed the communication with one connection also. May I ask how many concurrent sessions can your system handle? Hynek On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson > wrote: If you think about performance I would say it's better to handle this without spawning lots of extra threads. First of all, most common commands can be queued into the the channel thread using the execute method (http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes it possible to queue commands directly to the channel's thread, and the ESL session will never wait for the result, it returns immediately. I've never used uuid_broadcast myself, so I'm not really sure how it's handled, but by looking quickly in the code it looks like it shouldn't block either, since it's actually just queuing a playback command to the channel's thread. So basically, keep away from uuid_X-commands, and use execute method as much as possible, and when using uuid_x-methods, make sure that won't block - but if they do, make sure to handle handle the events for the bgapi correctly. I've built a complete IVR system using these methods, with a single ESL connection, and I've never had any performace issues. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [hynek.cihlar at gmail.com] Skickat: den 19 november 2011 09:12 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Playback race condition Two reasons I am not using api in this case. 1. Performance 2. Doesn't seem to work 1. Blocking issuing the command will eventually reduce the system throughput. Agree this is only a potential theoretical problem and I haven't gotten into production where I could get the actual results. 2. When issuing api uuid_playback and api uuid_break right after, I'm always getting an error response from the uuid_break. The ESL message body is "-ERR no reply". Nothing in the logs (no idea which log type/level will could generate other useful info). Hynek On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson >> wrote: Since you're using bgapi, it's impossible to guarantee that uuid_broadcast has actually executed before you exeute uuid_break (they will be executed in two different threads). You will have to do this in a more controlled way (wait for events to show up etc.). And also, do you really need to use bgapi all the time? I'm not sure how uuid_broadcast is handled, but uuid_break is totally safe to use without bgapi, since it will just update a flag and then return immediately. I think uuid_broadcast will do the same thing - so getting rid of bgapi should be a good ebnough solution. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org>] för Hynek Cihlar [hynek.cihlar at gmail.com>] Skickat: den 19 november 2011 08:31 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Playback race condition As a workaround I put a synthetic a delay between the start and stop of the playback. I wouldn't want to keep this solution however since it blocks one of the worker threads (decreasing overall system throughput) and second, I really don't know what the synthetic delay should be or whether it could differ during different conditions like during a high load. Was anybody facing similar problem? How did you solve it? Am I using the API wrong and is there another way to invoke playback/stop? Thanks! Hynek On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar >>>> wrote: Hello all, I got on shaky grounds issuing commands to Freeswitch over ESL. For example, issuing the following commands close enough (on my system at around 100 ms and less) causes troubles. bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 playback::tone_stream://%(150,4000,425);loops=-1 <100ms appart bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all previous playback is not stopped, the following is queued (not played) bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 playback::tone_stream://%(1000,4000,425);loops=-1 It looks like the first tone playback is not properly initialized when the break arrives. Waiting for the start of the first tone playback (waiting for the right event) is not a solution, I don't want the tone to be played at all if the events come this close. Any ideas? Hynek _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 !DSPAM:4ec770ca32761488716642! From hynek.cihlar at gmail.com Sat Nov 19 12:37:21 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sat, 19 Nov 2011 10:37:21 +0100 Subject: [Freeswitch-users] Playback race condition In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> Message-ID: So I changed the uuid_X-commands to execute and the use case with stopping tone playback works like a charm now! Regarding the blocking, you are right. I'm limiting blocking (and context switching) by handling all events and issuing commands always on one (and the same) thread per session. This design also allows me to simplify the programming model, decrease concurrency bugs, simplify debugging, etc. That's why the synthetic delay was giving me troubles since the events and commands must fly through as fast as possible. Thanks again! Hynek On Sat, Nov 19, 2011 at 10:28 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > bgapi can of course be used also, and in some cases you might still need > it. But the important thing to keep in mind when doing bgapi is to handle > the events correctly. When executing bgapi you will get a job-id back, this > job id must then be stored internally, so you can link this to the > background job event. > > I try to use the sendMsg/execute as much as possible, you will get > CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE events, so you will always know > what's going on. > > I've only tested my system for about 100 concurrent calls, but that was > handled without any problems at all. What's really important is to make > really sure you never execute something that will block - since that will > cause everything to "hang". > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [ > hynek.cihlar at gmail.com] > Skickat: den 19 november 2011 10:04 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Playback race condition > > Peter, thanks for pointing me to the right direction. I will try the > execute command, since queueing is exactly what I would need to preserve > consistency across commands (I had an impression that bgapi will give me > this consistency). > > Also I like to hear that it is possible to use a single connection since I > have designed the communication with one connection also. May I ask how > many concurrent sessions can your system handle? > > Hynek > > > > On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson < > peter.olsson at visionutveckling.se> > wrote: > If you think about performance I would say it's better to handle this > without spawning lots of extra threads. > > First of all, most common commands can be queued into the the channel > thread using the execute method ( > http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes it > possible to queue commands directly to the channel's thread, and the ESL > session will never wait for the result, it returns immediately. > > I've never used uuid_broadcast myself, so I'm not really sure how it's > handled, but by looking quickly in the code it looks like it shouldn't > block either, since it's actually just queuing a playback command to the > channel's thread. > > So basically, keep away from uuid_X-commands, and use execute method as > much as possible, and when using uuid_x-methods, make sure that won't block > - but if they do, make sure to handle handle the events for the bgapi > correctly. > > I've built a complete IVR system using these methods, with a single ESL > connection, and I've never had any performace issues. > > /Peter > > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> [ > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] för Hynek Cihlar [ > hynek.cihlar at gmail.com] > Skickat: den 19 november 2011 09:12 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Playback race condition > > Two reasons I am not using api in this case. > > 1. Performance > 2. Doesn't seem to work > > 1. Blocking issuing the command will eventually reduce the system > throughput. Agree this is only a potential theoretical problem and I > haven't gotten into production where I could get the actual results. > > 2. When issuing api uuid_playback and api uuid_break right after, I'm > always getting an error response from the uuid_break. The ESL message body > is "-ERR no reply". Nothing in the logs (no idea which log type/level will > could generate other useful info). > > Hynek > > > > On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < > peter.olsson at visionutveckling.se > peter.olsson at visionutveckling.se>>> wrote: > Since you're using bgapi, it's impossible to guarantee that uuid_broadcast > has actually executed before you exeute uuid_break (they will be executed > in two different threads). You will have to do this in a more controlled > way (wait for events to show up etc.). And also, do you really need to use > bgapi all the time? I'm not sure how uuid_broadcast is handled, but > uuid_break is totally safe to use without bgapi, since it will just update > a flag and then return immediately. I think uuid_broadcast will do the same > thing - so getting rid of bgapi should be a good ebnough solution. > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>> [ > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>>] för Hynek Cihlar [ > hynek.cihlar at gmail.com hynek.cihlar at gmail.com>] > Skickat: den 19 november 2011 08:31 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Playback race condition > > As a workaround I put a synthetic a delay between the start and stop of > the playback. I wouldn't want to keep this solution however since it blocks > one of the worker threads (decreasing overall system throughput) and > second, I really don't know what the synthetic delay should be or whether > it could differ during different conditions like during a high load. > > Was anybody facing similar problem? How did you solve it? Am I using the > API wrong and is there another way to invoke playback/stop? > > Thanks! > Hynek > > > > On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar hynek.cihlar at gmail.com>> hynek.cihlar at gmail.com> hynek.cihlar at gmail.com>>>> wrote: > Hello all, > > I got on shaky grounds issuing commands to Freeswitch over ESL. > > For example, issuing the following commands close enough (on my system at > around 100 ms and less) causes troubles. > > bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 > playback::tone_stream://%(150,4000,425);loops=-1 > <100ms appart > bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all > previous playback is not stopped, the following is queued (not played) > bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 > playback::tone_stream://%(1000,4000,425);loops=-1 > > It looks like the first tone playback is not properly initialized when the > break arrives. Waiting for the start of the first tone playback (waiting > for the right event) is not a solution, I don't want the tone to be played > at all if the events come this close. > > Any ideas? > > > Hynek > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org consulting at freeswitch.org> > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4ec770ca32761488716642! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111119/9eee4b99/attachment.html From paul at cupis.co.uk Sat Nov 19 13:04:37 2011 From: paul at cupis.co.uk (Paul Cupis) Date: Sat, 19 Nov 2011 10:04:37 +0000 Subject: [Freeswitch-users] Authorization User In-Reply-To: References: Message-ID: <4EC77F35.3060208@cupis.co.uk> On 19/11/11 03:12, David Villasmil wrote: > When I create a GW it MUST have a username and password in the sofia > config, so i set a generic one like "123" > > Problem is, i need to send out in the FROM the callerid of the actual > caller... Try effective_caller_id_number: http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number Regards, From david.villasmil.work at gmail.com Sat Nov 19 17:26:49 2011 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 19 Nov 2011 15:26:49 +0100 Subject: [Freeswitch-users] Authorization User In-Reply-To: <4EC77F35.3060208@cupis.co.uk> References: <4EC77F35.3060208@cupis.co.uk> Message-ID: Thanks for your reply Paul, That didn't do it though... From: "David" ;tag=1Brrjt9Hev49c SIP Display info: "David" SIP from address: sip:123 at 1.2.3.4:5060 SIP from address User Part: 123 SIP from address Host Part: 1.2.3.4 SIP from address Host Port: 5060 SIP tag: 1Brrjt9Hev49c David On Sat, Nov 19, 2011 at 11:04 AM, Paul Cupis wrote: > On 19/11/11 03:12, David Villasmil wrote: > > When I create a GW it MUST have a username and password in the sofia > > config, so i set a generic one like "123" > > > > Problem is, i need to send out in the FROM the callerid of the actual > > caller... > > Try effective_caller_id_number: > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111119/f4d34521/attachment.html From fieldpeak at gmail.com Sat Nov 19 17:47:20 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Sat, 19 Nov 2011 22:47:20 +0800 Subject: [Freeswitch-users] FS compile error Message-ID: Dear fridends, i copy a FS source code (*FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400*) from a 32bit PC to a 64bit PC, on 32bit PC it compiles well, but on 64bit PC, there were below errors when compiling, could anyone help advise? Thanks. making all mod_hash /usr/bin/ld: skipping incompatible ./libesl.a when searching for -lesl /usr/bin/ld: cannot find -lesl collect2: ld returned 1 exit status make[5]: *** [fs_cli] Error 1 make[4]: *** [/usr/local/src/freeswitch/libs/esl/libesl.so] Error 2 make[3]: *** [mod_hash-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 [root at freeswitch freeswitch]# -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111119/7ac06fe2/attachment.html From jan.berger at video24.no Sat Nov 19 18:24:30 2011 From: jan.berger at video24.no (Jan Berger) Date: Sat, 19 Nov 2011 16:24:30 +0100 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: <4EC5B5F6.80600@puzzled.xs4all.nl> References: <4EC5B5F6.80600@puzzled.xs4all.nl> Message-ID: <5E6189A05FD641B9817B44A7CFD3D745@dell9400> Patrick, Those where 2 very good links - you don't accidentally know about a MAP, CAMEL or INAP protocol in open source as well? I notice that these SIGTRAN/SS7 protocols are LGPL so we could actually use them in FreeSWITCH. Is anyone doing work in this area? Jan --- I am not aware of one Open Source project that does all you are looking for. OpenBSC (http://openbsc.osmocom.org/trac/) seems to come close (MSC, BSC, AUC, HLR, VLR, EIR). Kannel (www.kannel.org) might be some piece of the SMSC puzzle and for SIGTRAN Mobicents-SS7 seems to fit the bill: http://www.mobicents.org/ss7/intro.html If you are going the TDM route too then Sangoma has excellent TDM cards (and hardware EC cards) and they are supported by Mobicents Have fun integrating all the parts. Seems like a fun job :) Regards, Patrick From paul at cupis.co.uk Sat Nov 19 19:07:46 2011 From: paul at cupis.co.uk (Paul Cupis) Date: Sat, 19 Nov 2011 16:07:46 +0000 Subject: [Freeswitch-users] Authorization User In-Reply-To: References: <4EC77F35.3060208@cupis.co.uk> Message-ID: <4EC7D452.1010205@cupis.co.uk> On 19/11/11 14:26, David Villasmil wrote: > Thanks for your reply Paul, > > That didn't do it though... Do you have caller-id-in-from set in the gateway? Can you provide the gateway configuration and the relevant dialplan XML? Regards, From jan.berger at video24.no Sat Nov 19 19:42:49 2011 From: jan.berger at video24.no (Jan Berger) Date: Sat, 19 Nov 2011 17:42:49 +0100 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: <5E6189A05FD641B9817B44A7CFD3D745@dell9400> References: <4EC5B5F6.80600@puzzled.xs4all.nl> <5E6189A05FD641B9817B44A7CFD3D745@dell9400> Message-ID: <80EC351A8FED43C189AE9BE232D0D1A9@dell9400> Answering myself - I found MAP/CAMEL/INAP on mibiscents site. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger Sent: 19. november 2011 16:25 To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Open Source MSC. Patrick, Those where 2 very good links - you don't accidentally know about a MAP, CAMEL or INAP protocol in open source as well? I notice that these SIGTRAN/SS7 protocols are LGPL so we could actually use them in FreeSWITCH. Is anyone doing work in this area? Jan --- I am not aware of one Open Source project that does all you are looking for. OpenBSC (http://openbsc.osmocom.org/trac/) seems to come close (MSC, BSC, AUC, HLR, VLR, EIR). Kannel (www.kannel.org) might be some piece of the SMSC puzzle and for SIGTRAN Mobicents-SS7 seems to fit the bill: http://www.mobicents.org/ss7/intro.html If you are going the TDM route too then Sangoma has excellent TDM cards (and hardware EC cards) and they are supported by Mobicents Have fun integrating all the parts. Seems like a fun job :) Regards, Patrick _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 engineerzuhairraza at gmail.com Sat Nov 19 20:11:25 2011 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Sat, 19 Nov 2011 21:11:25 +0400 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: <80EC351A8FED43C189AE9BE232D0D1A9@dell9400> References: <4EC5B5F6.80600@puzzled.xs4all.nl> <5E6189A05FD641B9817B44A7CFD3D745@dell9400> <80EC351A8FED43C189AE9BE232D0D1A9@dell9400> Message-ID: Hi, I have checked open BTS few days ago but couldnt find any sort of BSC and MSC, will check your links http://gnuradio.org/redmine/wiki/gnuradio/OpenBTS http://sourceforge.net/apps/trac/openbts/wiki/OpenBTS/BM2009FAQ http://www.youtube.com/watch?v=0vSG7H38J4g http://www.youtube.com/watch?v=kd9ceJbxfls&feature=related On Sat, Nov 19, 2011 at 8:42 PM, Jan Berger wrote: > Answering myself - I found MAP/CAMEL/INAP on mibiscents site. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan > Berger > Sent: 19. november 2011 16:25 > To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] Open Source MSC. > > Patrick, > > Those where 2 very good links - you don't accidentally know about a MAP, > CAMEL or INAP protocol in open source as well? > > I notice that these SIGTRAN/SS7 protocols are LGPL so we could actually use > them in FreeSWITCH. Is anyone doing work in this area? > > Jan > --- > I am not aware of one Open Source project that does all you are looking > for. OpenBSC (http://openbsc.osmocom.org/trac/) seems to come close > (MSC, BSC, AUC, HLR, VLR, EIR). Kannel (www.kannel.org) might be some > piece of the SMSC puzzle and for SIGTRAN Mobicents-SS7 seems to fit the > bill: http://www.mobicents.org/ss7/intro.html > > If you are going the TDM route too then Sangoma has excellent TDM cards > (and hardware EC cards) and they are supported by Mobicents > > Have fun integrating all the parts. Seems like a fun job :) > > Regards, > Patrick > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111119/a210bcc1/attachment-0001.html From david.villasmil.work at gmail.com Sat Nov 19 20:48:22 2011 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 19 Nov 2011 18:48:22 +0100 Subject: [Freeswitch-users] Authorization User In-Reply-To: <4EC7D452.1010205@cupis.co.uk> References: <4EC77F35.3060208@cupis.co.uk> <4EC7D452.1010205@cupis.co.uk> Message-ID: Hello Paul, The gw is the following: Thanks again for your help David On Sat, Nov 19, 2011 at 5:07 PM, Paul Cupis wrote: > On 19/11/11 14:26, David Villasmil wrote: > > Thanks for your reply Paul, > > > > That didn't do it though... > > Do you have caller-id-in-from set in the gateway? > > Can you provide the gateway configuration and the relevant dialplan XML? > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111119/ca1e8b65/attachment.html From david.villasmil.work at gmail.com Sat Nov 19 20:49:42 2011 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 19 Nov 2011 18:49:42 +0100 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: References: Message-ID: Hello, have a look here: http://www.openss7.org/sigtran.html Cheers David 2011/11/18 Henrik Aagaard S?rensen > Hi everyone. > > I do know that this is the mailing list for FreeSwitch. But as I'm > struggling finding anything (on Google etc.) about any open source MSC > (HLR/AUC/SMSC/MMSC etc.) I'm trying here. > > I want to use FreeSwitch in a MVNO setup with a operator to make a > proof-of-concept idea. > > But I need the MSC handling SIGTRAN and acts as HLR/AUC etc. > > I've looked at http://www.openimscore.org/ and http://www.mobicents.org, > but I'm struggling figuring out there place in such setup. > > This is just for the proof-of-concept, so scale-ability, stability etc. is > not as big a priority. > > Can anyone help me in the right direction? All help is appreciated! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111119/786856f9/attachment.html From freeswitch-list at puzzled.xs4all.nl Sat Nov 19 20:59:25 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Sat, 19 Nov 2011 18:59:25 +0100 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: References: <4EC5B5F6.80600@puzzled.xs4all.nl> <5E6189A05FD641B9817B44A7CFD3D745@dell9400> <80EC351A8FED43C189AE9BE232D0D1A9@dell9400> Message-ID: <4EC7EE7D.9080906@puzzled.xs4all.nl> On 11/19/2011 06:11 PM, Zohair Raza wrote: > Hi, > > I have checked open BTS few days ago but couldnt find any sort of BSC > and MSC, will check your links The project name is OpenBSC. Not OpenBTS :) Regards, Patrick From govoiper at gmail.com Sat Nov 19 21:52:59 2011 From: govoiper at gmail.com (Sammy Govind) Date: Sat, 19 Nov 2011 23:52:59 +0500 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: <4EC7EE7D.9080906@puzzled.xs4all.nl> References: <4EC5B5F6.80600@puzzled.xs4all.nl> <5E6189A05FD641B9817B44A7CFD3D745@dell9400> <80EC351A8FED43C189AE9BE232D0D1A9@dell9400> <4EC7EE7D.9080906@puzzled.xs4all.nl> Message-ID: The project name is OpenBSC. AND the project name is OpenBTS...So, so many open things now, all one need to do is combine them and make a Telecom infrastructure. !! simple !! On Sat, Nov 19, 2011 at 10:59 PM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 11/19/2011 06:11 PM, Zohair Raza wrote: > > Hi, > > > > I have checked open BTS few days ago but couldnt find any sort of BSC > > and MSC, will check your links > > The project name is OpenBSC. Not OpenBTS :) > > Regards, > Patrick > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111119/c234fc44/attachment.html From paul at cupis.co.uk Sat Nov 19 23:50:14 2011 From: paul at cupis.co.uk (Paul Cupis) Date: Sat, 19 Nov 2011 20:50:14 +0000 Subject: [Freeswitch-users] Authorization User In-Reply-To: References: <4EC77F35.3060208@cupis.co.uk> <4EC7D452.1010205@cupis.co.uk> Message-ID: <4EC81686.3000905@cupis.co.uk> On 19/11/11 17:48, David Villasmil wrote: > The gw is the following: > > > > > > > > Thanks again for your help You are missing: Try adding that, restarting the sofia profile (or freeswitch) and re-testing. Regards, From kheimerl at cs.berkeley.edu Sun Nov 20 01:11:08 2011 From: kheimerl at cs.berkeley.edu (Kurtis Heimerl) Date: Sat, 19 Nov 2011 14:11:08 -0800 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: References: <4EC5B5F6.80600@puzzled.xs4all.nl> <5E6189A05FD641B9817B44A7CFD3D745@dell9400> <80EC351A8FED43C189AE9BE232D0D1A9@dell9400> <4EC7EE7D.9080906@puzzled.xs4all.nl> Message-ID: OpenBTS bridges GSM directly to VoIP, removing the need for an HLR/SMSC/BSC/etc. It's not designed to interoperate with OpenBSC, it's a fundamentally different architecture. On Sat, Nov 19, 2011 at 10:52 AM, Sammy Govind wrote: > The project name is OpenBSC. AND the project name is OpenBTS...So, so many > open things now, all one need to do is combine them and make a Telecom > infrastructure. !! simple !! > > On Sat, Nov 19, 2011 at 10:59 PM, Patrick Lists > wrote: >> >> On 11/19/2011 06:11 PM, Zohair Raza wrote: >> > Hi, >> > >> > I have checked open BTS few days ago but couldnt find any sort of BSC >> > and MSC, will check your links >> >> The project name is OpenBSC. Not OpenBTS :) >> >> Regards, >> Patrick >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 jan.berger at video24.no Sun Nov 20 02:23:23 2011 From: jan.berger at video24.no (Jan Berger) Date: Sun, 20 Nov 2011 00:23:23 +0100 Subject: [Freeswitch-users] Open Source MSC. In-Reply-To: References: Message-ID: No thanks ? that site is a waste of time. Firstly it is GPL, secondly the stacks are not available to the public ? we used to call that a ?closed source project? :-) Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Sent: 19. november 2011 18:50 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Open Source MSC. Hello, have a look here: http://www.openss7.org/sigtran.html Cheers David 2011/11/18 Henrik Aagaard S?rensen Hi everyone. I do know that this is the mailing list for FreeSwitch. But as I'm struggling finding anything (on Google etc.) about any open source MSC (HLR/AUC/SMSC/MMSC etc.) I'm trying here. I want to use FreeSwitch in a MVNO setup with a operator to make a proof-of-concept idea. But I need the MSC handling SIGTRAN and acts as HLR/AUC etc. I've looked at http://www.openimscore.org/ and http://www.mobicents.org , but I'm struggling figuring out there place in such setup. This is just for the proof-of-concept, so scale-ability, stability etc. is not as big a priority. Can anyone help me in the right direction? All help is appreciated! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111120/d7c6f82f/attachment-0001.html From fraserredmond at gmail.com Sun Nov 20 03:42:52 2011 From: fraserredmond at gmail.com (Fraser Redmond) Date: Sat, 19 Nov 2011 19:42:52 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: <005501cca52f$aa0286c0$fe079440$@com> References: <005501cca52f$aa0286c0$fe079440$@com> Message-ID: Hi Stephen, I spent most of last week working out how to upgrade the kernel timer from 100HZ to 1000HZ on ubuntu on AWS. I documented the steps I took here: http://wiki.freeswitch.org/wiki/Amazon_EC2#Updating_Kernel_Timer_to_1000HZ Making that change gave us a noticeable jump in call quality. Cheers, Fraser On 17 November 2011 08:49, Stephen Dame wrote: > I?m running freeswitch on about 40 different m1.small c1.medium?s in AWS > regions, us-east, us-west, eu-west, and asia?. They are used in videoconf > component of BigBlueButton.org . For most part they work great? There are > occasional issues with voip quality but the app is 100% voip, with the BBB > client all browser based. So the conference is subject to every ones local > network connections and most issues ?blamed? on the internet instead of > freeswitch J We also tie in skype and DID direct to improve latency for > some clients. But expectations are set so we meet them.**** > > ** ** > > You probably have tougher business conferencing clients that want perfect > audio. But these are production deployed and generate revenue. All > these are on Ubuntu 10.04 official amis with no mods to kernels. **** > > ** ** > > I?m sure there are tweaks that can be made, and bare metal solutions that > would work a little better? Does anyone have any ideas how to optimize a > Ubuntu instance. I would engage in a few hours of consulting is so.**** > > ** ** > > The tests below are interesting but above my paygrade to understand what > they mean? **** > > ** ** > > I?m running them but don?t have a clue how to interpret. If you want a > simple of real running data, would be glad to run sample tests on these > distributed servers and provide back for analysis.**** > > ** ** > > Regards,**** > > Stephen**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Picher > *Sent:* Thursday, November 17, 2011 8:03 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Timing issues in AWS?**** > > ** ** > > Not having anywhere near the same trouble in a Xen Server in my lab... > > Although test_time is skewed 180 degrees from what it was in AWS.... > > freeswitch at internal> timer_test 120 10**** > > Avg: 119.945ms Total Time: 1199.727ms > > 2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: > samplecount after init: 1 > 2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: > samplecount after first step: 2 > 2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 > sleep 120 120993 > 2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 > sleep 120 118914 > 2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 > sleep 120 119919 > 2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 > sleep 120 120032 > 2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 > sleep 120 119802 > 2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 > sleep 120 119999 > 2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 > sleep 120 119975 > 2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 > sleep 120 120003 > 2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 > sleep 120 119849 > 2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 > sleep 120 119971**** > > > freeswitch at internal> time_test 600 10**** > > test 1 sleep 600 1 > test 2 sleep 600 1 > test 3 sleep 600 0 > test 4 sleep 600 1 > test 5 sleep 600 0 > test 6 sleep 600 1 > test 7 sleep 600 1 > test 8 sleep 600 0 > test 9 sleep 600 0 > test 10 sleep 600 1 > avg 0**** > > > [root at openuc bin]# uname -r**** > > 2.6.18-274.7.1.el5**** > > > [root at openuc bin]# grep CONFIG_HZ /boot/config-***** > > /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000**** > > > Thoughts as to why AWS results are so different from XenServer? Other > than not knowing who else is on the AWS box? > > Thanks, > Mike > > > **** > > On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher > wrote:**** > > m1.large > > I have a c1.xlarge queued up and ready to test...**** > > ** ** > > On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen > wrote:**** > > Just a simple question, what kind of AWS instance are you running your > FreeSWITCH?**** > > It makes huge difference.**** > > Thanks,**** > > Chris**** > > On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher > wrote:**** > > Hi guys, > > Trying to get to the bottom of some conference bridge issues I'm having > with running the system in AWS. > > We're hearing a bunch of snap-crackle-pops in conference bridges and when > I tcpdum on the server itself I see them in the RTP and see RTP timestamp > problems. > > I've run the following: > > > timer_test**** > > freeswitch at 127.0.0.1@internal> timer_test 120 10**** > > Avg: 120.004ms Total Time: 1200.315ms**** > > ** ** > > 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: > samplecount after init: 1 > freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] > mod_commands.c:466 Timer Test: samplecount after first step: 2 > 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 > sleep 120 120006 > 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 > sleep 120 120040 > 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 > sleep 120 119976 > 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 > sleep 120 120007 > 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 > sleep 120 120001 > 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 > sleep 120 120031 > 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 > sleep 120 119996 > 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 > sleep 120 119994 > 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 > sleep 120 120005 > 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 > sleep 120 119991**** > > > test_time**** > > freeswitch at 127.0.0.1@internal> time_test 600 10**** > > > test 1 sleep 600 1592 > test 2 sleep 600 986 > test 3 sleep 600 1018 > test 4 sleep 600 980 > test 5 sleep 600 1005 > test 6 sleep 600 1000 > test 7 sleep 600 972 > test 8 sleep 600 990 > test 9 sleep 600 1006 > test 10 sleep 600 994 > avg 1054**** > > > For kernel:**** > > [root at openuc bin]# uname -r > > 2.6.21.7-2.fc8xen**** > > > CONFIG_HZ:**** > > [root at openuc bin]# grep CONFIG_HZ /boot/config-* > > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000**** > > are the xenU kernel settings screwing me here? > > Thanks, > Mike > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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**** > > > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info**** > > > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111119/a02b6383/attachment-0001.html From anthony.minessale at gmail.com Sun Nov 20 06:27:32 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Nov 2011 21:27:32 -0600 Subject: [Freeswitch-users] Playback race condition In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> Message-ID: Every way you are discussing does it the same way, doing it over bgapi just wastes an extra thread for no reason. All of the above queue the command to the session and it will not execute it until the next time its convenient for the session to execute queued instructions. So any way you slice it, unless you wait for the execute event telling you that the playback you tried to start, has actually started then its not going to help. You are most likely trying to had implement some convention that is dumbed down to one command already in FS so I suggest you re-evaluate what you are even trying to do. On Sat, Nov 19, 2011 at 3:37 AM, Hynek Cihlar wrote: > So I changed the uuid_X-commands to execute and the use case with stopping > tone playback works like a charm now! > > Regarding the blocking, you are right. I'm limiting blocking (and context > switching) by handling all events and issuing commands always on one (and > the same) thread per session. This design also allows me to simplify the > programming model, decrease concurrency bugs, simplify debugging, etc. > That's why the synthetic delay was giving me troubles since the events and > commands must fly through as fast as possible. > > Thanks again! > > Hynek > > > > > On Sat, Nov 19, 2011 at 10:28 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > >> bgapi can of course be used also, and in some cases you might still need >> it. But the important thing to keep in mind when doing bgapi is to handle >> the events correctly. When executing bgapi you will get a job-id back, this >> job id must then be stored internally, so you can link this to the >> background job event. >> >> I try to use the sendMsg/execute as much as possible, you will get >> CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE events, so you will always know >> what's going on. >> >> I've only tested my system for about 100 concurrent calls, but that was >> handled without any problems at all. What's really important is to make >> really sure you never execute something that will block - since that will >> cause everything to "hang". >> >> /Peter >> ________________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >> freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [ >> hynek.cihlar at gmail.com] >> Skickat: den 19 november 2011 10:04 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Playback race condition >> >> Peter, thanks for pointing me to the right direction. I will try the >> execute command, since queueing is exactly what I would need to preserve >> consistency across commands (I had an impression that bgapi will give me >> this consistency). >> >> Also I like to hear that it is possible to use a single connection since >> I have designed the communication with one connection also. May I ask how >> many concurrent sessions can your system handle? >> >> Hynek >> >> >> >> On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson < >> peter.olsson at visionutveckling.se> >> wrote: >> If you think about performance I would say it's better to handle this >> without spawning lots of extra threads. >> >> First of all, most common commands can be queued into the the channel >> thread using the execute method ( >> http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes it >> possible to queue commands directly to the channel's thread, and the ESL >> session will never wait for the result, it returns immediately. >> >> I've never used uuid_broadcast myself, so I'm not really sure how it's >> handled, but by looking quickly in the code it looks like it shouldn't >> block either, since it's actually just queuing a playback command to the >> channel's thread. >> >> So basically, keep away from uuid_X-commands, and use execute method as >> much as possible, and when using uuid_x-methods, make sure that won't block >> - but if they do, make sure to handle handle the events for the bgapi >> correctly. >> >> I've built a complete IVR system using these methods, with a single ESL >> connection, and I've never had any performace issues. >> >> /Peter >> >> ________________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org> [ >> freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org>] för Hynek Cihlar [ >> hynek.cihlar at gmail.com] >> Skickat: den 19 november 2011 09:12 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Playback race condition >> >> Two reasons I am not using api in this case. >> >> 1. Performance >> 2. Doesn't seem to work >> >> 1. Blocking issuing the command will eventually reduce the system >> throughput. Agree this is only a potential theoretical problem and I >> haven't gotten into production where I could get the actual results. >> >> 2. When issuing api uuid_playback and api uuid_break right after, I'm >> always getting an error response from the uuid_break. The ESL message body >> is "-ERR no reply". Nothing in the logs (no idea which log type/level will >> could generate other useful info). >> >> Hynek >> >> >> >> On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < >> peter.olsson at visionutveckling.se> >> peter.olsson at visionutveckling.se>>> wrote: >> Since you're using bgapi, it's impossible to guarantee that >> uuid_broadcast has actually executed before you exeute uuid_break (they >> will be executed in two different threads). You will have to do this in a >> more controlled way (wait for events to show up etc.). And also, do you >> really need to use bgapi all the time? I'm not sure how uuid_broadcast is >> handled, but uuid_break is totally safe to use without bgapi, since it will >> just update a flag and then return immediately. I think uuid_broadcast will >> do the same thing - so getting rid of bgapi should be a good ebnough >> solution. >> >> /Peter >> ________________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org>> [ >> freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org>>] för Hynek Cihlar [ >> hynek.cihlar at gmail.com> hynek.cihlar at gmail.com>] >> Skickat: den 19 november 2011 08:31 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Playback race condition >> >> As a workaround I put a synthetic a delay between the start and stop of >> the playback. I wouldn't want to keep this solution however since it blocks >> one of the worker threads (decreasing overall system throughput) and >> second, I really don't know what the synthetic delay should be or whether >> it could differ during different conditions like during a high load. >> >> Was anybody facing similar problem? How did you solve it? Am I using the >> API wrong and is there another way to invoke playback/stop? >> >> Thanks! >> Hynek >> >> >> >> On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar > > hynek.cihlar at gmail.com>>> hynek.cihlar at gmail.com>> hynek.cihlar at gmail.com>>>> wrote: >> Hello all, >> >> I got on shaky grounds issuing commands to Freeswitch over ESL. >> >> For example, issuing the following commands close enough (on my system at >> around 100 ms and less) causes troubles. >> >> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >> playback::tone_stream://%(150,4000,425);loops=-1 >> <100ms appart >> bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all >> previous playback is not stopped, the following is queued (not played) >> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >> playback::tone_stream://%(1000,4000,425);loops=-1 >> >> It looks like the first tone playback is not properly initialized when >> the break arrives. Waiting for the start of the first tone playback >> (waiting for the right event) is not a solution, I don't want the tone to >> be played at all if the events come this close. >> >> Any ideas? >> >> >> Hynek >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org> consulting at freeswitch.org> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org> FreeSWITCH-users at lists.freeswitch.org>> FreeSWITCH-users at lists.freeswitch.org> FreeSWITCH-users at lists.freeswitch.org>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org> FreeSWITCH-users at lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:4ec770ca32761488716642! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111119/75dd045e/attachment-0001.html From anthony.minessale at gmail.com Sun Nov 20 06:28:10 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Nov 2011 21:28:10 -0600 Subject: [Freeswitch-users] Playback race condition In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> Message-ID: s/had/hand/ On Sat, Nov 19, 2011 at 9:27 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Every way you are discussing does it the same way, doing it over bgapi > just wastes an extra thread for no reason. > > All of the above queue the command to the session and it will not execute > it until the next time its convenient for the session to execute > queued instructions. > > So any way you slice it, unless you wait for the execute event telling you > that the playback you tried to start, has actually started then its not > going to help. > > You are most likely trying to had implement some convention that is dumbed > down to one command already in FS so I suggest you re-evaluate what you are > even trying to do. > > > > On Sat, Nov 19, 2011 at 3:37 AM, Hynek Cihlar wrote: > >> So I changed the uuid_X-commands to execute and the use case with >> stopping tone playback works like a charm now! >> >> Regarding the blocking, you are right. I'm limiting blocking (and context >> switching) by handling all events and issuing commands always on one (and >> the same) thread per session. This design also allows me to simplify the >> programming model, decrease concurrency bugs, simplify debugging, etc. >> That's why the synthetic delay was giving me troubles since the events and >> commands must fly through as fast as possible. >> >> Thanks again! >> >> Hynek >> >> >> >> >> On Sat, Nov 19, 2011 at 10:28 AM, Peter Olsson < >> peter.olsson at visionutveckling.se> wrote: >> >>> bgapi can of course be used also, and in some cases you might still need >>> it. But the important thing to keep in mind when doing bgapi is to handle >>> the events correctly. When executing bgapi you will get a job-id back, this >>> job id must then be stored internally, so you can link this to the >>> background job event. >>> >>> I try to use the sendMsg/execute as much as possible, you will get >>> CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE events, so you will always know >>> what's going on. >>> >>> I've only tested my system for about 100 concurrent calls, but that was >>> handled without any problems at all. What's really important is to make >>> really sure you never execute something that will block - since that will >>> cause everything to "hang". >>> >>> /Peter >>> ________________________________________ >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >>> freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [ >>> hynek.cihlar at gmail.com] >>> Skickat: den 19 november 2011 10:04 >>> Till: FreeSWITCH Users Help >>> ?mne: Re: [Freeswitch-users] Playback race condition >>> >>> Peter, thanks for pointing me to the right direction. I will try the >>> execute command, since queueing is exactly what I would need to preserve >>> consistency across commands (I had an impression that bgapi will give me >>> this consistency). >>> >>> Also I like to hear that it is possible to use a single connection since >>> I have designed the communication with one connection also. May I ask how >>> many concurrent sessions can your system handle? >>> >>> Hynek >>> >>> >>> >>> On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson < >>> peter.olsson at visionutveckling.se> >>> wrote: >>> If you think about performance I would say it's better to handle this >>> without spawning lots of extra threads. >>> >>> First of all, most common commands can be queued into the the channel >>> thread using the execute method ( >>> http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes >>> it possible to queue commands directly to the channel's thread, and the ESL >>> session will never wait for the result, it returns immediately. >>> >>> I've never used uuid_broadcast myself, so I'm not really sure how it's >>> handled, but by looking quickly in the code it looks like it shouldn't >>> block either, since it's actually just queuing a playback command to the >>> channel's thread. >>> >>> So basically, keep away from uuid_X-commands, and use execute method as >>> much as possible, and when using uuid_x-methods, make sure that won't block >>> - but if they do, make sure to handle handle the events for the bgapi >>> correctly. >>> >>> I've built a complete IVR system using these methods, with a single ESL >>> connection, and I've never had any performace issues. >>> >>> /Peter >>> >>> ________________________________________ >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org> [ >>> freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org>] för Hynek Cihlar [ >>> hynek.cihlar at gmail.com] >>> Skickat: den 19 november 2011 09:12 >>> Till: FreeSWITCH Users Help >>> ?mne: Re: [Freeswitch-users] Playback race condition >>> >>> Two reasons I am not using api in this case. >>> >>> 1. Performance >>> 2. Doesn't seem to work >>> >>> 1. Blocking issuing the command will eventually reduce the system >>> throughput. Agree this is only a potential theoretical problem and I >>> haven't gotten into production where I could get the actual results. >>> >>> 2. When issuing api uuid_playback and api uuid_break right after, I'm >>> always getting an error response from the uuid_break. The ESL message body >>> is "-ERR no reply". Nothing in the logs (no idea which log type/level will >>> could generate other useful info). >>> >>> Hynek >>> >>> >>> >>> On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < >>> peter.olsson at visionutveckling.se>> >>> peter.olsson at visionutveckling.se>>> wrote: >>> Since you're using bgapi, it's impossible to guarantee that >>> uuid_broadcast has actually executed before you exeute uuid_break (they >>> will be executed in two different threads). You will have to do this in a >>> more controlled way (wait for events to show up etc.). And also, do you >>> really need to use bgapi all the time? I'm not sure how uuid_broadcast is >>> handled, but uuid_break is totally safe to use without bgapi, since it will >>> just update a flag and then return immediately. I think uuid_broadcast will >>> do the same thing - so getting rid of bgapi should be a good ebnough >>> solution. >>> >>> /Peter >>> ________________________________________ >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org>> [ >>> freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org>>] för Hynek Cihlar [ >>> hynek.cihlar at gmail.com>> hynek.cihlar at gmail.com>] >>> Skickat: den 19 november 2011 08:31 >>> Till: FreeSWITCH Users Help >>> ?mne: Re: [Freeswitch-users] Playback race condition >>> >>> As a workaround I put a synthetic a delay between the start and stop of >>> the playback. I wouldn't want to keep this solution however since it blocks >>> one of the worker threads (decreasing overall system throughput) and >>> second, I really don't know what the synthetic delay should be or whether >>> it could differ during different conditions like during a high load. >>> >>> Was anybody facing similar problem? How did you solve it? Am I using the >>> API wrong and is there another way to invoke playback/stop? >>> >>> Thanks! >>> Hynek >>> >>> >>> >>> On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar >> >> hynek.cihlar at gmail.com>>>> hynek.cihlar at gmail.com>>> hynek.cihlar at gmail.com>>>> wrote: >>> Hello all, >>> >>> I got on shaky grounds issuing commands to Freeswitch over ESL. >>> >>> For example, issuing the following commands close enough (on my system >>> at around 100 ms and less) causes troubles. >>> >>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>> playback::tone_stream://%(150,4000,425);loops=-1 >>> <100ms appart >>> bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all >>> previous playback is not stopped, the following is queued (not played) >>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>> playback::tone_stream://%(1000,4000,425);loops=-1 >>> >>> It looks like the first tone playback is not properly initialized when >>> the break arrives. Waiting for the start of the first tone playback >>> (waiting for the right event) is not a solution, I don't want the tone to >>> be played at all if the events come this close. >>> >>> Any ideas? >>> >>> >>> Hynek >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org>> consulting at freeswitch.org> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org>> FreeSWITCH-users at lists.freeswitch.org>>> FreeSWITCH-users at lists.freeswitch.org>> FreeSWITCH-users at lists.freeswitch.org>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org>> FreeSWITCH-users at lists.freeswitch.org> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> !DSPAM:4ec770ca32761488716642! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111119/2749bc9a/attachment-0001.html From gabe at gundy.org Sun Nov 20 09:33:43 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 19 Nov 2011 23:33:43 -0700 Subject: [Freeswitch-users] Loading an IVR menu on the fly via URL In-Reply-To: <8B94625BC339264DBA61E314BE9EC2CF230EC8A6@EXCH125.IFN.com> References: <8B94625BC339264DBA61E314BE9EC2CF230EC8A6@EXCH125.IFN.com> Message-ID: On Fri, Nov 18, 2011 at 2:58 AM, RaviRaj Mulasa wrote: > We might be able to play IVR menu on the fly ?by combining the mod_xml_curl > and ?dialplan tool ?- ivr. Yes, this is exactly what you want to do. You don't have to configure the whole server via curl, but you can respond with the config for your IVR. Also, it's been a while since I've use the IVR module, but as I recall, it's been getting more support for channel variables that control its operation. You might also consider how those might help you control the IVR dynamically. Best, Gabe From gabe at gundy.org Sun Nov 20 09:45:13 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 19 Nov 2011 23:45:13 -0700 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: Message-ID: On Wed, Nov 16, 2011 at 9:56 AM, Michael Picher wrote: > That being said, it has been implied that FS should work on a large AWS > instance. I've also heard it implied that FS *might* run flawlessly on AWS and that most of the time you'll have no problems. However, one should also consider the recommendations of people with extensive experience (like Ken), who spend lots and lots of time helping others get to the root of their issues. Personally, I will not set a client's FS server up on virtual hosts for anything more than dev or test servers. If they want support, it has to be on bare-metal. Also, quality hardware is a must. Just my 2 cents :) Best, Gabe From gabe at gundy.org Sun Nov 20 09:49:24 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 19 Nov 2011 23:49:24 -0700 Subject: [Freeswitch-users] User Directory In-Reply-To: References: Message-ID: On Fri, Nov 18, 2011 at 8:33 AM, Elliott Vogel wrote: > I have been trying to put a user outside of a group like how it?s shown on > http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide but when I put a > user outside of a group the user cant authenticate. Can anyone see what I?m > doing wrong or is there error in the documentation? I would trim the example down to a minimally working snippet of XML (make it easy on us) and turn on SIP tracing for that profile. Selectively share the revenant portions of the SIP trace and then we'll be able to help you out. Good luck, Gabe From gabe at gundy.org Sun Nov 20 10:08:55 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Sun, 20 Nov 2011 00:08:55 -0700 Subject: [Freeswitch-users] Cannot receive CUSTOM event in Python In-Reply-To: References: Message-ID: On Sun, Nov 13, 2011 at 4:38 PM, Aasish Pappu wrote: > How do I catch this event from python? I am using a input_callback method to > handle events. The testevent never gets there, however DTMF events fired > using the flex keypad do reach. > What should I do? What about just using Event Socket? Gabe From hynek.cihlar at gmail.com Sun Nov 20 11:07:37 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sun, 20 Nov 2011 09:07:37 +0100 Subject: [Freeswitch-users] Playback race condition In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> Message-ID: So why would the bgapi uuid_break all fail if it queues the same way as execute does? Hynek On Sun, Nov 20, 2011 at 4:27 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Every way you are discussing does it the same way, doing it over bgapi > just wastes an extra thread for no reason. > > All of the above queue the command to the session and it will not execute > it until the next time its convenient for the session to execute > queued instructions. > > So any way you slice it, unless you wait for the execute event telling you > that the playback you tried to start, has actually started then its not > going to help. > > You are most likely trying to had implement some convention that is dumbed > down to one command already in FS so I suggest you re-evaluate what you are > even trying to do. > > > > On Sat, Nov 19, 2011 at 3:37 AM, Hynek Cihlar wrote: > >> So I changed the uuid_X-commands to execute and the use case with >> stopping tone playback works like a charm now! >> >> Regarding the blocking, you are right. I'm limiting blocking (and context >> switching) by handling all events and issuing commands always on one (and >> the same) thread per session. This design also allows me to simplify the >> programming model, decrease concurrency bugs, simplify debugging, etc. >> That's why the synthetic delay was giving me troubles since the events and >> commands must fly through as fast as possible. >> >> Thanks again! >> >> Hynek >> >> >> >> >> On Sat, Nov 19, 2011 at 10:28 AM, Peter Olsson < >> peter.olsson at visionutveckling.se> wrote: >> >>> bgapi can of course be used also, and in some cases you might still need >>> it. But the important thing to keep in mind when doing bgapi is to handle >>> the events correctly. When executing bgapi you will get a job-id back, this >>> job id must then be stored internally, so you can link this to the >>> background job event. >>> >>> I try to use the sendMsg/execute as much as possible, you will get >>> CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE events, so you will always know >>> what's going on. >>> >>> I've only tested my system for about 100 concurrent calls, but that was >>> handled without any problems at all. What's really important is to make >>> really sure you never execute something that will block - since that will >>> cause everything to "hang". >>> >>> /Peter >>> ________________________________________ >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >>> freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [ >>> hynek.cihlar at gmail.com] >>> Skickat: den 19 november 2011 10:04 >>> Till: FreeSWITCH Users Help >>> ?mne: Re: [Freeswitch-users] Playback race condition >>> >>> Peter, thanks for pointing me to the right direction. I will try the >>> execute command, since queueing is exactly what I would need to preserve >>> consistency across commands (I had an impression that bgapi will give me >>> this consistency). >>> >>> Also I like to hear that it is possible to use a single connection since >>> I have designed the communication with one connection also. May I ask how >>> many concurrent sessions can your system handle? >>> >>> Hynek >>> >>> >>> >>> On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson < >>> peter.olsson at visionutveckling.se> >>> wrote: >>> If you think about performance I would say it's better to handle this >>> without spawning lots of extra threads. >>> >>> First of all, most common commands can be queued into the the channel >>> thread using the execute method ( >>> http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes >>> it possible to queue commands directly to the channel's thread, and the ESL >>> session will never wait for the result, it returns immediately. >>> >>> I've never used uuid_broadcast myself, so I'm not really sure how it's >>> handled, but by looking quickly in the code it looks like it shouldn't >>> block either, since it's actually just queuing a playback command to the >>> channel's thread. >>> >>> So basically, keep away from uuid_X-commands, and use execute method as >>> much as possible, and when using uuid_x-methods, make sure that won't block >>> - but if they do, make sure to handle handle the events for the bgapi >>> correctly. >>> >>> I've built a complete IVR system using these methods, with a single ESL >>> connection, and I've never had any performace issues. >>> >>> /Peter >>> >>> ________________________________________ >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org> [ >>> freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org>] för Hynek Cihlar [ >>> hynek.cihlar at gmail.com] >>> Skickat: den 19 november 2011 09:12 >>> Till: FreeSWITCH Users Help >>> ?mne: Re: [Freeswitch-users] Playback race condition >>> >>> Two reasons I am not using api in this case. >>> >>> 1. Performance >>> 2. Doesn't seem to work >>> >>> 1. Blocking issuing the command will eventually reduce the system >>> throughput. Agree this is only a potential theoretical problem and I >>> haven't gotten into production where I could get the actual results. >>> >>> 2. When issuing api uuid_playback and api uuid_break right after, I'm >>> always getting an error response from the uuid_break. The ESL message body >>> is "-ERR no reply". Nothing in the logs (no idea which log type/level will >>> could generate other useful info). >>> >>> Hynek >>> >>> >>> >>> On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < >>> peter.olsson at visionutveckling.se>> >>> peter.olsson at visionutveckling.se>>> wrote: >>> Since you're using bgapi, it's impossible to guarantee that >>> uuid_broadcast has actually executed before you exeute uuid_break (they >>> will be executed in two different threads). You will have to do this in a >>> more controlled way (wait for events to show up etc.). And also, do you >>> really need to use bgapi all the time? I'm not sure how uuid_broadcast is >>> handled, but uuid_break is totally safe to use without bgapi, since it will >>> just update a flag and then return immediately. I think uuid_broadcast will >>> do the same thing - so getting rid of bgapi should be a good ebnough >>> solution. >>> >>> /Peter >>> ________________________________________ >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org>> [ >>> freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>> freeswitch-users-bounces at lists.freeswitch.org>>] för Hynek Cihlar [ >>> hynek.cihlar at gmail.com>> hynek.cihlar at gmail.com>] >>> Skickat: den 19 november 2011 08:31 >>> Till: FreeSWITCH Users Help >>> ?mne: Re: [Freeswitch-users] Playback race condition >>> >>> As a workaround I put a synthetic a delay between the start and stop of >>> the playback. I wouldn't want to keep this solution however since it blocks >>> one of the worker threads (decreasing overall system throughput) and >>> second, I really don't know what the synthetic delay should be or whether >>> it could differ during different conditions like during a high load. >>> >>> Was anybody facing similar problem? How did you solve it? Am I using the >>> API wrong and is there another way to invoke playback/stop? >>> >>> Thanks! >>> Hynek >>> >>> >>> >>> On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar >> >> hynek.cihlar at gmail.com>>>> hynek.cihlar at gmail.com>>> hynek.cihlar at gmail.com>>>> wrote: >>> Hello all, >>> >>> I got on shaky grounds issuing commands to Freeswitch over ESL. >>> >>> For example, issuing the following commands close enough (on my system >>> at around 100 ms and less) causes troubles. >>> >>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>> playback::tone_stream://%(150,4000,425);loops=-1 >>> <100ms appart >>> bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all >>> previous playback is not stopped, the following is queued (not played) >>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>> playback::tone_stream://%(1000,4000,425);loops=-1 >>> >>> It looks like the first tone playback is not properly initialized when >>> the break arrives. Waiting for the start of the first tone playback >>> (waiting for the right event) is not a solution, I don't want the tone to >>> be played at all if the events come this close. >>> >>> Any ideas? >>> >>> >>> Hynek >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org>> consulting at freeswitch.org> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org>> FreeSWITCH-users at lists.freeswitch.org>>> FreeSWITCH-users at lists.freeswitch.org>> FreeSWITCH-users at lists.freeswitch.org>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org>> FreeSWITCH-users at lists.freeswitch.org> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> !DSPAM:4ec770ca32761488716642! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111120/364dcc2c/attachment-0001.html From henrikaagaardsorensen at gmail.com Sun Nov 20 18:33:13 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Sun, 20 Nov 2011 16:33:13 +0100 Subject: [Freeswitch-users] Kernel timer, how does that affect FS? Message-ID: Hi all. I just saw a post (in this mailing list) recently regarding the kernel timer in Linux and how it affects sound quality in FS. Can anyone help me understand what the kernel timer does, how it affects FS and how to solve a possible problem on Linux? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111120/c15587ff/attachment.html From david.villasmil.work at gmail.com Sun Nov 20 22:44:16 2011 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 20 Nov 2011 20:44:16 +0100 Subject: [Freeswitch-users] Authorization User In-Reply-To: <4EC81686.3000905@cupis.co.uk> References: <4EC77F35.3060208@cupis.co.uk> <4EC7D452.1010205@cupis.co.uk> <4EC81686.3000905@cupis.co.uk> Message-ID: Perfect, thanks! On Sat, Nov 19, 2011 at 9:50 PM, Paul Cupis wrote: > On 19/11/11 17:48, David Villasmil wrote: > > The gw is the following: > > > > > > > > > > > > > > > > Thanks again for your help > > You are missing: > > > > Try adding that, restarting the sofia profile (or freeswitch) and > re-testing. > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111120/2238358d/attachment.html From elliott at zoogmedia.com Mon Nov 21 00:22:40 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Sun, 20 Nov 2011 21:22:40 +0000 Subject: [Freeswitch-users] Dialplan help Message-ID: Hello, How can I bridge and transfer a call at the same time. Basically I would like to ring an external number and a local extension at the same time, also if the external number doesn't pick up after 30 seconds stop I need to stop calling the external number and continue ringing local extension. How can I set this up in the dial plan? Elliott -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111120/7d6839b2/attachment.html From brian at freeswitch.org Mon Nov 21 00:27:18 2011 From: brian at freeswitch.org (Brian West) Date: Sun, 20 Nov 2011 15:27:18 -0600 Subject: [Freeswitch-users] Dialplan help In-Reply-To: References: Message-ID: <6C8BD0F8-1A6B-4940-AAF9-18871C2DD494@freeswitch.org> set leg_timeout in [] on the originate for the leg to the external number then that one will fall off and the rest will continue. Stop thinking of it as a bridge then transfer.. its just a bridge with various timeouts. for the target legs. /b On Nov 20, 2011, at 3:22 PM, Elliott Vogel wrote: > Hello, > > How can I bridge and transfer a call at the same time. Basically I would like to ring an external number and a local extension at the same time, also if the external number doesn't pick up after 30 seconds stop I need to stop calling the external number and continue ringing local extension. How can I set this up in the dial plan? > > Elliott -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111120/3d03de88/attachment.html From sdame at 207me.com Mon Nov 21 00:31:55 2011 From: sdame at 207me.com (Stephen Dame) Date: Sun, 20 Nov 2011 16:31:55 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: <005501cca52f$aa0286c0$fe079440$@com> Message-ID: <013301cca7cb$d3f525c0$7bdf7140$@com> Fraser, Thanks for sharing , i'll try to upgrade an instance and give it a try. Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fraser Redmond Sent: Saturday, November 19, 2011 7:43 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Timing issues in AWS? Hi Stephen, I spent most of last week working out how to upgrade the kernel timer from 100HZ to 1000HZ on ubuntu on AWS. I documented the steps I took here: http://wiki.freeswitch.org/wiki/Amazon_EC2#Updating_Kernel_Timer_to_1000HZ Making that change gave us a noticeable jump in call quality. Cheers, Fraser On 17 November 2011 08:49, Stephen Dame wrote: I'm running freeswitch on about 40 different m1.small c1.medium's in AWS regions, us-east, us-west, eu-west, and asia.. They are used in videoconf component of BigBlueButton.org . For most part they work great. There are occasional issues with voip quality but the app is 100% voip, with the BBB client all browser based. So the conference is subject to every ones local network connections and most issues "blamed" on the internet instead of freeswitch J We also tie in skype and DID direct to improve latency for some clients. But expectations are set so we meet them. You probably have tougher business conferencing clients that want perfect audio. But these are production deployed and generate revenue. All these are on Ubuntu 10.04 official amis with no mods to kernels. I'm sure there are tweaks that can be made, and bare metal solutions that would work a little better. Does anyone have any ideas how to optimize a Ubuntu instance. I would engage in a few hours of consulting is so. The tests below are interesting but above my paygrade to understand what they mean? I'm running them but don't have a clue how to interpret. If you want a simple of real running data, would be glad to run sample tests on these distributed servers and provide back for analysis. Regards, Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Picher Sent: Thursday, November 17, 2011 8:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Timing issues in AWS? Not having anywhere near the same trouble in a Xen Server in my lab... Although test_time is skewed 180 degrees from what it was in AWS.... freeswitch at internal> timer_test 120 10 Avg: 119.945ms Total Time: 1199.727ms 2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120993 2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 118914 2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119919 2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120032 2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 119802 2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 119999 2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119975 2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 120003 2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 119849 2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119971 freeswitch at internal> time_test 600 10 test 1 sleep 600 1 test 2 sleep 600 1 test 3 sleep 600 0 test 4 sleep 600 1 test 5 sleep 600 0 test 6 sleep 600 1 test 7 sleep 600 1 test 8 sleep 600 0 test 9 sleep 600 0 test 10 sleep 600 1 avg 0 [root at openuc bin]# uname -r 2.6.18-274.7.1.el5 [root at openuc bin]# grep CONFIG_HZ /boot/config-* /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000 Thoughts as to why AWS results are so different from XenServer? Other than not knowing who else is on the AWS box? Thanks, Mike On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher wrote: m1.large I have a c1.xlarge queued up and ready to test... On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen wrote: Just a simple question, what kind of AWS instance are you running your FreeSWITCH? It makes huge difference. Thanks, Chris On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher wrote: Hi guys, Trying to get to the bottom of some conference bridge issues I'm having with running the system in AWS. We're hearing a bunch of snap-crackle-pops in conference bridges and when I tcpdum on the server itself I see them in the RTP and see RTP timestamp problems. I've run the following: timer_test freeswitch at 127.0.0.1@internal> timer_test 120 10 Avg: 120.004ms Total Time: 1200.315ms 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120006 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 120040 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119976 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120007 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 120001 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 120031 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119996 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 119994 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 120005 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119991 test_time freeswitch at 127.0.0.1@internal> time_test 600 10 test 1 sleep 600 1592 test 2 sleep 600 986 test 3 sleep 600 1018 test 4 sleep 600 980 test 5 sleep 600 1005 test 6 sleep 600 1000 test 7 sleep 600 972 test 8 sleep 600 990 test 9 sleep 600 1006 test 10 sleep 600 994 avg 1054 For kernel: [root at openuc bin]# uname -r 2.6.21.7-2.fc8xen CONFIG_HZ: [root at openuc bin]# grep CONFIG_HZ /boot/config-* /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 are the xenU kernel settings screwing me here? Thanks, Mike -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111120/31c2398d/attachment-0001.html From lon at kickasspixels.com Mon Nov 21 00:58:52 2011 From: lon at kickasspixels.com (Lon Baker) Date: Sun, 20 Nov 2011 13:58:52 -0800 Subject: [Freeswitch-users] Timing issues in AWS? (Stephen Dame) Message-ID: It would be interesting to compare timing issues across major could services. Amazon, Rackspace, linode, etc. I've had poor experiences under ESX and avoid virtualization for anything handling media. -- Lon -- Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111120/29471d4b/attachment.html From avi at avimarcus.net Mon Nov 21 01:06:30 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 21 Nov 2011 00:06:30 +0200 Subject: [Freeswitch-users] Unified logging of incoming mobile calls - partial solution for android Message-ID: I know many people run/use a PBX of some sort, and I figured I'd share what I discovered: I have a mobile phone that gets a lot of calls directly, but I wanted to log those calls to a CRM. This is a partial solution.. Android Notifier ( http://code.google.com/p/android-notifier/) is an app that can be installed on the android and send a UDP message to a server (or also to a desktop client) when calls come in. The desktop client can launch a website to pull data for a screen pop of some sort.... It seemed to sometimes lose sync (when wifi went off and then came back..) but it's got the last 10 messages or so. I have it send to UDP, and this small node.js script saves the time-stamped messages to a file. (Note: timestamp is in milliseconds..) //http://code.google.com/p/android-notifier/wiki/NotificationProtocol > //DEVICE_ID/NOTIFICATION_ID/EVENT_TYPE/EVENT_CONTENTS > var fs = require('fs'); > var log = fs.createWriteStream("incoming_call_log.txt", {'flags': 'a'} ); > var dgram = require("dgram"); > var server = dgram.createSocket("udp4"); > server.on("message", function (msg, rinfo) { > // console.log("server got: " + msg + " from " + rinfo.address + ":" + > rinfo.port); > // console.log("Got: " + Date.now() + "/" + rinfo.address +"/" + > rinfo.port + "/" + msg); > log.write(Date.now() + "/" + rinfo.address +"/" + rinfo.port > + "/" + msg + "\n"); > }); > server.on("listening", function () { > var address = server.address(); > console.log("server listening " + address.address + ":" + > address.port); > }); > server.bind(10600, "YOUR IP"); And you can background and disown it, so it stays running.. after you launch node.js. Due to the abnormal method of submitting - not HTTP - I didn't know how to write anything in another language to do this. I suppose you could do it with python twisted. And for a screenpop.. set the desktop notifier to launch: /home/folder/web.sh "deviceId={deviceId}&id={id}&type={type}&data={data}" web.sh contains: > #!/bin/bash > echo "$*"; #>> /tmp/websh.log; #logging... > var=$* > /usr/bin/chromium-browser "http://website.com/script.php?$var" Anyway, this is just the basics. Feel free to share how you integrate it. I didn't see the specs on how to decrypt the signal algorithimically, but it's part of the desktop app. -Avi Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111121/2df6450d/attachment.html From boris at tagnet.ru Mon Nov 21 08:01:19 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Mon, 21 Nov 2011 11:01:19 +0600 Subject: [Freeswitch-users] Call to registered user Message-ID: <4EC9DB1F.1000001@tagnet.ru> Hello! My FS has different sip profiles (internet, local net, dmz etc). There is some type of mobile users that may register with any profile. So, is there a simple way to call registered user without known on which profile it has registered? AFAIK sofia_contact need profile and user/ is looking in profile of incoming call. -- Regards, Boris From vetali100 at gmail.com Mon Nov 21 09:10:20 2011 From: vetali100 at gmail.com (Vitali Colosov) Date: Sun, 20 Nov 2011 22:10:20 -0800 Subject: [Freeswitch-users] Call to registered user In-Reply-To: <4EC9DB1F.1000001@tagnet.ru> References: <4EC9DB1F.1000001@tagnet.ru> Message-ID: <47DB6473-FF92-4C17-AA40-29AAED903280@gmail.com> Ugly, but 100% working approach would be to dial all profiles, adding continue_on_fail=USER_NOT_REGISTERED in all your dialplans which are linked to all your profiles. The following example works fine: Regards, Vitalie On Nov 20, 2011, at 9:01 PM, Boris Kovalenko wrote: > Hello! > > My FS has different sip profiles (internet, local net, dmz etc). > There is some type of mobile users that may register with any profile. > So, is there a simple way to call registered user without known on which > profile it has registered? AFAIK sofia_contact need profile and user/ is > looking in profile of incoming call. > > -- > Regards, > Boris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 boris at tagnet.ru Mon Nov 21 09:21:20 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Mon, 21 Nov 2011 12:21:20 +0600 Subject: [Freeswitch-users] Call to registered user In-Reply-To: <47DB6473-FF92-4C17-AA40-29AAED903280@gmail.com> References: <4EC9DB1F.1000001@tagnet.ru> <47DB6473-FF92-4C17-AA40-29AAED903280@gmail.com> Message-ID: <4EC9EDE0.4050707@tagnet.ru> Hello! Yes, I know about this way :) I'm looking for a more simple solution. And I really can't understand why developers haven't implemented a function that can search in all profiles. > Ugly, but 100% working approach would be to dial all profiles, adding continue_on_fail=USER_NOT_REGISTERED in all your dialplans which are linked to all your profiles. > > The following example works fine: > > > > > > > > > Regards, > Vitalie > > > On Nov 20, 2011, at 9:01 PM, Boris Kovalenko wrote: > >> Hello! >> >> My FS has different sip profiles (internet, local net, dmz etc). >> There is some type of mobile users that may register with any profile. >> So, is there a simple way to call registered user without known on which >> profile it has registered? AFAIK sofia_contact need profile and user/ is >> looking in profile of incoming call. >> >> -- >> Regards, >> Boris >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Regards, Boris From vetali100 at gmail.com Mon Nov 21 09:23:44 2011 From: vetali100 at gmail.com (Vitali Colosov) Date: Sun, 20 Nov 2011 22:23:44 -0800 Subject: [Freeswitch-users] Call to registered user In-Reply-To: <4EC9EDE0.4050707@tagnet.ru> References: <4EC9DB1F.1000001@tagnet.ru> <47DB6473-FF92-4C17-AA40-29AAED903280@gmail.com> <4EC9EDE0.4050707@tagnet.ru> Message-ID: <0205F022-780D-425E-9633-5E3E19650656@gmail.com> Maybe the function exists, and we are just not aware :) Let's wait what the developers would say... On Nov 20, 2011, at 10:21 PM, Boris Kovalenko wrote: > Hello! > > Yes, I know about this way :) I'm looking for a more simple > solution. And I really can't understand why developers haven't > implemented a function that can search in all profiles. > >> Ugly, but 100% working approach would be to dial all profiles, adding continue_on_fail=USER_NOT_REGISTERED in all your dialplans which are linked to all your profiles. >> >> The following example works fine: >> >> >> >> >> >> >> >> >> Regards, >> Vitalie >> >> >> On Nov 20, 2011, at 9:01 PM, Boris Kovalenko wrote: >> >>> Hello! >>> >>> My FS has different sip profiles (internet, local net, dmz etc). >>> There is some type of mobile users that may register with any profile. >>> So, is there a simple way to call registered user without known on which >>> profile it has registered? AFAIK sofia_contact need profile and user/ is >>> looking in profile of incoming call. >>> >>> -- >>> Regards, >>> Boris >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > -- > Regards, > Boris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 sid at eltc.ru Mon Nov 21 09:38:53 2011 From: sid at eltc.ru (Sergey Scheglov) Date: Mon, 21 Nov 2011 13:38:53 +0700 Subject: [Freeswitch-users] Call to registered user In-Reply-To: <4EC9DB1F.1000001@tagnet.ru> References: <4EC9DB1F.1000001@tagnet.ru> Message-ID: <20111121133853.3b5e17e8@shadow> ? Mon, 21 Nov 2011 11:01:19 +0600 Boris Kovalenko wrote: > My FS has different sip profiles (internet, local net, dmz etc). > There is some type of mobile users that may register with any > profile. So, is there a simple way to call registered user without > known on which profile it has registered? AFAIK sofia_contact need > profile and user/ is looking in profile of incoming call. Try to use aliases for each profile. In this case, you don't need to know about profile name. -- Sergey Scheglov From boris at tagnet.ru Mon Nov 21 10:17:58 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Mon, 21 Nov 2011 13:17:58 +0600 Subject: [Freeswitch-users] Call to registered user In-Reply-To: <20111121133853.3b5e17e8@shadow> References: <4EC9DB1F.1000001@tagnet.ru> <20111121133853.3b5e17e8@shadow> Message-ID: <4EC9FB26.808@tagnet.ru> But AFAIK I can't use the same alias for two or more profiles? Am I wrong? > ? Mon, 21 Nov 2011 11:01:19 +0600 > Boris Kovalenko wrote: > >> My FS has different sip profiles (internet, local net, dmz etc). >> There is some type of mobile users that may register with any >> profile. So, is there a simple way to call registered user without >> known on which profile it has registered? AFAIK sofia_contact need >> profile and user/ is looking in profile of incoming call. > Try to use aliases for each profile. In this case, you don't need to > know about profile name. > -- Regards, Boris From SPapineni at enghouse.com Mon Nov 21 13:28:54 2011 From: SPapineni at enghouse.com (Papineni, Suneel) Date: Mon, 21 Nov 2011 10:28:54 +0000 Subject: [Freeswitch-users] Joining a call from Gtalk to a conference on FreeSwitch Message-ID: <9438D04074E0DE45A49CD7609982127225B662F2@CORP-MAIL-002.edge.local> Hi, I got the new GIT version and enabled mod_dingling and compiled. Everything went through and able to establish call to an extension if I configure that extension number in "client" profile. What I am trying to do is, I want to bridge or join a call coming from GTalk to an existing conference in FreeSwitch. For this purpose I configured a different number on "client profile" and created a dial-plan for this number to 'park' the call first before trying to join to the conference. Then using eventSockets I am trying to join this call to conference and issued following command. (tried with "uuid_bridge" command as well) "api uuid_transfer [Unique-ID] conference:xyz at default inline" Command is successful and also I can hear a sound that someone joined in the conference, but I didn't hear any voice at either side. I couldn't see any RTP flow as well (checked wireshark traces at FS). After sometime like 30 seconds call at GTalk is disconnected automatically. I am not sure why nothing is heard at both sides and why call got disconnected. Also tried answering the call first (after Park) and then bridging to conference, still got the same issue. Could someone please let me know if I am missing anything or need to configure in a different way for conferencing. Thanks & Regards Suneel Client.xml Dial-plan.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111121/faa10821/attachment.html From william at xofap.com Mon Nov 21 14:08:35 2011 From: william at xofap.com (William Alianto) Date: Mon, 21 Nov 2011 18:08:35 +0700 Subject: [Freeswitch-users] SIP INVITE manipulation Message-ID: <4ECA3133.80706@xofap.com> Hi, I'm looking for a way to manipulate the SIP INVITE message. Currently my SIP server sending 622X to call to provider. But the provider only expect 2X from the invite, which cause the call won't be recognized. Is there any string on diaplan I can add to adjust the invite message, so the provider recognize the call? -- Regards, William From jcgpoza at gmail.com Mon Nov 21 17:35:06 2011 From: jcgpoza at gmail.com (jcgpoza gonzalez) Date: Mon, 21 Nov 2011 15:35:06 +0100 Subject: [Freeswitch-users] SIP INVITE manipulation In-Reply-To: <4ECA3133.80706@xofap.com> References: <4ECA3133.80706@xofap.com> Message-ID: Hello William, I'm interested in this too. As far as I know the only way is to to patch the mod_sofia. Good luck! 2011/11/21 William Alianto > Hi, > > I'm looking for a way to manipulate the SIP INVITE message. Currently my > SIP server sending 622X to call to provider. But the provider only > expect 2X from the invite, which cause the call won't be recognized. Is > there any string on diaplan I can add to adjust the invite message, so > the provider recognize the call? > > -- > Regards, > > William > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111121/07d06414/attachment.html From brian at freeswitch.org Mon Nov 21 18:39:57 2011 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Nov 2011 09:39:57 -0600 Subject: [Freeswitch-users] SIP INVITE manipulation In-Reply-To: <4ECA3133.80706@xofap.com> References: <4ECA3133.80706@xofap.com> Message-ID: <3FAD407D-AC37-49B6-8245-9D3CD9E4822F@freeswitch.org> What do you mean? Show me how you bridge the call and what is in the packet. /b On Nov 21, 2011, at 5:08 AM, William Alianto wrote: > Hi, > > I'm looking for a way to manipulate the SIP INVITE message. Currently my > SIP server sending 622X to call to provider. But the provider only > expect 2X from the invite, which cause the call won't be recognized. Is > there any string on diaplan I can add to adjust the invite message, so > the provider recognize the call? > > -- > Regards, > > William -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111121/d1060ebf/attachment.html From debianmailz at gmail.com Mon Nov 21 14:00:30 2011 From: debianmailz at gmail.com (D M) Date: Mon, 21 Nov 2011 12:00:30 +0100 Subject: [Freeswitch-users] SIP invalid call attempts from unknown dialer Message-ID: <4ECA2F4E.3060909@gmail.com> Hello, I have noticed quite a few different attempts on accessing my freeswitch machine via SIP that does not come from the company. I have attached part of a log with such an attempt. My main concern is ensuring that similar attempts will not be able to make external calls. I realize that any number made available externally will also be accessable via this method so my secondary concern is throttling or preventing these type of attempts to avoid autodialer spam. This log repeats with around 12 call attempts per second for almost a minute times with different attempts on seemingly random numbers. There has been multiple different attempts to spam random numbers via SIP but so far none has been successful. This log is the most relevant since it made a single login attempt on an nonexistent user after which it has either successfully spoofed the ip of the freeswitch machine or used an vulnerability in either my config or freeswitch. My config is the default freeswitch+fusionpbx installation on Ubuntu 10.04.3 LTS with instructions from here (http://wiki.fusionpbx.com/index.php?title=Easy_Ubuntu_10.04&oldid=1574). With a few minor configuration changes: * Registering is done via external domain pointing to the freeswitch machine, NOT using the default port 5060 * Port 5060 is generally used for traffic with SIP provider that connects us to phone network but the port not firewalled/restricted in any other way This is an example log of a single login attempt and single call attempt, the following modifications have been made: * Freeswitch public ip has been changed to xx.xx.xx.xx * 2 regexps have been changed from public telephone number to /^publicnumber$/ and /^publicnumber2$/ * A large list of regexps have been replaced with Please let me know if you need any more details or longer logs Thanks, Daniel ##### LOG BEGIN ##### 2011-11-18 15:27:33.293146 [WARNING] sofia_reg.c:2283 Can't find user [1010 at xx.xx.xx.xx] You must define a domain called 'xx.xx.xx.xx' in your directory and add a user with the id="1010" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2011-11-18 15:27:36.633145 [NOTICE] switch_channel.c:897 New Channel sofia/external/1010 at xx.xx.xx.xx:5060 [75cf1808-11f1-11e1-9c95-494fea388543] 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5084 Channel sofia/external/1010 at xx.xx.xx.xx:5060 entering state [received][100] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_NEW 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5095 Remote SDP: v=0^M o=1010 13216264671138 13216264671138 IN IP4 192.168.1.3^M s=VaxSoft^M c=IN IP4 192.168.1.3^M t=0 0^M m=audio 7000 RTP/AVP 0 8 3 98 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:3 GSM/8000^M a=rtpmap:98 iLBC/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:343 (sofia/external/1010 at xx.xx.xx.xx:5060) State NEW 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4711 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000] 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4711 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/1010 at xx.xx.xx.xx:5060 PCMA/8000 20 ms 160 samples 64000 bits 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4825 Set 2833 dtmf send/recv payload to 101 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5284 (sofia/external/1010 at xx.xx.xx.xx:5060) State Change CS_NEW -> CS_INIT 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_INIT 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1010 at xx.xx.xx.xx:5060) State INIT 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:85 sofia/external/1010 at xx.xx.xx.xx:5060 SOFIA INIT 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:125 (sofia/external/1010 at xx.xx.xx.xx:5060) State Change CS_INIT -> CS_ROUTING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1010 at xx.xx.xx.xx:5060) State INIT going to sleep 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_ROUTING 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:1821 (sofia/external/1010 at xx.xx.xx.xx:5060) Callstate Change DOWN -> RINGING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:373 (sofia/external/1010 at xx.xx.xx.xx:5060) State ROUTING 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:148 sofia/external/1010 at xx.xx.xx.xx:5060 SOFIA ROUTING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:77 sofia/external/1010 at xx.xx.xx.xx:5060 Standard ROUTING 2011-11-18 15:27:36.633145 [INFO] mod_dialplan_xml.c:336 Processing MyName <1010>->972592182076 in context public Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing [public->unloop] continue=false Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing [public->outside_call] continue=true Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Absolute Condition [outside_call] Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Action set(outside_call=true) Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing [public->call_debug] continue=true Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing [public->public_extensions] continue=false Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [public_extensions] destination_number(972592182076) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing [public->TEMP] continue=false Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (PASS) [TEMP] context(public) =~ /public/ break=on-false Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [TEMP] destination_number(972592182076) =~ /^publicnumber$/ break=on-false Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing [public->Misc_Number] continue=false Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (PASS) [Misc_Number] context(public) =~ /public/ break=on-false Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [Misc_Number] destination_number(972592182076) =~ /^publicnumber2$/ break=on-false 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:119 (sofia/external/1010 at xx.xx.xx.xx:5060) State Change CS_ROUTING -> CS_EXECUTE 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:373 (sofia/external/1010 at xx.xx.xx.xx:5060) State ROUTING going to sleep 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_EXECUTE 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:380 (sofia/external/1010 at xx.xx.xx.xx:5060) State EXECUTE 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:241 sofia/external/1010 at xx.xx.xx.xx:5060 SOFIA EXECUTE 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:157 sofia/external/1010 at xx.xx.xx.xx:5060 Standard EXECUTE EXECUTE sofia/external/1010 at xx.xx.xx.xx:5060 set(outside_call=true) 2011-11-18 15:27:36.633145 [DEBUG] mod_dptools.c:1063 sofia/external/1010 at xx.xx.xx.xx:5060 SET [outside_call]=[true] EXECUTE sofia/external/1010 at xx.xx.xx.xx:5060 set(RFC2822_DATE=Fri, 18 Nov 2011 15:27:36 +0100) 2011-11-18 15:27:36.633145 [DEBUG] mod_dptools.c:1063 sofia/external/1010 at xx.xx.xx.xx:5060 SET [RFC2822_DATE]=[Fri, 18 Nov 2011 15:27:36 +0100] 2011-11-18 15:27:36.633145 [NOTICE] switch_core_state_machine.c:189 sofia/external/1010 at xx.xx.xx.xx:5060 has executed the last dialplan instruction, hanging up. 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:2739 (sofia/external/1010 at xx.xx.xx.xx:5060) Callstate Change RINGING -> HANGUP 2011-11-18 15:27:36.633145 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/external/1010 at xx.xx.xx.xx:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:2755 Send signal sofia/external/1010 at xx.xx.xx.xx:5060 [KILL] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:380 (sofia/external/1010 at xx.xx.xx.xx:5060) State EXECUTE going to sleep 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_HANGUP 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:575 (sofia/external/1010 at xx.xx.xx.xx:5060) State HANGUP 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:458 Channel sofia/external/1010 at xx.xx.xx.xx:5060 hanging up, cause: NORMAL_CLEARING 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:522 Responding to INVITE with: 480 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:46 sofia/external/1010 at xx.xx.xx.xx:5060 Standard HANGUP, cause: NORMAL_CLEARING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:575 (sofia/external/1010 at xx.xx.xx.xx:5060) State HANGUP going to sleep 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:356 (sofia/external/1010 at xx.xx.xx.xx:5060) State Change CS_HANGUP -> CS_REPORTING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_REPORTING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:635 (sofia/external/1010 at xx.xx.xx.xx:5060) State REPORTING From sos at sokhapkin.dyndns.org Mon Nov 21 19:11:53 2011 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 21 Nov 2011 11:11:53 -0500 Subject: [Freeswitch-users] SIP invalid call attempts from unknown dialer In-Reply-To: <4ECA2F4E.3060909@gmail.com> References: <4ECA2F4E.3060909@gmail.com> Message-ID: <201111211111.53970.sos@sokhapkin.dyndns.org> http://wiki.freeswitch.org/wiki/Fail2ban On Monday 21 November 2011, D M wrote: > Hello, > I have noticed quite a few different attempts on accessing my freeswitch > machine via SIP that does not come from the company. I have attached > part of a log with such an attempt. > > My main concern is ensuring that similar attempts will not be able to > make external calls. I realize that any number made available externally > will also be accessable via this method so my secondary concern is > throttling or preventing these type of attempts to avoid autodialer spam. > > This log repeats with around 12 call attempts per second for almost a > minute times with different attempts on seemingly random numbers. There > has been multiple different attempts to spam random numbers via SIP but > so far none has been successful. This log is the most relevant since it > made a single login attempt on an nonexistent user after which it has > either successfully spoofed the ip of the freeswitch machine or used an > vulnerability in either my config or freeswitch. > > My config is the default freeswitch+fusionpbx installation on Ubuntu > 10.04.3 LTS with instructions from here > (http://wiki.fusionpbx.com/index.php?title=Easy_Ubuntu_10.04&oldid=1574). > With a few minor configuration changes: > * Registering is done via external domain pointing to the freeswitch > machine, NOT using the default port 5060 > * Port 5060 is generally used for traffic with SIP provider that > connects us to phone network but the port not firewalled/restricted in > any other way > > This is an example log of a single login attempt and single call > attempt, the following modifications have been made: > * Freeswitch public ip has been changed to xx.xx.xx.xx > * 2 regexps have been changed from public telephone number to > /^publicnumber$/ and /^publicnumber2$/ > * A large list of regexps have been replaced with > > Please let me know if you need any more details or longer logs > > Thanks, > Daniel > > ##### LOG BEGIN ##### > > 2011-11-18 15:27:33.293146 [WARNING] sofia_reg.c:2283 Can't find user > [1010 at xx.xx.xx.xx] > You must define a domain called 'xx.xx.xx.xx' in your directory and add > a user with the id="1010" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > 2011-11-18 15:27:36.633145 [NOTICE] switch_channel.c:897 New Channel > sofia/external/1010 at xx.xx.xx.xx:5060 [75cf1808-11f1-11e1-9c95-494fea388543] > 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5084 Channel > sofia/external/1010 at xx.xx.xx.xx:5060 entering state [received][100] > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 > (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_NEW > 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5095 Remote SDP: > v=0^M > o=1010 13216264671138 13216264671138 IN IP4 192.168.1.3^M > s=VaxSoft^M > c=IN IP4 192.168.1.3^M > t=0 0^M > m=audio 7000 RTP/AVP 0 8 3 98 101^M > a=rtpmap:0 PCMU/8000^M > a=rtpmap:8 PCMA/8000^M > a=rtpmap:3 GSM/8000^M > a=rtpmap:98 iLBC/8000^M > a=rtpmap:101 telephone-event/8000^M > a=fmtp:101 0-16^M > > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:343 > (sofia/external/1010 at xx.xx.xx.xx:5060) State NEW > 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4711 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000] > 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4711 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] > 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:2819 Set Codec > sofia/external/1010 at xx.xx.xx.xx:5060 PCMA/8000 20 ms 160 samples 64000 bits > 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4825 Set 2833 dtmf > send/recv payload to 101 > 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5284 > (sofia/external/1010 at xx.xx.xx.xx:5060) State Change CS_NEW -> CS_INIT > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send > signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 > (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_INIT > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:364 > (sofia/external/1010 at xx.xx.xx.xx:5060) State INIT > 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:85 > sofia/external/1010 at xx.xx.xx.xx:5060 SOFIA INIT > 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:125 > (sofia/external/1010 at xx.xx.xx.xx:5060) State Change CS_INIT -> CS_ROUTING > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send > signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:364 > (sofia/external/1010 at xx.xx.xx.xx:5060) State INIT going to sleep > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 > (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_ROUTING > 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:1821 > (sofia/external/1010 at xx.xx.xx.xx:5060) Callstate Change DOWN -> RINGING > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:373 > (sofia/external/1010 at xx.xx.xx.xx:5060) State ROUTING > 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:148 > sofia/external/1010 at xx.xx.xx.xx:5060 SOFIA ROUTING > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:77 > sofia/external/1010 at xx.xx.xx.xx:5060 Standard ROUTING > 2011-11-18 15:27:36.633145 [INFO] mod_dialplan_xml.c:336 Processing > MyName <1010>->972592182076 in context public > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing [public->unloop] > continue=false > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing > [public->outside_call] continue=true > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Absolute Condition > [outside_call] > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Action > set(outside_call=true) Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 > Action > set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing > [public->call_debug] continue=true > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [call_debug] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing > [public->public_extensions] continue=false > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) > [public_extensions] destination_number(972592182076) =~ > /^(10[01][0-9])$/ break=on-false > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing [public->TEMP] > continue=false > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (PASS) [TEMP] > context(public) =~ /public/ break=on-false > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [TEMP] > destination_number(972592182076) =~ /^publicnumber$/ break=on-false > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing > [public->Misc_Number] continue=false > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (PASS) > [Misc_Number] context(public) =~ /public/ break=on-false > Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) > [Misc_Number] destination_number(972592182076) =~ /^publicnumber2$/ > break=on-false > > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:119 > (sofia/external/1010 at xx.xx.xx.xx:5060) State Change CS_ROUTING -> > CS_EXECUTE 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 > Send signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:373 > (sofia/external/1010 at xx.xx.xx.xx:5060) State ROUTING going to sleep > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 > (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_EXECUTE > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:380 > (sofia/external/1010 at xx.xx.xx.xx:5060) State EXECUTE > 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:241 > sofia/external/1010 at xx.xx.xx.xx:5060 SOFIA EXECUTE > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:157 > sofia/external/1010 at xx.xx.xx.xx:5060 Standard EXECUTE > EXECUTE sofia/external/1010 at xx.xx.xx.xx:5060 set(outside_call=true) > 2011-11-18 15:27:36.633145 [DEBUG] mod_dptools.c:1063 > sofia/external/1010 at xx.xx.xx.xx:5060 SET [outside_call]=[true] > EXECUTE sofia/external/1010 at xx.xx.xx.xx:5060 set(RFC2822_DATE=Fri, 18 > Nov 2011 15:27:36 +0100) > 2011-11-18 15:27:36.633145 [DEBUG] mod_dptools.c:1063 > sofia/external/1010 at xx.xx.xx.xx:5060 SET [RFC2822_DATE]=[Fri, 18 Nov > 2011 15:27:36 +0100] > 2011-11-18 15:27:36.633145 [NOTICE] switch_core_state_machine.c:189 > sofia/external/1010 at xx.xx.xx.xx:5060 has executed the last dialplan > instruction, hanging up. > 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:2739 > (sofia/external/1010 at xx.xx.xx.xx:5060) Callstate Change RINGING -> HANGUP > 2011-11-18 15:27:36.633145 [NOTICE] switch_core_state_machine.c:191 > Hangup sofia/external/1010 at xx.xx.xx.xx:5060 [CS_EXECUTE] [NORMAL_CLEARING] > 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:2755 Send signal > sofia/external/1010 at xx.xx.xx.xx:5060 [KILL] > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send > signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:380 > (sofia/external/1010 at xx.xx.xx.xx:5060) State EXECUTE going to sleep > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 > (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_HANGUP > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:575 > (sofia/external/1010 at xx.xx.xx.xx:5060) State HANGUP > 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:458 Channel > sofia/external/1010 at xx.xx.xx.xx:5060 hanging up, cause: NORMAL_CLEARING > 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:522 Responding to INVITE > with: 480 > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:46 > sofia/external/1010 at xx.xx.xx.xx:5060 Standard HANGUP, cause: > NORMAL_CLEARING 2011-11-18 15:27:36.633145 [DEBUG] > switch_core_state_machine.c:575 (sofia/external/1010 at xx.xx.xx.xx:5060) > State HANGUP going to sleep 2011-11-18 15:27:36.633145 [DEBUG] > switch_core_state_machine.c:356 (sofia/external/1010 at xx.xx.xx.xx:5060) > State Change CS_HANGUP -> > CS_REPORTING > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send > signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 > (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_REPORTING > 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:635 > (sofia/external/1010 at xx.xx.xx.xx:5060) State REPORTING > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 msc at freeswitch.org Mon Nov 21 19:24:35 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Nov 2011 08:24:35 -0800 Subject: [Freeswitch-users] Call to registered user In-Reply-To: <4EC9DB1F.1000001@tagnet.ru> References: <4EC9DB1F.1000001@tagnet.ru> Message-ID: On Sun, Nov 20, 2011 at 9:01 PM, Boris Kovalenko wrote: > Hello! > > My FS has different sip profiles (internet, local net, dmz etc). > There is some type of mobile users that may register with any profile. > So, is there a simple way to call registered user without known on which > profile it has registered? AFAIK sofia_contact need profile and user/ is > looking in profile of incoming call. > sofia_contact *can* accept a profile argument but it is not required. The required argument is "user at domain". If you don't have multiple domains then there should not be a reason that you cannot use sofia_contact. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111121/01af56c7/attachment.html From anthony.minessale at gmail.com Mon Nov 21 19:57:36 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Nov 2011 10:57:36 -0600 Subject: [Freeswitch-users] Playback race condition In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> Message-ID: execute over event socket does the same thing as broadcast: The channel is very busy, its doing I/O in a very critical loop in its own thread. You cannot simply gain access to it and make it start executing your application because then it would be doing it inside that other thread and possibly stopping it when it's supposed to be doing something else. Instead the instruction is queued to the channel and the channel does periodic calls to see if it has any tasks to execute when its tolerable to perform such a task and also ensuring its carried out in it's own dedicated session thread. >From the time you send the command there is an expected latency until the command is actually executed. From your application you should send the command then wait until you receive the channel_execute corresponding to the desired command before continuing and having clearance to send the uuid_break. On Sun, Nov 20, 2011 at 2:07 AM, Hynek Cihlar wrote: > So why would the bgapi uuid_break all fail if it queues the same way as > execute does? > > Hynek > > > > > On Sun, Nov 20, 2011 at 4:27 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Every way you are discussing does it the same way, doing it over bgapi >> just wastes an extra thread for no reason. >> >> All of the above queue the command to the session and it will not execute >> it until the next time its convenient for the session to execute >> queued instructions. >> >> So any way you slice it, unless you wait for the execute event telling >> you that the playback you tried to start, has actually started then its not >> going to help. >> >> You are most likely trying to had implement some convention that is >> dumbed down to one command already in FS so I suggest you re-evaluate what >> you are even trying to do. >> >> >> >> On Sat, Nov 19, 2011 at 3:37 AM, Hynek Cihlar wrote: >> >>> So I changed the uuid_X-commands to execute and the use case with >>> stopping tone playback works like a charm now! >>> >>> Regarding the blocking, you are right. I'm limiting blocking (and >>> context switching) by handling all events and issuing commands always on >>> one (and the same) thread per session. This design also allows me to >>> simplify the programming model, decrease concurrency bugs, simplify >>> debugging, etc. That's why the synthetic delay was giving me troubles since >>> the events and commands must fly through as fast as possible. >>> >>> Thanks again! >>> >>> Hynek >>> >>> >>> >>> >>> On Sat, Nov 19, 2011 at 10:28 AM, Peter Olsson < >>> peter.olsson at visionutveckling.se> wrote: >>> >>>> bgapi can of course be used also, and in some cases you might still >>>> need it. But the important thing to keep in mind when doing bgapi is to >>>> handle the events correctly. When executing bgapi you will get a job-id >>>> back, this job id must then be stored internally, so you can link this to >>>> the background job event. >>>> >>>> I try to use the sendMsg/execute as much as possible, you will get >>>> CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE events, so you will always know >>>> what's going on. >>>> >>>> I've only tested my system for about 100 concurrent calls, but that was >>>> handled without any problems at all. What's really important is to make >>>> really sure you never execute something that will block - since that will >>>> cause everything to "hang". >>>> >>>> /Peter >>>> ________________________________________ >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >>>> freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [ >>>> hynek.cihlar at gmail.com] >>>> Skickat: den 19 november 2011 10:04 >>>> Till: FreeSWITCH Users Help >>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>> >>>> Peter, thanks for pointing me to the right direction. I will try the >>>> execute command, since queueing is exactly what I would need to preserve >>>> consistency across commands (I had an impression that bgapi will give me >>>> this consistency). >>>> >>>> Also I like to hear that it is possible to use a single connection >>>> since I have designed the communication with one connection also. May I ask >>>> how many concurrent sessions can your system handle? >>>> >>>> Hynek >>>> >>>> >>>> >>>> On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson < >>>> peter.olsson at visionutveckling.se>>> peter.olsson at visionutveckling.se>> wrote: >>>> If you think about performance I would say it's better to handle this >>>> without spawning lots of extra threads. >>>> >>>> First of all, most common commands can be queued into the the channel >>>> thread using the execute method ( >>>> http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes >>>> it possible to queue commands directly to the channel's thread, and the ESL >>>> session will never wait for the result, it returns immediately. >>>> >>>> I've never used uuid_broadcast myself, so I'm not really sure how it's >>>> handled, but by looking quickly in the code it looks like it shouldn't >>>> block either, since it's actually just queuing a playback command to the >>>> channel's thread. >>>> >>>> So basically, keep away from uuid_X-commands, and use execute method as >>>> much as possible, and when using uuid_x-methods, make sure that won't block >>>> - but if they do, make sure to handle handle the events for the bgapi >>>> correctly. >>>> >>>> I've built a complete IVR system using these methods, with a single ESL >>>> connection, and I've never had any performace issues. >>>> >>>> /Peter >>>> >>>> ________________________________________ >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org> [ >>>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>] för Hynek Cihlar [ >>>> hynek.cihlar at gmail.com] >>>> Skickat: den 19 november 2011 09:12 >>>> Till: FreeSWITCH Users Help >>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>> >>>> Two reasons I am not using api in this case. >>>> >>>> 1. Performance >>>> 2. Doesn't seem to work >>>> >>>> 1. Blocking issuing the command will eventually reduce the system >>>> throughput. Agree this is only a potential theoretical problem and I >>>> haven't gotten into production where I could get the actual results. >>>> >>>> 2. When issuing api uuid_playback and api uuid_break right after, I'm >>>> always getting an error response from the uuid_break. The ESL message body >>>> is "-ERR no reply". Nothing in the logs (no idea which log type/level will >>>> could generate other useful info). >>>> >>>> Hynek >>>> >>>> >>>> >>>> On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < >>>> peter.olsson at visionutveckling.se>>> peter.olsson at visionutveckling.se>>>> peter.olsson at visionutveckling.se>>> peter.olsson at visionutveckling.se>>> wrote: >>>> Since you're using bgapi, it's impossible to guarantee that >>>> uuid_broadcast has actually executed before you exeute uuid_break (they >>>> will be executed in two different threads). You will have to do this in a >>>> more controlled way (wait for events to show up etc.). And also, do you >>>> really need to use bgapi all the time? I'm not sure how uuid_broadcast is >>>> handled, but uuid_break is totally safe to use without bgapi, since it will >>>> just update a flag and then return immediately. I think uuid_broadcast will >>>> do the same thing - so getting rid of bgapi should be a good ebnough >>>> solution. >>>> >>>> /Peter >>>> ________________________________________ >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>>>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>> [ >>>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>>>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>>] för Hynek Cihlar >>>> [hynek.cihlar at gmail.com>>> hynek.cihlar at gmail.com>] >>>> Skickat: den 19 november 2011 08:31 >>>> Till: FreeSWITCH Users Help >>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>> >>>> As a workaround I put a synthetic a delay between the start and stop of >>>> the playback. I wouldn't want to keep this solution however since it blocks >>>> one of the worker threads (decreasing overall system throughput) and >>>> second, I really don't know what the synthetic delay should be or whether >>>> it could differ during different conditions like during a high load. >>>> >>>> Was anybody facing similar problem? How did you solve it? Am I using >>>> the API wrong and is there another way to invoke playback/stop? >>>> >>>> Thanks! >>>> Hynek >>>> >>>> >>>> >>>> On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar >>> >>> hynek.cihlar at gmail.com>>>>> hynek.cihlar at gmail.com>>>> hynek.cihlar at gmail.com>>>> wrote: >>>> Hello all, >>>> >>>> I got on shaky grounds issuing commands to Freeswitch over ESL. >>>> >>>> For example, issuing the following commands close enough (on my system >>>> at around 100 ms and less) causes troubles. >>>> >>>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>>> playback::tone_stream://%(150,4000,425);loops=-1 >>>> <100ms appart >>>> bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all >>>> previous playback is not stopped, the following is queued (not played) >>>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>>> playback::tone_stream://%(1000,4000,425);loops=-1 >>>> >>>> It looks like the first tone playback is not properly initialized when >>>> the break arrives. Waiting for the start of the first tone playback >>>> (waiting for the right event) is not a solution, I don't want the tone to >>>> be played at all if the events come this close. >>>> >>>> Any ideas? >>>> >>>> >>>> Hynek >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org>>> consulting at freeswitch.org> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org>>> FreeSWITCH-users at lists.freeswitch.org>>>> FreeSWITCH-users at lists.freeswitch.org>>> FreeSWITCH-users at lists.freeswitch.org>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org>>> FreeSWITCH-users at lists.freeswitch.org> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> !DSPAM:4ec770ca32761488716642! >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111121/6d8ca596/attachment-0001.html From hynek.cihlar at gmail.com Mon Nov 21 22:10:07 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Mon, 21 Nov 2011 20:10:07 +0100 Subject: [Freeswitch-users] Playback race condition In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> Message-ID: <4841539842807372022@unknownmsgid> I send execute playback at time t0. I send execute break all also at time t0. Execute playback is queued in the channel queue. Execute break all is also queued in the channel queue. Channel thread picks up the execute playback, posts the execute notification, playback is queued for playback. Channel thread picks up the execute break all command... Now, what is the particular reason that channel thread cannot stop the initiated playback? Having to wait for the execute notification to be able to stop the playback makes the esl protocol unnecessary complicated. It's like having to wait for a TV to come up to be able to turn it off. Sent from my mobile device On Nov 21, 2011, at 18:01, Anthony Minessale wrote: execute over event socket does the same thing as broadcast: The channel is very busy, its doing I/O in a very critical loop in its own thread. You cannot simply gain access to it and make it start executing your application because then it would be doing it inside that other thread and possibly stopping it when it's supposed to be doing something else. Instead the instruction is queued to the channel and the channel does periodic calls to see if it has any tasks to execute when its tolerable to perform such a task and also ensuring its carried out in it's own dedicated session thread. >From the time you send the command there is an expected latency until the command is actually executed. From your application you should send the command then wait until you receive the channel_execute corresponding to the desired command before continuing and having clearance to send the uuid_break. On Sun, Nov 20, 2011 at 2:07 AM, Hynek Cihlar wrote: > So why would the bgapi uuid_break all fail if it queues the same way as > execute does? > > Hynek > > > > > On Sun, Nov 20, 2011 at 4:27 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Every way you are discussing does it the same way, doing it over bgapi >> just wastes an extra thread for no reason. >> >> All of the above queue the command to the session and it will not execute >> it until the next time its convenient for the session to execute >> queued instructions. >> >> So any way you slice it, unless you wait for the execute event telling >> you that the playback you tried to start, has actually started then its not >> going to help. >> >> You are most likely trying to had implement some convention that is >> dumbed down to one command already in FS so I suggest you re-evaluate what >> you are even trying to do. >> >> >> >> On Sat, Nov 19, 2011 at 3:37 AM, Hynek Cihlar wrote: >> >>> So I changed the uuid_X-commands to execute and the use case with >>> stopping tone playback works like a charm now! >>> >>> Regarding the blocking, you are right. I'm limiting blocking (and >>> context switching) by handling all events and issuing commands always on >>> one (and the same) thread per session. This design also allows me to >>> simplify the programming model, decrease concurrency bugs, simplify >>> debugging, etc. That's why the synthetic delay was giving me troubles since >>> the events and commands must fly through as fast as possible. >>> >>> Thanks again! >>> >>> Hynek >>> >>> >>> >>> >>> On Sat, Nov 19, 2011 at 10:28 AM, Peter Olsson < >>> peter.olsson at visionutveckling.se> wrote: >>> >>>> bgapi can of course be used also, and in some cases you might still >>>> need it. But the important thing to keep in mind when doing bgapi is to >>>> handle the events correctly. When executing bgapi you will get a job-id >>>> back, this job id must then be stored internally, so you can link this to >>>> the background job event. >>>> >>>> I try to use the sendMsg/execute as much as possible, you will get >>>> CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE events, so you will always know >>>> what's going on. >>>> >>>> I've only tested my system for about 100 concurrent calls, but that was >>>> handled without any problems at all. What's really important is to make >>>> really sure you never execute something that will block - since that will >>>> cause everything to "hang". >>>> >>>> /Peter >>>> ________________________________________ >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >>>> freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [ >>>> hynek.cihlar at gmail.com] >>>> Skickat: den 19 november 2011 10:04 >>>> Till: FreeSWITCH Users Help >>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>> >>>> Peter, thanks for pointing me to the right direction. I will try the >>>> execute command, since queueing is exactly what I would need to preserve >>>> consistency across commands (I had an impression that bgapi will give me >>>> this consistency). >>>> >>>> Also I like to hear that it is possible to use a single connection >>>> since I have designed the communication with one connection also. May I ask >>>> how many concurrent sessions can your system handle? >>>> >>>> Hynek >>>> >>>> >>>> >>>> On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson < >>>> peter.olsson at visionutveckling.se>>> peter.olsson at visionutveckling.se>> wrote: >>>> If you think about performance I would say it's better to handle this >>>> without spawning lots of extra threads. >>>> >>>> First of all, most common commands can be queued into the the channel >>>> thread using the execute method ( >>>> http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes >>>> it possible to queue commands directly to the channel's thread, and the ESL >>>> session will never wait for the result, it returns immediately. >>>> >>>> I've never used uuid_broadcast myself, so I'm not really sure how it's >>>> handled, but by looking quickly in the code it looks like it shouldn't >>>> block either, since it's actually just queuing a playback command to the >>>> channel's thread. >>>> >>>> So basically, keep away from uuid_X-commands, and use execute method as >>>> much as possible, and when using uuid_x-methods, make sure that won't block >>>> - but if they do, make sure to handle handle the events for the bgapi >>>> correctly. >>>> >>>> I've built a complete IVR system using these methods, with a single ESL >>>> connection, and I've never had any performace issues. >>>> >>>> /Peter >>>> >>>> ________________________________________ >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org> [ >>>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>] för Hynek Cihlar [ >>>> hynek.cihlar at gmail.com] >>>> Skickat: den 19 november 2011 09:12 >>>> Till: FreeSWITCH Users Help >>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>> >>>> Two reasons I am not using api in this case. >>>> >>>> 1. Performance >>>> 2. Doesn't seem to work >>>> >>>> 1. Blocking issuing the command will eventually reduce the system >>>> throughput. Agree this is only a potential theoretical problem and I >>>> haven't gotten into production where I could get the actual results. >>>> >>>> 2. When issuing api uuid_playback and api uuid_break right after, I'm >>>> always getting an error response from the uuid_break. The ESL message body >>>> is "-ERR no reply". Nothing in the logs (no idea which log type/level will >>>> could generate other useful info). >>>> >>>> Hynek >>>> >>>> >>>> >>>> On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < >>>> peter.olsson at visionutveckling.se>>> peter.olsson at visionutveckling.se>>>> peter.olsson at visionutveckling.se>>> peter.olsson at visionutveckling.se>>> wrote: >>>> Since you're using bgapi, it's impossible to guarantee that >>>> uuid_broadcast has actually executed before you exeute uuid_break (they >>>> will be executed in two different threads). You will have to do this in a >>>> more controlled way (wait for events to show up etc.). And also, do you >>>> really need to use bgapi all the time? I'm not sure how uuid_broadcast is >>>> handled, but uuid_break is totally safe to use without bgapi, since it will >>>> just update a flag and then return immediately. I think uuid_broadcast will >>>> do the same thing - so getting rid of bgapi should be a good ebnough >>>> solution. >>>> >>>> /Peter >>>> ________________________________________ >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>>>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>> [ >>>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>>>> freeswitch-users-bounces at lists.freeswitch.org>>> freeswitch-users-bounces at lists.freeswitch.org>>] för Hynek Cihlar >>>> [hynek.cihlar at gmail.com>>> hynek.cihlar at gmail.com>] >>>> Skickat: den 19 november 2011 08:31 >>>> Till: FreeSWITCH Users Help >>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>> >>>> As a workaround I put a synthetic a delay between the start and stop of >>>> the playback. I wouldn't want to keep this solution however since it blocks >>>> one of the worker threads (decreasing overall system throughput) and >>>> second, I really don't know what the synthetic delay should be or whether >>>> it could differ during different conditions like during a high load. >>>> >>>> Was anybody facing similar problem? How did you solve it? Am I using >>>> the API wrong and is there another way to invoke playback/stop? >>>> >>>> Thanks! >>>> Hynek >>>> >>>> >>>> >>>> On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar >>> >>> hynek.cihlar at gmail.com>>>>> hynek.cihlar at gmail.com>>>> hynek.cihlar at gmail.com>>>> wrote: >>>> Hello all, >>>> >>>> I got on shaky grounds issuing commands to Freeswitch over ESL. >>>> >>>> For example, issuing the following commands close enough (on my system >>>> at around 100 ms and less) causes troubles. >>>> >>>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>>> playback::tone_stream://%(150,4000,425);loops=-1 >>>> <100ms appart >>>> bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all >>>> previous playback is not stopped, the following is queued (not played) >>>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>>> playback::tone_stream://%(1000,4000,425);loops=-1 >>>> >>>> It looks like the first tone playback is not properly initialized when >>>> the break arrives. Waiting for the start of the first tone playback >>>> (waiting for the right event) is not a solution, I don't want the tone to >>>> be played at all if the events come this close. >>>> >>>> Any ideas? >>>> >>>> >>>> Hynek >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org>>> consulting at freeswitch.org> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org>>> FreeSWITCH-users at lists.freeswitch.org>>>> FreeSWITCH-users at lists.freeswitch.org>>> FreeSWITCH-users at lists.freeswitch.org>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org>>> FreeSWITCH-users at lists.freeswitch.org> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> !DSPAM:4ec770ca32761488716642! >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111121/8db04d90/attachment-0001.html From x.liu at hw.ac.uk Mon Nov 21 22:12:19 2011 From: x.liu at hw.ac.uk (xl127) Date: Mon, 21 Nov 2011 19:12:19 +0000 Subject: [Freeswitch-users] problems in Using Embedded FreeSWITCH In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> Message-ID: <4ECAA293.50408@hw.ac.uk> Hello, Following the instructions in http://wiki.freeswitch.org/wiki/Embedding_FreeSWITCH I am trying to use Embedded FreeSwitch. I installed FS in Fedora linux as a root user in /usr/local/freeswitch. Now as non-root user, I compiled following codes: #include int main(int argc, char** argv) { switch_core_flag_t flags = SCF_USE_SQL; bool console = true; const char *err = NULL; switch_core_set_globals(); switch_core_init_and_modload(flags, console ? SWITCH_TRUE : SWITCH_FALSE, &err); switch_core_runtime_loop(!console); return 0; } And copy the executable and conf, mod, contents of lib of the installed FS to my $MyWorkingDir. My directory structure looks like: MyWorkingDir--bin/ (contains my executable: FS_Embed) --conf/ --mod/ --lib/ --grammars --sounds --scripts --log (I also tried to copy the contents of lib directory into MyWorkingDir) When I run my embedded FS: ./FS_Embed, I got error 2011-11-21 18:46:26.606783 [INFO] switch_event.c:631 Activate Eventing Engine. 2011-11-21 18:46:26.617941 [DEBUG] switch_event.c:610 Create event dispatch thread 0 FS_Embed: src/switch_xml.c:2225: switch_xml_open_cfg: Assertion `MAIN_XML_ROOT != ((void *)0)' failed. Aborted (core dumped) and occasionally on another run, I got error: 2011-11-21 16:56:51.756040 [INFO] switch_event.c:631 Activate Eventing Engine. 2011-11-21 16:56:51.767198 [DEBUG] switch_event.c:610 Create event dispatch thread 0 Segmentation fault (core dumped) It looks like there are some things wrong with my Environment Setup for Embedded FS, but after a while of checking/trying/googling I couldn't find what's wrong. Any advice please? Many thanks! Xing -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 From davidwaf at gmail.com Mon Nov 21 23:35:19 2011 From: davidwaf at gmail.com (David Wafula) Date: Mon, 21 Nov 2011 22:35:19 +0200 Subject: [Freeswitch-users] Kernel timer ...those with experience please ? Message-ID: Hi all, I have very recent git (20th Nov 2011) on linode instance on Ubuntu 10.04, kernel 2.6.39. When alone in a conference, the music plays very neatly and clear. When another person joins the conference, then everything becomes distorted, cant even hold a simple conversion. Reading around, everyone attributes this to kernel timer. Could the kind experienced souls share their knowledge on kernel timer, how it affects freeswitch, and how best to correct it please. BTW, am considering during new image build with the latest kernel, will it help ? David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111121/90f09c8b/attachment.html From charlie.orford at attackplan.net Tue Nov 22 00:33:29 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Mon, 21 Nov 2011 22:33:29 +0100 Subject: [Freeswitch-users] 401 Unauthorized / phone, router or my fs config? Message-ID: <4ECAC3A9.3080901@attackplan.net> Hello list I am an asterisk refugee and currently in the midst of moving our voip platform across to freeswitch. The goal is to have FS in the cloud (on a dedicated Linode virtual machine running Debian Squeeze), with all office phones (Aastra 57i units) connecting via the public internet. FS is compiled and running on the linode machine (using the latest git build from a week ago). It is setup to listen on the public IP only so there is no NAT happening at the server end. All relevant firewall ports are open (tcp/udp 5060, tcp/udp 5080 and udp 16384:32768). Because our office net connection has a dynamic IP, we are using (or trying to use) digest authentication rather than ACLs in order to control user/extension access to the internal sip profile. The problem: For some reason, none of our phones are able to successfully register with FS. Running fs_cli with logging at 7 and enabling "sofia global siptrace on" shows that the phones make contact and try to REGISTER but when FS replies with a 401 Unauthorized and requests the phone authenticate via digest, the phone seems to ignore this and just repeatedly keeps sending the same original REGISTER request with no accompanying Authorization header. My hunch is that the problem must lie with the phone or our router rather than FS but I'm a little out of my depth with this problem and so would appreciate any insight or advice. For a transcript of a failed registration between our FS server and a phone at the office, please see: http://pastebin.com/1qRudrvE (note: server and phone ip has been changed to protect the innocent). I also have a screen shot of the phone's SIP config here: http://imgur.com/2lwiN (we are running the latest publically available Aastra firmware on the phones - v3.2.2.56). Finally, in case it is relevant, the router at the office is a Draytek Vigor 2600 ADSL router (about 5 years old now but working happily as far as we know). Thanks + Regards, Charlie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111121/73f7833d/attachment.html From anthony.minessale at gmail.com Tue Nov 22 02:28:06 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Nov 2011 17:28:06 -0600 Subject: [Freeswitch-users] Playback race condition In-Reply-To: <4841539842807372022@unknownmsgid> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> <4841539842807372022@unknownmsgid> Message-ID: Your tone has changed to be unsatisfactorily arrogant to warrant my attention. Please refrain from any more disrespectful analogies. It would be like you came to a free forum to get help on a free software then questioned the response from the guy who designed it..... oh wait that's not an analogy that's what you are doing! If you sent sendevent with execute to send the playback then sent another one to execute break all (Assuming you did not have event_lock header which will force it to execute each instruction one at a time) it would indeed execute in order. You asked about using broadcast and uuid_break (broadcast is the same as sendmsg + execute) uuid_break is instant real time call so if you broadcast then uuid_break the break will easily beat the queuing. BTW I have 3 TV in my house all of which require you to wait for them to come all the way up to shut them back down. On Mon, Nov 21, 2011 at 1:10 PM, Hynek Cihlar wrote: > I send execute playback at time t0. I send execute break all also at time > t0. Execute playback is queued in the channel queue. Execute break all is > also queued in the channel queue. Channel thread picks up the execute > playback, posts the execute notification, playback is queued for playback. > Channel thread picks up the execute break all command... > > Now, what is the particular reason that channel thread cannot stop the > initiated playback? > > Having to wait for the execute notification to be able to stop the > playback makes the esl protocol unnecessary complicated. It's like having > to wait for a TV to come up to be able to turn it off. > > Sent from my mobile device > > On Nov 21, 2011, at 18:01, Anthony Minessale > wrote: > > execute over event socket does the same thing as broadcast: > > The channel is very busy, its doing I/O in a very critical loop in its own > thread. > You cannot simply gain access to it and make it start executing your > application because then it would be doing it inside that other thread and > possibly stopping it when it's supposed to be doing something else. > > Instead the instruction is queued to the channel and the channel does > periodic calls to see if it has any tasks to execute when its tolerable to > perform such a task and also ensuring its carried out in it's own dedicated > session thread. > > From the time you send the command there is an expected latency until the > command is actually executed. From your application you should send the > command then wait until you receive the channel_execute corresponding to > the desired command before continuing and having clearance to send the > uuid_break. > > > > On Sun, Nov 20, 2011 at 2:07 AM, Hynek Cihlar < > hynek.cihlar at gmail.com> wrote: > >> So why would the bgapi uuid_break all fail if it queues the same way as >> execute does? >> >> Hynek >> >> >> >> >> On Sun, Nov 20, 2011 at 4:27 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Every way you are discussing does it the same way, doing it over bgapi >>> just wastes an extra thread for no reason. >>> >>> All of the above queue the command to the session and it will not >>> execute it until the next time its convenient for the session to execute >>> queued instructions. >>> >>> So any way you slice it, unless you wait for the execute event telling >>> you that the playback you tried to start, has actually started then its not >>> going to help. >>> >>> You are most likely trying to had implement some convention that is >>> dumbed down to one command already in FS so I suggest you re-evaluate what >>> you are even trying to do. >>> >>> >>> >>> On Sat, Nov 19, 2011 at 3:37 AM, Hynek Cihlar < >>> hynek.cihlar at gmail.com> wrote: >>> >>>> So I changed the uuid_X-commands to execute and the use case with >>>> stopping tone playback works like a charm now! >>>> >>>> Regarding the blocking, you are right. I'm limiting blocking (and >>>> context switching) by handling all events and issuing commands always on >>>> one (and the same) thread per session. This design also allows me to >>>> simplify the programming model, decrease concurrency bugs, simplify >>>> debugging, etc. That's why the synthetic delay was giving me troubles since >>>> the events and commands must fly through as fast as possible. >>>> >>>> Thanks again! >>>> >>>> Hynek >>>> >>>> >>>> >>>> >>>> On Sat, Nov 19, 2011 at 10:28 AM, Peter Olsson < >>>> peter.olsson at visionutveckling.se> wrote: >>>> >>>>> bgapi can of course be used also, and in some cases you might still >>>>> need it. But the important thing to keep in mind when doing bgapi is to >>>>> handle the events correctly. When executing bgapi you will get a job-id >>>>> back, this job id must then be stored internally, so you can link this to >>>>> the background job event. >>>>> >>>>> I try to use the sendMsg/execute as much as possible, you will get >>>>> CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE events, so you will always know >>>>> what's going on. >>>>> >>>>> I've only tested my system for about 100 concurrent calls, but that >>>>> was handled without any problems at all. What's really important is to make >>>>> really sure you never execute something that will block - since that will >>>>> cause everything to "hang". >>>>> >>>>> /Peter >>>>> ________________________________________ >>>>> Fr?n: >>>>> freeswitch-users-bounces at lists.freeswitch.org [ >>>>> freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [ >>>>> hynek.cihlar at gmail.com] >>>>> Skickat: den 19 november 2011 10:04 >>>>> Till: FreeSWITCH Users Help >>>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>>> >>>>> Peter, thanks for pointing me to the right direction. I will try the >>>>> execute command, since queueing is exactly what I would need to preserve >>>>> consistency across commands (I had an impression that bgapi will give me >>>>> this consistency). >>>>> >>>>> Also I like to hear that it is possible to use a single connection >>>>> since I have designed the communication with one connection also. May I ask >>>>> how many concurrent sessions can your system handle? >>>>> >>>>> Hynek >>>>> >>>>> >>>>> >>>>> On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson < >>>>> peter.olsson at visionutveckling.se >>>>> peter.olsson at visionutveckling.se>> wrote: >>>>> If you think about performance I would say it's better to handle this >>>>> without spawning lots of extra threads. >>>>> >>>>> First of all, most common commands can be queued into the the channel >>>>> thread using the execute method ( >>>>> http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes >>>>> it possible to queue commands directly to the channel's thread, and the ESL >>>>> session will never wait for the result, it returns immediately. >>>>> >>>>> I've never used uuid_broadcast myself, so I'm not really sure how it's >>>>> handled, but by looking quickly in the code it looks like it shouldn't >>>>> block either, since it's actually just queuing a playback command to the >>>>> channel's thread. >>>>> >>>>> So basically, keep away from uuid_X-commands, and use execute method >>>>> as much as possible, and when using uuid_x-methods, make sure that won't >>>>> block - but if they do, make sure to handle handle the events for the bgapi >>>>> correctly. >>>>> >>>>> I've built a complete IVR system using these methods, with a single >>>>> ESL connection, and I've never had any performace issues. >>>>> >>>>> /Peter >>>>> >>>>> ________________________________________ >>>>> Fr?n: >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org> [ >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org>] för Hynek Cihlar >>>>> [ hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com>] >>>>> Skickat: den 19 november 2011 09:12 >>>>> Till: FreeSWITCH Users Help >>>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>>> >>>>> Two reasons I am not using api in this case. >>>>> >>>>> 1. Performance >>>>> 2. Doesn't seem to work >>>>> >>>>> 1. Blocking issuing the command will eventually reduce the system >>>>> throughput. Agree this is only a potential theoretical problem and I >>>>> haven't gotten into production where I could get the actual results. >>>>> >>>>> 2. When issuing api uuid_playback and api uuid_break right after, I'm >>>>> always getting an error response from the uuid_break. The ESL message body >>>>> is "-ERR no reply". Nothing in the logs (no idea which log type/level will >>>>> could generate other useful info). >>>>> >>>>> Hynek >>>>> >>>>> >>>>> >>>>> On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < >>>>> peter.olsson at visionutveckling.se >>>>> peter.olsson at visionutveckling.se> >>>>> peter.olsson at visionutveckling.se >>>>> peter.olsson at visionutveckling.se>>> wrote: >>>>> Since you're using bgapi, it's impossible to guarantee that >>>>> uuid_broadcast has actually executed before you exeute uuid_break (they >>>>> will be executed in two different threads). You will have to do this in a >>>>> more controlled way (wait for events to show up etc.). And also, do you >>>>> really need to use bgapi all the time? I'm not sure how uuid_broadcast is >>>>> handled, but uuid_break is totally safe to use without bgapi, since it will >>>>> just update a flag and then return immediately. I think uuid_broadcast will >>>>> do the same thing - so getting rid of bgapi should be a good ebnough >>>>> solution. >>>>> >>>>> /Peter >>>>> ________________________________________ >>>>> Fr?n: >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org> >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org>> [ >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org> >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org>>] för Hynek >>>>> Cihlar [ hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com> >>>>> hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com>>] >>>>> Skickat: den 19 november 2011 08:31 >>>>> Till: FreeSWITCH Users Help >>>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>>> >>>>> As a workaround I put a synthetic a delay between the start and stop >>>>> of the playback. I wouldn't want to keep this solution however since it >>>>> blocks one of the worker threads (decreasing overall system throughput) and >>>>> second, I really don't know what the synthetic delay should be or whether >>>>> it could differ during different conditions like during a high load. >>>>> >>>>> Was anybody facing similar problem? How did you solve it? Am I using >>>>> the API wrong and is there another way to invoke playback/stop? >>>>> >>>>> Thanks! >>>>> Hynek >>>>> >>>>> >>>>> >>>>> On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar < >>>>> hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com> >>>>> hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com>> >>>>> hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com> >>>>> hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com>>>> wrote: >>>>> Hello all, >>>>> >>>>> I got on shaky grounds issuing commands to Freeswitch over ESL. >>>>> >>>>> For example, issuing the following commands close enough (on my system >>>>> at around 100 ms and less) causes troubles. >>>>> >>>>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>>>> playback::tone_stream://%(150,4000,425);loops=-1 >>>>> <100ms appart >>>>> bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all >>>>> previous playback is not stopped, the following is queued (not played) >>>>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>>>> playback::tone_stream://%(1000,4000,425);loops=-1 >>>>> >>>>> It looks like the first tone playback is not properly initialized when >>>>> the break arrives. Waiting for the start of the first tone playback >>>>> (waiting for the right event) is not a solution, I don't want the tone to >>>>> be played at all if the events come this close. >>>>> >>>>> Any ideas? >>>>> >>>>> >>>>> Hynek >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> consulting at freeswitch.org> >>>>> consulting at freeswitch.org >>>>> consulting at freeswitch.org>> >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> <> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> FreeSWITCH-users at lists.freeswitch.org> >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> FreeSWITCH-users at lists.freeswitch.org>> >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> consulting at freeswitch.org> >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> <> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> FreeSWITCH-users at lists.freeswitch.org> >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> !DSPAM:4ec770ca32761488716642! >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> <> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> <> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> <> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> <> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111121/9102696f/attachment-0001.html From anthony.minessale at gmail.com Tue Nov 22 04:38:48 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Nov 2011 19:38:48 -0600 Subject: [Freeswitch-users] problems in Using Embedded FreeSWITCH In-Reply-To: <4ECAA293.50408@hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> <4ECAA293.50408@hw.ac.uk> Message-ID: try if (apr_initialize() != SWITCH_STATUS_SUCCESS) { fprintf(stderr, "FATAL ERROR! Could not initialize APR\n"); return 255; } early on in main() On Mon, Nov 21, 2011 at 1:12 PM, xl127 wrote: > Hello, > > Following the instructions in > http://wiki.freeswitch.org/wiki/Embedding_FreeSWITCH > I am trying to use Embedded FreeSwitch. > > I installed FS in Fedora linux as a root user in /usr/local/freeswitch. > Now as non-root user, > I compiled following codes: > > #include > int main(int argc, char** argv) > { > switch_core_flag_t flags = SCF_USE_SQL; > bool console = true; > const char *err = NULL; > switch_core_set_globals(); > switch_core_init_and_modload(flags, console ? SWITCH_TRUE : > SWITCH_FALSE, &err); > switch_core_runtime_loop(!console); > return 0; > } > > And copy the executable and conf, mod, contents of lib of the installed > FS to my $MyWorkingDir. > > My directory structure looks like: > MyWorkingDir--bin/ (contains my executable: FS_Embed) > --conf/ > --mod/ > --lib/ > --grammars > --sounds > --scripts > --log > (I also tried to copy the contents of lib directory into MyWorkingDir) > > When I run my embedded FS: ./FS_Embed, I got error > > 2011-11-21 18:46:26.606783 [INFO] switch_event.c:631 Activate Eventing > Engine. > 2011-11-21 18:46:26.617941 [DEBUG] switch_event.c:610 Create event > dispatch thread 0 > FS_Embed: src/switch_xml.c:2225: switch_xml_open_cfg: Assertion > `MAIN_XML_ROOT != ((void *)0)' failed. > Aborted (core dumped) > > and occasionally on another run, I got error: > > 2011-11-21 16:56:51.756040 [INFO] switch_event.c:631 Activate Eventing > Engine. > 2011-11-21 16:56:51.767198 [DEBUG] switch_event.c:610 Create event > dispatch thread 0 > Segmentation fault (core dumped) > > It looks like there are some things wrong with my Environment Setup for > Embedded FS, > but after a while of checking/trying/googling I couldn't find what's wrong. > > Any advice please? > > Many thanks! > Xing > > > > -- > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > Heriot-Watt University is the Sunday Times > Scottish University of the Year 2011-2012 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111121/9e94ed76/attachment.html From msc at freeswitch.org Tue Nov 22 06:03:41 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Nov 2011 19:03:41 -0800 Subject: [Freeswitch-users] Wednesday Conf Call - SIP Presence? Message-ID: Hi all, I would like to have someone speak on the subject of SIP presence this coming Wednesday. If you are up for giving the community a brief introduction to the concepts involved with SIP presence please let me know. Also, if you are experienced and/or knowledgeable on the subject but would rather not give a presentation also let me know as you may be in a position to type up some material that could be used by the presenter. Please email me off list. Thanks all! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111121/be7a9ebd/attachment.html From boris at tagnet.ru Tue Nov 22 07:14:25 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Tue, 22 Nov 2011 10:14:25 +0600 Subject: [Freeswitch-users] Call to registered user In-Reply-To: References: <4EC9DB1F.1000001@tagnet.ru> Message-ID: <4ECB21A1.1090600@tagnet.ru> Hello! Michael, please explain: sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 15020233664170-17093105930554 at 192.168.15.48 User: 230172 at rcen-nt.ru Contact: 230172 Agent: Greenlite ATOM V2.0 Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:19) EXPSECS(77) Host: voip-gw IP: 192.168.15.48 Port: 5060 Auth-User: 230172 Auth-Realm: rcen-nt.ru MWI-Account: 230172 at rcen-nt.ru Call-ID: 279492684728797-11326384312512 at 192.168.15.42 User: 230175 at rcen-nt.ru Contact: 230175 Agent: Greenlite ATOM V2.0 Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:28) EXPSECS(86) Host: voip-gw IP: 192.168.15.42 Port: 5060 Auth-User: 230175 Auth-Realm: rcen-nt.ru MWI-Account: 230175 at rcen-nt.ru Call-ID: 25095839831609-809787075531 at 192.168.15.46 User: 230171 at rcen-nt.ru Contact: 230171 Agent: Greenlite ATOM V2.0 Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:29) EXPSECS(87) Host: voip-gw IP: 192.168.15.46 Port: 5060 Auth-User: 230171 Auth-Realm: rcen-nt.ru MWI-Account: 230171 at rcen-nt.ru Call-ID: 1577286384367-21354577016286 at 192.168.15.45 User: 230173 at rcen-nt.ru Contact: 230173 Agent: Greenlite ATOM V2.0 Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:29) EXPSECS(87) Host: voip-gw IP: 192.168.15.45 Port: 5060 Auth-User: 230173 Auth-Realm: rcen-nt.ru MWI-Account: 230173 at rcen-nt.ru Call-ID: 59571353025698-224172903926056 at 192.168.15.43 User: 230174 at rcen-nt.ru Contact: 230174 Agent: Greenlite ATOM V2.0 Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:33) EXPSECS(91) Host: voip-gw IP: 192.168.15.43 Port: 5060 Auth-User: 230174 Auth-Realm: rcen-nt.ru MWI-Account: 230174 at rcen-nt.ru Total items returned: 5 ================================================================================================= freeswitch at internal> sofia_contact 230171 error/user_not_registered freeswitch at internal> sofia_contact 230171 at rcen-nt.ru error/user_not_registered freeswitch at internal> sofia_contact internal/230171 at rcent-nt.ru error/user_not_registered freeswitch at internal> sofia_contact internal/230171 sofia/internal/sip:230171 at 192.168.15.46:5060 freeswitch at internal> version FreeSWITCH Version 1.0.head (git-cdabd56 2011-11-17 08-23-21 +0100) > > > On Sun, Nov 20, 2011 at 9:01 PM, Boris Kovalenko > wrote: > > Hello! > > My FS has different sip profiles (internet, local net, dmz etc). > There is some type of mobile users that may register with any profile. > So, is there a simple way to call registered user without known on > which > profile it has registered? AFAIK sofia_contact need profile and > user/ is > looking in profile of incoming call. > > sofia_contact *can* accept a profile argument but it is not required. > The required argument is "user at domain". If you don't have multiple > domains then there should not be a reason that you cannot use > sofia_contact. > > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- ? ?????????, ????? ????????? ??? "??????" ???. +7 (3435) 230001 ???? +7 (3435) 230005 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/af63cccd/attachment-0001.html From hynek.cihlar at gmail.com Tue Nov 22 07:53:58 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Tue, 22 Nov 2011 05:53:58 +0100 Subject: [Freeswitch-users] Playback race condition In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> <4841539842807372022@unknownmsgid> Message-ID: <-5424725111232075662@unknownmsgid> I really didn't mean to be arrogant. The analogy was just to simplify the communication and express my concerns with the protocol, without the thorough understanding I don't think I can put together a serious esl app. I guess I will have to dive into the source code now and try to figure out by myself. Thank you for your inputs! Sent from my mobile device On Nov 22, 2011, at 0:30, Anthony Minessale wrote: Your tone has changed to be unsatisfactorily arrogant to warrant my attention. Please refrain from any more disrespectful analogies. It would be like you came to a free forum to get help on a free software then questioned the response from the guy who designed it..... oh wait that's not an analogy that's what you are doing! If you sent sendevent with execute to send the playback then sent another one to execute break all (Assuming you did not have event_lock header which will force it to execute each instruction one at a time) it would indeed execute in order. You asked about using broadcast and uuid_break (broadcast is the same as sendmsg + execute) uuid_break is instant real time call so if you broadcast then uuid_break the break will easily beat the queuing. BTW I have 3 TV in my house all of which require you to wait for them to come all the way up to shut them back down. On Mon, Nov 21, 2011 at 1:10 PM, Hynek Cihlar wrote: > I send execute playback at time t0. I send execute break all also at time > t0. Execute playback is queued in the channel queue. Execute break all is > also queued in the channel queue. Channel thread picks up the execute > playback, posts the execute notification, playback is queued for playback. > Channel thread picks up the execute break all command... > > Now, what is the particular reason that channel thread cannot stop the > initiated playback? > > Having to wait for the execute notification to be able to stop the > playback makes the esl protocol unnecessary complicated. It's like having > to wait for a TV to come up to be able to turn it off. > > Sent from my mobile device > > On Nov 21, 2011, at 18:01, Anthony Minessale > wrote: > > execute over event socket does the same thing as broadcast: > > The channel is very busy, its doing I/O in a very critical loop in its own > thread. > You cannot simply gain access to it and make it start executing your > application because then it would be doing it inside that other thread and > possibly stopping it when it's supposed to be doing something else. > > Instead the instruction is queued to the channel and the channel does > periodic calls to see if it has any tasks to execute when its tolerable to > perform such a task and also ensuring its carried out in it's own dedicated > session thread. > > From the time you send the command there is an expected latency until the > command is actually executed. From your application you should send the > command then wait until you receive the channel_execute corresponding to > the desired command before continuing and having clearance to send the > uuid_break. > > > > On Sun, Nov 20, 2011 at 2:07 AM, Hynek Cihlar < > hynek.cihlar at gmail.com> wrote: > >> So why would the bgapi uuid_break all fail if it queues the same way as >> execute does? >> >> Hynek >> >> >> >> >> On Sun, Nov 20, 2011 at 4:27 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Every way you are discussing does it the same way, doing it over bgapi >>> just wastes an extra thread for no reason. >>> >>> All of the above queue the command to the session and it will not >>> execute it until the next time its convenient for the session to execute >>> queued instructions. >>> >>> So any way you slice it, unless you wait for the execute event telling >>> you that the playback you tried to start, has actually started then its not >>> going to help. >>> >>> You are most likely trying to had implement some convention that is >>> dumbed down to one command already in FS so I suggest you re-evaluate what >>> you are even trying to do. >>> >>> >>> >>> On Sat, Nov 19, 2011 at 3:37 AM, Hynek Cihlar < >>> hynek.cihlar at gmail.com> wrote: >>> >>>> So I changed the uuid_X-commands to execute and the use case with >>>> stopping tone playback works like a charm now! >>>> >>>> Regarding the blocking, you are right. I'm limiting blocking (and >>>> context switching) by handling all events and issuing commands always on >>>> one (and the same) thread per session. This design also allows me to >>>> simplify the programming model, decrease concurrency bugs, simplify >>>> debugging, etc. That's why the synthetic delay was giving me troubles since >>>> the events and commands must fly through as fast as possible. >>>> >>>> Thanks again! >>>> >>>> Hynek >>>> >>>> >>>> >>>> >>>> On Sat, Nov 19, 2011 at 10:28 AM, Peter Olsson < >>>> peter.olsson at visionutveckling.se> wrote: >>>> >>>>> bgapi can of course be used also, and in some cases you might still >>>>> need it. But the important thing to keep in mind when doing bgapi is to >>>>> handle the events correctly. When executing bgapi you will get a job-id >>>>> back, this job id must then be stored internally, so you can link this to >>>>> the background job event. >>>>> >>>>> I try to use the sendMsg/execute as much as possible, you will get >>>>> CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE events, so you will always know >>>>> what's going on. >>>>> >>>>> I've only tested my system for about 100 concurrent calls, but that >>>>> was handled without any problems at all. What's really important is to make >>>>> really sure you never execute something that will block - since that will >>>>> cause everything to "hang". >>>>> >>>>> /Peter >>>>> ________________________________________ >>>>> Fr?n: >>>>> freeswitch-users-bounces at lists.freeswitch.org [ >>>>> freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar [ >>>>> hynek.cihlar at gmail.com] >>>>> Skickat: den 19 november 2011 10:04 >>>>> Till: FreeSWITCH Users Help >>>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>>> >>>>> Peter, thanks for pointing me to the right direction. I will try the >>>>> execute command, since queueing is exactly what I would need to preserve >>>>> consistency across commands (I had an impression that bgapi will give me >>>>> this consistency). >>>>> >>>>> Also I like to hear that it is possible to use a single connection >>>>> since I have designed the communication with one connection also. May I ask >>>>> how many concurrent sessions can your system handle? >>>>> >>>>> Hynek >>>>> >>>>> >>>>> >>>>> On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson < >>>>> peter.olsson at visionutveckling.se >>>>> peter.olsson at visionutveckling.se>> wrote: >>>>> If you think about performance I would say it's better to handle this >>>>> without spawning lots of extra threads. >>>>> >>>>> First of all, most common commands can be queued into the the channel >>>>> thread using the execute method ( >>>>> http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this makes >>>>> it possible to queue commands directly to the channel's thread, and the ESL >>>>> session will never wait for the result, it returns immediately. >>>>> >>>>> I've never used uuid_broadcast myself, so I'm not really sure how it's >>>>> handled, but by looking quickly in the code it looks like it shouldn't >>>>> block either, since it's actually just queuing a playback command to the >>>>> channel's thread. >>>>> >>>>> So basically, keep away from uuid_X-commands, and use execute method >>>>> as much as possible, and when using uuid_x-methods, make sure that won't >>>>> block - but if they do, make sure to handle handle the events for the bgapi >>>>> correctly. >>>>> >>>>> I've built a complete IVR system using these methods, with a single >>>>> ESL connection, and I've never had any performace issues. >>>>> >>>>> /Peter >>>>> >>>>> ________________________________________ >>>>> Fr?n: >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org> [ >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org>] för Hynek Cihlar >>>>> [ hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com>] >>>>> Skickat: den 19 november 2011 09:12 >>>>> Till: FreeSWITCH Users Help >>>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>>> >>>>> Two reasons I am not using api in this case. >>>>> >>>>> 1. Performance >>>>> 2. Doesn't seem to work >>>>> >>>>> 1. Blocking issuing the command will eventually reduce the system >>>>> throughput. Agree this is only a potential theoretical problem and I >>>>> haven't gotten into production where I could get the actual results. >>>>> >>>>> 2. When issuing api uuid_playback and api uuid_break right after, I'm >>>>> always getting an error response from the uuid_break. The ESL message body >>>>> is "-ERR no reply". Nothing in the logs (no idea which log type/level will >>>>> could generate other useful info). >>>>> >>>>> Hynek >>>>> >>>>> >>>>> >>>>> On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < >>>>> peter.olsson at visionutveckling.se >>>>> peter.olsson at visionutveckling.se> >>>>> peter.olsson at visionutveckling.se >>>>> peter.olsson at visionutveckling.se>>> wrote: >>>>> Since you're using bgapi, it's impossible to guarantee that >>>>> uuid_broadcast has actually executed before you exeute uuid_break (they >>>>> will be executed in two different threads). You will have to do this in a >>>>> more controlled way (wait for events to show up etc.). And also, do you >>>>> really need to use bgapi all the time? I'm not sure how uuid_broadcast is >>>>> handled, but uuid_break is totally safe to use without bgapi, since it will >>>>> just update a flag and then return immediately. I think uuid_broadcast will >>>>> do the same thing - so getting rid of bgapi should be a good ebnough >>>>> solution. >>>>> >>>>> /Peter >>>>> ________________________________________ >>>>> Fr?n: >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org> >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org>> [ >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org> >>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>> freeswitch-users-bounces at lists.freeswitch.org>>] för Hynek >>>>> Cihlar [ hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com> >>>>> hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com>>] >>>>> Skickat: den 19 november 2011 08:31 >>>>> Till: FreeSWITCH Users Help >>>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>>> >>>>> As a workaround I put a synthetic a delay between the start and stop >>>>> of the playback. I wouldn't want to keep this solution however since it >>>>> blocks one of the worker threads (decreasing overall system throughput) and >>>>> second, I really don't know what the synthetic delay should be or whether >>>>> it could differ during different conditions like during a high load. >>>>> >>>>> Was anybody facing similar problem? How did you solve it? Am I using >>>>> the API wrong and is there another way to invoke playback/stop? >>>>> >>>>> Thanks! >>>>> Hynek >>>>> >>>>> >>>>> >>>>> On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar < >>>>> hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com> >>>>> hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com>> >>>>> hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com> >>>>> hynek.cihlar at gmail.com >>>>> hynek.cihlar at gmail.com>>>> wrote: >>>>> Hello all, >>>>> >>>>> I got on shaky grounds issuing commands to Freeswitch over ESL. >>>>> >>>>> For example, issuing the following commands close enough (on my system >>>>> at around 100 ms and less) causes troubles. >>>>> >>>>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>>>> playback::tone_stream://%(150,4000,425);loops=-1 >>>>> <100ms appart >>>>> bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all >>>>> previous playback is not stopped, the following is queued (not played) >>>>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>>>> playback::tone_stream://%(1000,4000,425);loops=-1 >>>>> >>>>> It looks like the first tone playback is not properly initialized when >>>>> the break arrives. Waiting for the start of the first tone playback >>>>> (waiting for the right event) is not a solution, I don't want the tone to >>>>> be played at all if the events come this close. >>>>> >>>>> Any ideas? >>>>> >>>>> >>>>> Hynek >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> consulting at freeswitch.org> >>>>> consulting at freeswitch.org >>>>> consulting at freeswitch.org>> >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> <> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> FreeSWITCH-users at lists.freeswitch.org> >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> FreeSWITCH-users at lists.freeswitch.org>> >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> consulting at freeswitch.org> >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> <> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> FreeSWITCH-users at lists.freeswitch.org> >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> !DSPAM:4ec770ca32761488716642! >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> <> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> <> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> <> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> <> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111122/d3f726a8/attachment-0001.html From msc at freeswitch.org Tue Nov 22 09:04:16 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Nov 2011 22:04:16 -0800 Subject: [Freeswitch-users] Call to registered user In-Reply-To: <4ECB21A1.1090600@tagnet.ru> References: <4EC9DB1F.1000001@tagnet.ru> <4ECB21A1.1090600@tagnet.ru> Message-ID: Boris, Do "eval ${domain}" and see what shakes out - IP address or what? Try this: expand sofia_contact 230172@${domain} -MC On Mon, Nov 21, 2011 at 8:14 PM, Boris Kovalenko wrote: > Hello! > > Michael, please explain: > > sofia status profile internal reg > Registrations: > > ================================================================================================= > Call-ID: 15020233664170-17093105930554 at 192.168.15.48 > User: 230172 at rcen-nt.ru > Contact: 230172 > Agent: Greenlite ATOM V2.0 > Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:19) > EXPSECS(77) > Host: voip-gw > IP: 192.168.15.48 > Port: 5060 > Auth-User: 230172 > Auth-Realm: rcen-nt.ru > MWI-Account: 230172 at rcen-nt.ru > > Call-ID: 279492684728797-11326384312512 at 192.168.15.42 > User: 230175 at rcen-nt.ru > Contact: 230175 > Agent: Greenlite ATOM V2.0 > Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:28) > EXPSECS(86) > Host: voip-gw > IP: 192.168.15.42 > Port: 5060 > Auth-User: 230175 > Auth-Realm: rcen-nt.ru > MWI-Account: 230175 at rcen-nt.ru > > Call-ID: 25095839831609-809787075531 at 192.168.15.46 > User: 230171 at rcen-nt.ru > Contact: 230171 > Agent: Greenlite ATOM V2.0 > Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:29) > EXPSECS(87) > Host: voip-gw > IP: 192.168.15.46 > Port: 5060 > Auth-User: 230171 > Auth-Realm: rcen-nt.ru > MWI-Account: 230171 at rcen-nt.ru > > Call-ID: 1577286384367-21354577016286 at 192.168.15.45 > User: 230173 at rcen-nt.ru > Contact: 230173 > Agent: Greenlite ATOM V2.0 > Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:29) > EXPSECS(87) > Host: voip-gw > IP: 192.168.15.45 > Port: 5060 > Auth-User: 230173 > Auth-Realm: rcen-nt.ru > MWI-Account: 230173 at rcen-nt.ru > > Call-ID: 59571353025698-224172903926056 at 192.168.15.43 > User: 230174 at rcen-nt.ru > Contact: 230174 > Agent: Greenlite ATOM V2.0 > Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:33) > EXPSECS(91) > Host: voip-gw > IP: 192.168.15.43 > Port: 5060 > Auth-User: 230174 > Auth-Realm: rcen-nt.ru > MWI-Account: 230174 at rcen-nt.ru > > Total items returned: 5 > > ================================================================================================= > > freeswitch at internal> sofia_contact 230171 > error/user_not_registered > > freeswitch at internal> sofia_contact 230171 at rcen-nt.ru > error/user_not_registered > > freeswitch at internal> sofia_contact internal/230171 at rcent-nt.ru > error/user_not_registered > > freeswitch at internal> sofia_contact internal/230171 > sofia/internal/sip:230171 at 192.168.15.46:5060 > > freeswitch at internal> version > FreeSWITCH Version 1.0.head (git-cdabd56 2011-11-17 08-23-21 +0100) > > > > > On Sun, Nov 20, 2011 at 9:01 PM, Boris Kovalenko wrote: > >> Hello! >> >> My FS has different sip profiles (internet, local net, dmz etc). >> There is some type of mobile users that may register with any profile. >> So, is there a simple way to call registered user without known on which >> profile it has registered? AFAIK sofia_contact need profile and user/ is >> looking in profile of incoming call. >> > sofia_contact *can* accept a profile argument but it is not required. The > required argument is "user at domain". If you don't have multiple domains > then there should not be a reason that you cannot use sofia_contact. > > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > ? ?????????, > ????? ????????? > ??? "??????" > ???. +7 (3435) 230001 > ???? +7 (3435) 230005 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111121/08ae63e7/attachment.html From boris at tagnet.ru Tue Nov 22 09:09:41 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Tue, 22 Nov 2011 12:09:41 +0600 Subject: [Freeswitch-users] Call to registered user In-Reply-To: References: <4EC9DB1F.1000001@tagnet.ru> <4ECB21A1.1090600@tagnet.ru> Message-ID: <4ECB3CA5.8030705@tagnet.ru> Hello! freeswitch at internal> eval ${domain} rcen-nt.ru freeswitch at internal> expand sofia_contact 230172@${domain} error/user_not_registered > Boris, > > Do "eval ${domain}" and see what shakes out - IP address or what? Try > this: > > expand sofia_contact 230172@${domain} > > -MC > > On Mon, Nov 21, 2011 at 8:14 PM, Boris Kovalenko > wrote: > > Hello! > > Michael, please explain: > > sofia status profile internal reg > Registrations: > ================================================================================================= > Call-ID: 15020233664170-17093105930554 at 192.168.15.48 > > User: 230172 at rcen-nt.ru > Contact: 230172 > Agent: Greenlite ATOM V2.0 > Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:19) > EXPSECS(77) > Host: voip-gw > IP: 192.168.15.48 > Port: 5060 > Auth-User: 230172 > Auth-Realm: rcen-nt.ru > MWI-Account: 230172 at rcen-nt.ru > > Call-ID: 279492684728797-11326384312512 at 192.168.15.42 > > User: 230175 at rcen-nt.ru > Contact: 230175 > Agent: Greenlite ATOM V2.0 > Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:28) > EXPSECS(86) > Host: voip-gw > IP: 192.168.15.42 > Port: 5060 > Auth-User: 230175 > Auth-Realm: rcen-nt.ru > MWI-Account: 230175 at rcen-nt.ru > > Call-ID: 25095839831609-809787075531 at 192.168.15.46 > > User: 230171 at rcen-nt.ru > Contact: 230171 > Agent: Greenlite ATOM V2.0 > Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:29) > EXPSECS(87) > Host: voip-gw > IP: 192.168.15.46 > Port: 5060 > Auth-User: 230171 > Auth-Realm: rcen-nt.ru > MWI-Account: 230171 at rcen-nt.ru > > Call-ID: 1577286384367-21354577016286 at 192.168.15.45 > > User: 230173 at rcen-nt.ru > Contact: 230173 > Agent: Greenlite ATOM V2.0 > Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:29) > EXPSECS(87) > Host: voip-gw > IP: 192.168.15.45 > Port: 5060 > Auth-User: 230173 > Auth-Realm: rcen-nt.ru > MWI-Account: 230173 at rcen-nt.ru > > Call-ID: 59571353025698-224172903926056 at 192.168.15.43 > > User: 230174 at rcen-nt.ru > Contact: 230174 > Agent: Greenlite ATOM V2.0 > Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:33) > EXPSECS(91) > Host: voip-gw > IP: 192.168.15.43 > Port: 5060 > Auth-User: 230174 > Auth-Realm: rcen-nt.ru > MWI-Account: 230174 at rcen-nt.ru > > Total items returned: 5 > ================================================================================================= > > freeswitch at internal> sofia_contact 230171 > error/user_not_registered > > freeswitch at internal> sofia_contact 230171 at rcen-nt.ru > > error/user_not_registered > > freeswitch at internal> sofia_contact internal/230171 at rcent-nt.ru > > error/user_not_registered > > freeswitch at internal> sofia_contact internal/230171 > sofia/internal/sip:230171 at 192.168.15.46:5060 > > freeswitch at internal> version > FreeSWITCH Version 1.0.head (git-cdabd56 2011-11-17 08-23-21 +0100) > > >> >> >> On Sun, Nov 20, 2011 at 9:01 PM, Boris Kovalenko > > wrote: >> >> Hello! >> >> My FS has different sip profiles (internet, local net, >> dmz etc). >> There is some type of mobile users that may register with any >> profile. >> So, is there a simple way to call registered user without >> known on which >> profile it has registered? AFAIK sofia_contact need profile >> and user/ is >> looking in profile of incoming call. >> >> sofia_contact *can* accept a profile argument but it is not >> required. The required argument is "user at domain". If you don't >> have multiple domains then there should not be a reason that you >> cannot use sofia_contact. >> >> -MC >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > -- > ? ?????????, > ????? ????????? > ??? "??????" > ???.+7 (3435) 230001 > ????+7 (3435) 230005 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Regards, Boris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/8d2b0c8a/attachment-0001.html From vetali100 at gmail.com Tue Nov 22 10:16:54 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Mon, 21 Nov 2011 23:16:54 -0800 Subject: [Freeswitch-users] 401 Unauthorized / phone, router or my fs config? In-Reply-To: <4ECAC3A9.3080901@attackplan.net> References: <4ECAC3A9.3080901@attackplan.net> Message-ID: Most probably some NAT issue happens on the client side. Router is not doing port translation as required. FS replies to a port which is indicated in REGISTER request (correct), however client expects the reply on a different port. Try to enable in the profile. http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB_.28A.K.A._No_device_left_behind.29 Please reply if this helped. Vitalie 2011/11/21 Charlie Orford > Hello list > > I am an asterisk refugee and currently in the midst of moving our voip > platform across to freeswitch. The goal is to have FS in the cloud (on a > dedicated Linode virtual machine running Debian Squeeze), with all office > phones (Aastra 57i units) connecting via the public internet. > > FS is compiled and running on the linode machine (using the latest git > build from a week ago). It is setup to listen on the public IP only so > there is no NAT happening at the server end. All relevant firewall ports > are open (tcp/udp 5060, tcp/udp 5080 and udp 16384:32768). > > Because our office net connection has a dynamic IP, we are using (or > trying to use) digest authentication rather than ACLs in order to control > user/extension access to the internal sip profile. > > The problem: > > For some reason, none of our phones are able to successfully register with > FS. Running fs_cli with logging at 7 and enabling "sofia global siptrace > on" shows that the phones make contact and try to REGISTER but when FS > replies with a 401 Unauthorized and requests the phone authenticate via > digest, the phone seems to ignore this and just repeatedly keeps sending > the same original REGISTER request with no accompanying Authorization > header. > > My hunch is that the problem must lie with the phone or our router rather > than FS but I'm a little out of my depth with this problem and so would > appreciate any insight or advice. > > > For a transcript of a failed registration between our FS server and a > phone at the office, please see: http://pastebin.com/1qRudrvE (note: > server and phone ip has been changed to protect the innocent). > > I also have a screen shot of the phone's SIP config here: > http://imgur.com/2lwiN (we are running the latest publically available > Aastra firmware on the phones - v3.2.2.56). > > Finally, in case it is relevant, the router at the office is a Draytek > Vigor 2600 ADSL router (about 5 years old now but working happily as far as > we know). > > > Thanks + Regards, > Charlie > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111121/40120d2e/attachment.html From debianmailz at gmail.com Tue Nov 22 10:42:29 2011 From: debianmailz at gmail.com (D M) Date: Tue, 22 Nov 2011 08:42:29 +0100 Subject: [Freeswitch-users] SIP invalid call attempts from unknown dialer In-Reply-To: <201111211111.53970.sos@sokhapkin.dyndns.org> References: <4ECA2F4E.3060909@gmail.com> <201111211111.53970.sos@sokhapkin.dyndns.org> Message-ID: <4ECB5265.5080604@gmail.com> Hello, I already have fail2ban configured with 2 freeswitch filters, normally adding another fail2ban filter would be fine but the log does not show what ip the attempt is coming from and xx.xx.xx.xx in the log is the IP of the freeswitch machine, obviously blocking that would be bad. Regards, Daniel Sergey Okhapkin skrev 2011-11-21 17:11: > http://wiki.freeswitch.org/wiki/Fail2ban > > On Monday 21 November 2011, D M wrote: >> Hello, >> I have noticed quite a few different attempts on accessing my freeswitch >> machine via SIP that does not come from the company. I have attached >> part of a log with such an attempt. >> >> My main concern is ensuring that similar attempts will not be able to >> make external calls. I realize that any number made available externally >> will also be accessable via this method so my secondary concern is >> throttling or preventing these type of attempts to avoid autodialer spam. >> >> This log repeats with around 12 call attempts per second for almost a >> minute times with different attempts on seemingly random numbers. There >> has been multiple different attempts to spam random numbers via SIP but >> so far none has been successful. This log is the most relevant since it >> made a single login attempt on an nonexistent user after which it has >> either successfully spoofed the ip of the freeswitch machine or used an >> vulnerability in either my config or freeswitch. >> >> My config is the default freeswitch+fusionpbx installation on Ubuntu >> 10.04.3 LTS with instructions from here >> (http://wiki.fusionpbx.com/index.php?title=Easy_Ubuntu_10.04&oldid=1574). >> With a few minor configuration changes: >> * Registering is done via external domain pointing to the freeswitch >> machine, NOT using the default port 5060 >> * Port 5060 is generally used for traffic with SIP provider that >> connects us to phone network but the port not firewalled/restricted in >> any other way >> >> This is an example log of a single login attempt and single call >> attempt, the following modifications have been made: >> * Freeswitch public ip has been changed to xx.xx.xx.xx >> * 2 regexps have been changed from public telephone number to >> /^publicnumber$/ and /^publicnumber2$/ >> * A large list of regexps have been replaced with >> >> Please let me know if you need any more details or longer logs >> >> Thanks, >> Daniel >> >> ##### LOG BEGIN ##### >> >> 2011-11-18 15:27:33.293146 [WARNING] sofia_reg.c:2283 Can't find user >> [1010 at xx.xx.xx.xx] >> You must define a domain called 'xx.xx.xx.xx' in your directory and add >> a user with the id="1010" attribute >> and you must configure your device to use the proper domain in it's >> authentication credentials. >> 2011-11-18 15:27:36.633145 [NOTICE] switch_channel.c:897 New Channel >> sofia/external/1010 at xx.xx.xx.xx:5060 [75cf1808-11f1-11e1-9c95-494fea388543] >> 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5084 Channel >> sofia/external/1010 at xx.xx.xx.xx:5060 entering state [received][100] >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 >> (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_NEW >> 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5095 Remote SDP: >> v=0^M >> o=1010 13216264671138 13216264671138 IN IP4 192.168.1.3^M >> s=VaxSoft^M >> c=IN IP4 192.168.1.3^M >> t=0 0^M >> m=audio 7000 RTP/AVP 0 8 3 98 101^M >> a=rtpmap:0 PCMU/8000^M >> a=rtpmap:8 PCMA/8000^M >> a=rtpmap:3 GSM/8000^M >> a=rtpmap:98 iLBC/8000^M >> a=rtpmap:101 telephone-event/8000^M >> a=fmtp:101 0-16^M >> >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:343 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State NEW >> 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4711 Audio Codec Compare >> [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000] >> 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4711 Audio Codec Compare >> [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] >> 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:2819 Set Codec >> sofia/external/1010 at xx.xx.xx.xx:5060 PCMA/8000 20 ms 160 samples 64000 bits >> 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4825 Set 2833 dtmf >> send/recv payload to 101 >> 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5284 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State Change CS_NEW -> CS_INIT >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send >> signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 >> (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_INIT >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:364 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State INIT >> 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:85 >> sofia/external/1010 at xx.xx.xx.xx:5060 SOFIA INIT >> 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:125 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State Change CS_INIT -> CS_ROUTING >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send >> signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:364 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State INIT going to sleep >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 >> (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_ROUTING >> 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:1821 >> (sofia/external/1010 at xx.xx.xx.xx:5060) Callstate Change DOWN -> RINGING >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:373 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State ROUTING >> 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:148 >> sofia/external/1010 at xx.xx.xx.xx:5060 SOFIA ROUTING >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:77 >> sofia/external/1010 at xx.xx.xx.xx:5060 Standard ROUTING >> 2011-11-18 15:27:36.633145 [INFO] mod_dialplan_xml.c:336 Processing >> MyName<1010>->972592182076 in context public >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing [public->unloop] >> continue=false >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing >> [public->outside_call] continue=true >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Absolute Condition >> [outside_call] >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Action >> set(outside_call=true) Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 >> Action >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing >> [public->call_debug] continue=true >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [call_debug] >> ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing >> [public->public_extensions] continue=false >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) >> [public_extensions] destination_number(972592182076) =~ >> /^(10[01][0-9])$/ break=on-false >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing [public->TEMP] >> continue=false >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (PASS) [TEMP] >> context(public) =~ /public/ break=on-false >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) [TEMP] >> destination_number(972592182076) =~ /^publicnumber$/ break=on-false >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 parsing >> [public->Misc_Number] continue=false >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (PASS) >> [Misc_Number] context(public) =~ /public/ break=on-false >> Dialplan: sofia/external/1010 at xx.xx.xx.xx:5060 Regex (FAIL) >> [Misc_Number] destination_number(972592182076) =~ /^publicnumber2$/ >> break=on-false >> >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:119 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State Change CS_ROUTING -> >> CS_EXECUTE 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 >> Send signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:373 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State ROUTING going to sleep >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 >> (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_EXECUTE >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:380 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State EXECUTE >> 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:241 >> sofia/external/1010 at xx.xx.xx.xx:5060 SOFIA EXECUTE >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:157 >> sofia/external/1010 at xx.xx.xx.xx:5060 Standard EXECUTE >> EXECUTE sofia/external/1010 at xx.xx.xx.xx:5060 set(outside_call=true) >> 2011-11-18 15:27:36.633145 [DEBUG] mod_dptools.c:1063 >> sofia/external/1010 at xx.xx.xx.xx:5060 SET [outside_call]=[true] >> EXECUTE sofia/external/1010 at xx.xx.xx.xx:5060 set(RFC2822_DATE=Fri, 18 >> Nov 2011 15:27:36 +0100) >> 2011-11-18 15:27:36.633145 [DEBUG] mod_dptools.c:1063 >> sofia/external/1010 at xx.xx.xx.xx:5060 SET [RFC2822_DATE]=[Fri, 18 Nov >> 2011 15:27:36 +0100] >> 2011-11-18 15:27:36.633145 [NOTICE] switch_core_state_machine.c:189 >> sofia/external/1010 at xx.xx.xx.xx:5060 has executed the last dialplan >> instruction, hanging up. >> 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:2739 >> (sofia/external/1010 at xx.xx.xx.xx:5060) Callstate Change RINGING -> HANGUP >> 2011-11-18 15:27:36.633145 [NOTICE] switch_core_state_machine.c:191 >> Hangup sofia/external/1010 at xx.xx.xx.xx:5060 [CS_EXECUTE] [NORMAL_CLEARING] >> 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:2755 Send signal >> sofia/external/1010 at xx.xx.xx.xx:5060 [KILL] >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send >> signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:380 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State EXECUTE going to sleep >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 >> (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_HANGUP >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:575 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State HANGUP >> 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:458 Channel >> sofia/external/1010 at xx.xx.xx.xx:5060 hanging up, cause: NORMAL_CLEARING >> 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:522 Responding to INVITE >> with: 480 >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:46 >> sofia/external/1010 at xx.xx.xx.xx:5060 Standard HANGUP, cause: >> NORMAL_CLEARING 2011-11-18 15:27:36.633145 [DEBUG] >> switch_core_state_machine.c:575 (sofia/external/1010 at xx.xx.xx.xx:5060) >> State HANGUP going to sleep 2011-11-18 15:27:36.633145 [DEBUG] >> switch_core_state_machine.c:356 (sofia/external/1010 at xx.xx.xx.xx:5060) >> State Change CS_HANGUP -> >> CS_REPORTING >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send >> signal sofia/external/1010 at xx.xx.xx.xx:5060 [BREAK] >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 >> (sofia/external/1010 at xx.xx.xx.xx:5060) Running State Change CS_REPORTING >> 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:635 >> (sofia/external/1010 at xx.xx.xx.xx:5060) State REPORTING >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 vetali100 at gmail.com Tue Nov 22 10:53:02 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Mon, 21 Nov 2011 23:53:02 -0800 Subject: [Freeswitch-users] new git - no more 'SIP auth challenge' messages in log In-Reply-To: References: <1320945234630-6982405.post@n2.nabble.com> <18097.1320948567@ccs.covici.com> Message-ID: Please don't consider me too annoying, even if it is true :) But how do we implement integration of FS and Fail2Ban now, after the new logic was implemented which removed register attempts from the log? Please help. Thanks Vitalie 2011/11/12 Vitalie Colosov > So how this integration with fail2ban should be performed? > Should we create a separate event_socket handler for listening > SWITCH_EVENT_AUTH? > And then log something into a file, which will be parsed by fail2ban? > > Or fail2ban is already able to listen such events? > > Could you please provide more details? > > Thank you, > Vitalie > > 2011/11/10 > > So what is the parameter to put those messages back to warning from >> DEBUG10? >> >> Jeff Lenk wrote: >> >> > Please see http://jira.freeswitch.org/browse/FS-3094 >> > >> > for a discussion on this. >> > >> > -- >> > View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/new-git-no-more-SIP-auth-challenge-messages-in-log-tp6980582p6982405.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 >> >> -- >> Your life is like a penny. You're going to lose it. The question is: >> How do >> you spend it? >> >> John Covici >> covici at ccs.covici.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/20111121/e03c10f5/attachment-0001.html From miha at softnet.si Tue Nov 22 13:00:04 2011 From: miha at softnet.si (Miha Zoubek) Date: Tue, 22 Nov 2011 11:00:04 +0100 Subject: [Freeswitch-users] How to disable one modul Message-ID: <4ECB72A4.6000404@softnet.si> Hi, I would like to disable mod_radius_cdr to start for the first time. I would like to have only the second one. Can someone help me where on configuration files or in which .c file this is set? 2011-11-22 09:16:35.541448 [DEBUG] switch_channel.c:1844 (sofia/internal/018108500 at xxx.xxx.xxx.xxx) Callstate Change DOWN -> RINGING 2011-11-22 09:16:35.541448 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/018108500 at xxx.xxx.xxx.xxx) State ROUTING 2011-11-22 09:16:35.541448 [DEBUG] mod_sofia.c:148 sofia/internal/018108500 at xxx.xxx.xxx.xxxSOFIA ROUTING 2011-11-22 09:16:35.541448 [DEBUG] mod_radius_cdr.c:156 [mod_radius_cdr] Entering my_on_routing, test, test 2011-11-22 09:16:35.541448 [DEBUG] mod_radius_cdr.c:374 [mod_radius_cdr] RADIUS Accounting OK 2011-11-22 09:16:35.541448 [DEBUG] switch_core_state_machine.c:104 sofia/internal/018108500 at xxx.xxx.xxx.xxxStandard ROUTING 2011-11-22 09:16:37.551441 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:018108501 at xxx.xxx.xxx.xxx:1417 [BREAK] 2011-11-22 09:16:37.551441 [DEBUG] mod_radius_cdr.c:156 [mod_radius_cdr] Entering my_on_routing, test, test 2011-11-22 09:16:37.551441 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/sip:018108501 at xxx.xxx.xxx.xxx:1417 [BREAK] 2011-11-22 09:16:37.551441 [DEBUG] mod_radius_cdr.c:374 [mod_radius_cdr] RADIUS Accounting OK 2011-11-22 09:16:37.551441 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:018108501 at xxx.xxx.xxx.xxx:1417) State ROUTING Regards, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/9ed4d17f/attachment.html From nagalenoj at gmail.com Tue Nov 22 14:54:04 2011 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Tue, 22 Nov 2011 17:24:04 +0530 Subject: [Freeswitch-users] using break application after bridge Message-ID: Hi, I wanted to cancel the bridge application after issuing it and I tried using 'break' application. break application didn't work. But, when I tried using playback application followed by break, it worked. connect sendmsg call-command: execute execute-app-name: answer sendmsg call-command: execute execute-app-name: bridge execute-app-arg: user/1001 sendmsg call-command: execute execute-app-name: break Logs: http://pastebin.freeswitch.org/17843 Git version as of 18/11/2011. -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/42d7ba9b/attachment.html From x.liu at hw.ac.uk Tue Nov 22 15:36:06 2011 From: x.liu at hw.ac.uk (xl127) Date: Tue, 22 Nov 2011 12:36:06 +0000 Subject: [Freeswitch-users] problems in Using Embedded FreeSWITCH In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> <4ECAA293.50408@hw.ac.uk> Message-ID: <4ECB9736.90005@hw.ac.uk> Thanks for the advice! I tried it but still got the same error. I added the #include and found it from /usr/local/sr/freeswitch/libs/apr/include rather than from the installed directory /usr/local/freeswitch. Hope it is the right one. any more clues? On 22/11/11 01:38, Anthony Minessale wrote: > try > > if (apr_initialize() != SWITCH_STATUS_SUCCESS) { > fprintf(stderr, "FATAL ERROR! Could not initialize APR\n"); > return 255; > } > > early on in main() > > > On Mon, Nov 21, 2011 at 1:12 PM, xl127 > wrote: > > Hello, > > Following the instructions in > http://wiki.freeswitch.org/wiki/Embedding_FreeSWITCH > I am trying to use Embedded FreeSwitch. > > I installed FS in Fedora linux as a root user in > /usr/local/freeswitch. > Now as non-root user, > I compiled following codes: > > #include > int main(int argc, char** argv) > { > switch_core_flag_t flags = SCF_USE_SQL; > bool console = true; > const char *err = NULL; > switch_core_set_globals(); > switch_core_init_and_modload(flags, console ? SWITCH_TRUE : > SWITCH_FALSE, &err); > switch_core_runtime_loop(!console); > return 0; > } > > And copy the executable and conf, mod, contents of lib of the > installed > FS to my $MyWorkingDir. > > My directory structure looks like: > MyWorkingDir--bin/ (contains my executable: FS_Embed) > --conf/ > --mod/ > --lib/ > --grammars > --sounds > --scripts > --log > (I also tried to copy the contents of lib directory into MyWorkingDir) > > When I run my embedded FS: ./FS_Embed, I got error > > 2011-11-21 18:46:26.606783 [INFO] switch_event.c:631 Activate Eventing > Engine. > 2011-11-21 18:46:26.617941 [DEBUG] switch_event.c:610 Create event > dispatch thread 0 > FS_Embed: src/switch_xml.c:2225: switch_xml_open_cfg: Assertion > `MAIN_XML_ROOT != ((void *)0)' failed. > Aborted (core dumped) > > and occasionally on another run, I got error: > > 2011-11-21 16:56:51.756040 [INFO] switch_event.c:631 Activate Eventing > Engine. > 2011-11-21 16:56:51.767198 [DEBUG] switch_event.c:610 Create event > dispatch thread 0 > Segmentation fault (core dumped) > > It looks like there are some things wrong with my Environment > Setup for > Embedded FS, > but after a while of checking/trying/googling I couldn't find > what's wrong. > > Any advice please? > > Many thanks! > Xing > > > > -- > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > Heriot-Watt University is the Sunday Times > Scottish University of the Year 2011-2012 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/f83f28c0/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/f83f28c0/attachment-0001.jpg From michel.daggelinckx at gmail.com Tue Nov 22 15:37:50 2011 From: michel.daggelinckx at gmail.com (Michel Daggelinckx) Date: Tue, 22 Nov 2011 13:37:50 +0100 Subject: [Freeswitch-users] Fwd: Newfies-Dialer - Voice Broadcast Application Launched In-Reply-To: References: Message-ID: ---------- Forwarded message ---------- From: Star2Billing Date: Mon, Nov 21, 2011 at 3:10 PM Subject: Newfies-Dialer - Voice Broadcast Application Launched To: "michel.daggelinckx at gmail.com" You are receiving this mail because you have downloaded one of Star2Billing's products (A2Billing, CDR-Stats, Newfies-Dialer) or we have installed it on your behalf. If you have no further interest in any of Star2Billing's products, please unsubscribe . Having trouble reading this? View it in your browser . [image: header bg] Newfies-Dialer - Voice Broadcasting Star2Billing S.L. launches Newfies-Dialer, a new application for mass voice broadcasting [image: space] Monday 21 November [image: space] [image: space] Newfies-Dialer 1.0 Released. [image: space] [image: space] Star2Billing S.L. is proud to announce the stable release of their latest product, Newfies-Dialer (http://www.newfies-dialer.org), a free and open source voice broadcast application for Freeswitch to enable the automated delivery of interactive phone calls to contacts, clients and the general public. Voice Broadcasting is a powerful business tool to automate phone calls for appointment reminders, delivery confirmations, event promotion, debt control, IVR campaigns, phone voting and surveys, automated teleconferencing and to relay vital information to your contacts over the phone. Newfies-Dialer?s voice broadcast system is powerful and adaptable: *> Scalable* ? Newfies-Dialer uses distributed message passing to make millions of calls per day via multiple cloud based instances of Freeswitch. *> Multi-User* ? Facilitates SaaS voice broadcast providers to let their customers create voice broadcast campaigns via an intuitive web interface. *> Integration* ? Comprehensive API ?s to integrate Newfies-Dialer into third party applications to automate call distribution. The Newfies-Dialer Voice Broadcasting Platform is assembled entirely from free and open source components including Freeswitch, Django, Plivo, Celery and RabbitMQ. For more details on Newfies-Dialer including comprehensive installation guides, an automated install script, user manuals and developer?s documentation view www.newfies-dialer.org, or contact us at newfies-dialer at star2billing.com [image: space] [image: space] [image: space] Other Information Get Started Install Newfies-Dialer with automated install scripts and step-by-step instructions. Newfies-Dialer Features Features and benefits of Newfies-Dialer [image: Forward to Friend] [image: Unsubscribe] [image: Follow us on Twitter] [image: header bg] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/817e897a/attachment.html From bwibowo at gmail.com Tue Nov 22 16:39:51 2011 From: bwibowo at gmail.com (bwibowo at gmail.com) Date: Tue, 22 Nov 2011 13:39:51 +0000 Subject: [Freeswitch-users] Sip client java based for mobile phone Message-ID: <194116879-1321969196-cardhu_decombobulator_blackberry.rim.net-625659330-@b26.c2.bise3.blackberry> As asked in subject, if somebody have this kind of product, please contact me directly Regards Budi wibowo From edpimentl at gmail.com Tue Nov 22 17:15:09 2011 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 22 Nov 2011 09:15:09 -0500 Subject: [Freeswitch-users] Sip client java based for mobile phone In-Reply-To: <194116879-1321969196-cardhu_decombobulator_blackberry.rim.net-625659330-@b26.c2.bise3.blackberry> References: <194116879-1321969196-cardhu_decombobulator_blackberry.rim.net-625659330-@b26.c2.bise3.blackberry> Message-ID: http://sipdroid.org/ From bwibowo at gmail.com Tue Nov 22 17:32:38 2011 From: bwibowo at gmail.com (bwibowo at gmail.com) Date: Tue, 22 Nov 2011 14:32:38 +0000 Subject: [Freeswitch-users] Sip client java based for mobile phone Message-ID: <2070922727-1321972363-cardhu_decombobulator_blackberry.rim.net-133929396-@b26.c2.bise3.blackberry> Not sure this can be run on old mobile phone that support java. ------Original Message------ From: EdPimentl Sender: freeswitch-users-bounces at lists.freeswitch.org To: FreeSWITCH Users Help ReplyTo: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sip client java based for mobile phone Sent: Nov 22, 2011 9:15 PM http://sipdroid.org/ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 hynek.cihlar at gmail.com Tue Nov 22 17:41:46 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Tue, 22 Nov 2011 15:41:46 +0100 Subject: [Freeswitch-users] Maximum number of gateways Message-ID: Hello all, is there a practical limit of the number of gateways per one Sofia profile? The documentation says that Sofia is single-threaded. Does it for example mean, that if a gateway is being registered that other net IO of the same profile is stalled? Thanks, Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/92a8a299/attachment.html From charlie.orford at attackplan.net Tue Nov 22 17:53:52 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Tue, 22 Nov 2011 15:53:52 +0100 Subject: [Freeswitch-users] 401 Unauthorized / phone, router or my fs config? In-Reply-To: References: <4ECAC3A9.3080901@attackplan.net> Message-ID: <4ECBB780.3060902@attackplan.net> Hi Vitalie Thank you very much for the hint, enabling worked and the phone was able to register. Watching the siptrace, one thing bothers me. Immediately after the phone registers, FS sends a NOTIFY but it looks like this is sent to the wrong port (i.e. it doesn't use the rport value). Is this correct? Pastebin copy of the transcript is here: http://pastebin.com/S2LWzrY7 I also decided to see if the Aastra phones had a setting to enable rport and it turned out they did. I tried enabling this and setting "NDLB-force-rport" back to false in the FS profile but this resulted in the phone being unable to register again. It gets further than before in that it replies to the FS server request for authorization, however, it seems the phone then tries to register again (this time specifying the rport value for the port in the Contact header), FS replies again with a 200 OK but the phone displays "No Service" and doesn't think it is registered. Running "sofia status profile internal" from the fs_cli seems to show the phone as registered twice (see: http://pastebin.com/PUWLNm7a ). For a pastebin of the complete registration transcript under this scenario, please see: http://pastebin.com/nRk8inVR Kind Regards, Charlie On 22/11/2011 08:16, Vitalie Colosov wrote: > Most probably some NAT issue happens on the client side. > > Router is not doing port translation as required. > > FS replies to a port which is indicated in REGISTER request (correct), > however client expects the reply on a different port. > > Try to enable in the > profile. > > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB_.28A.K.A._No_device_left_behind.29 > > > Please reply if this helped. > > Vitalie > > 2011/11/21 Charlie Orford > > > Hello list > > I am an asterisk refugee and currently in the midst of moving our > voip platform across to freeswitch. The goal is to have FS in the > cloud (on a dedicated Linode virtual machine running Debian > Squeeze), with all office phones (Aastra 57i units) connecting via > the public internet. > > FS is compiled and running on the linode machine (using the latest > git build from a week ago). It is setup to listen on the public IP > only so there is no NAT happening at the server end. All relevant > firewall ports are open (tcp/udp 5060, tcp/udp 5080 and udp > 16384:32768). > > Because our office net connection has a dynamic IP, we are using > (or trying to use) digest authentication rather than ACLs in order > to control user/extension access to the internal sip profile. > > The problem: > > For some reason, none of our phones are able to successfully > register with FS. Running fs_cli with logging at 7 and enabling > "sofia global siptrace on" shows that the phones make contact and > try to REGISTER but when FS replies with a 401 Unauthorized and > requests the phone authenticate via digest, the phone seems to > ignore this and just repeatedly keeps sending the same original > REGISTER request with no accompanying Authorization header. > > My hunch is that the problem must lie with the phone or our router > rather than FS but I'm a little out of my depth with this problem > and so would appreciate any insight or advice. > > > For a transcript of a failed registration between our FS server > and a phone at the office, please see: > http://pastebin.com/1qRudrvE (note: server and phone ip has been > changed to protect the innocent). > > I also have a screen shot of the phone's SIP config here: > http://imgur.com/2lwiN (we are running the latest publically > available Aastra firmware on the phones - v3.2.2.56). > > Finally, in case it is relevant, the router at the office is a > Draytek Vigor 2600 ADSL router (about 5 years old now but working > happily as far as we know). > > > Thanks + Regards, > Charlie > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111122/aa3fb4af/attachment.html From lazy.charles at gmail.com Tue Nov 22 18:16:36 2011 From: lazy.charles at gmail.com (Charles Wang) Date: Tue, 22 Nov 2011 23:16:36 +0800 Subject: [Freeswitch-users] (no subject) Message-ID: http://fitnessembassy.com/modules/mod_wdbanners/site.php?html143 From erik.dekkers at certhon.com Tue Nov 22 13:52:35 2011 From: erik.dekkers at certhon.com (Erik Dekkers) Date: Tue, 22 Nov 2011 11:52:35 +0100 Subject: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on groupcall Message-ID: Hi List, Following my recent message on this list "P-Asserted-Identity: "Outbound Call" on internal calls" I've managed to get the phones to display the correct callee name. Unfortunately adding origination_callee_id_name to the groupcall doesn't work since I can't predict which groupmember will pickup that call. Is there a way to make this work correctly? Regards, Erik Dekkers (wvds-nl) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/b29ef16e/attachment.html From anthony.minessale at gmail.com Tue Nov 22 19:08:15 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Nov 2011 10:08:15 -0600 Subject: [Freeswitch-users] problems in Using Embedded FreeSWITCH In-Reply-To: <4ECB9736.90005@hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> <4ECAA293.50408@hw.ac.uk> <4ECB9736.90005@hw.ac.uk> Message-ID: i know you needed apr, not sure, maybe try strace or gdb On Tue, Nov 22, 2011 at 6:36 AM, xl127 wrote: > Thanks for the advice! > > I tried it but still got the same error. > > I added the #include and found it from > /usr/local/sr/freeswitch/libs/apr/include > rather than from the installed directory /usr/local/freeswitch. Hope it is > the right one. > > any more clues? > > > > On 22/11/11 01:38, Anthony Minessale wrote: > > try > > if (apr_initialize() != SWITCH_STATUS_SUCCESS) { > fprintf(stderr, "FATAL ERROR! Could not initialize APR\n"); > return 255; > } > > early on in main() > > > On Mon, Nov 21, 2011 at 1:12 PM, xl127 wrote: > >> Hello, >> >> Following the instructions in >> http://wiki.freeswitch.org/wiki/Embedding_FreeSWITCH >> I am trying to use Embedded FreeSwitch. >> >> I installed FS in Fedora linux as a root user in /usr/local/freeswitch. >> Now as non-root user, >> I compiled following codes: >> >> #include >> int main(int argc, char** argv) >> { >> switch_core_flag_t flags = SCF_USE_SQL; >> bool console = true; >> const char *err = NULL; >> switch_core_set_globals(); >> switch_core_init_and_modload(flags, console ? SWITCH_TRUE : >> SWITCH_FALSE, &err); >> switch_core_runtime_loop(!console); >> return 0; >> } >> >> And copy the executable and conf, mod, contents of lib of the installed >> FS to my $MyWorkingDir. >> >> My directory structure looks like: >> MyWorkingDir--bin/ (contains my executable: FS_Embed) >> --conf/ >> --mod/ >> --lib/ >> --grammars >> --sounds >> --scripts >> --log >> (I also tried to copy the contents of lib directory into MyWorkingDir) >> >> When I run my embedded FS: ./FS_Embed, I got error >> >> 2011-11-21 18:46:26.606783 [INFO] switch_event.c:631 Activate Eventing >> Engine. >> 2011-11-21 18:46:26.617941 [DEBUG] switch_event.c:610 Create event >> dispatch thread 0 >> FS_Embed: src/switch_xml.c:2225: switch_xml_open_cfg: Assertion >> `MAIN_XML_ROOT != ((void *)0)' failed. >> Aborted (core dumped) >> >> and occasionally on another run, I got error: >> >> 2011-11-21 16:56:51.756040 [INFO] switch_event.c:631 Activate Eventing >> Engine. >> 2011-11-21 16:56:51.767198 [DEBUG] switch_event.c:610 Create event >> dispatch thread 0 >> Segmentation fault (core dumped) >> >> It looks like there are some things wrong with my Environment Setup for >> Embedded FS, >> but after a while of checking/trying/googling I couldn't find what's >> wrong. >> >> Any advice please? >> >> Many thanks! >> Xing >> >> >> >> -- >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> Heriot-Watt University is the Sunday Times >> Scottish University of the Year 2011-2012 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > ------------------------------ > > [image: MailScanner Signature HW] *Heriot-Watt University is the Sunday > Times > > Scottish University of the Year 2011-2012 > * > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/386b219b/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/386b219b/attachment-0001.jpe From x.liu at hw.ac.uk Tue Nov 22 19:20:40 2011 From: x.liu at hw.ac.uk (xl127) Date: Tue, 22 Nov 2011 16:20:40 +0000 Subject: [Freeswitch-users] problems in Using Embedded FreeSWITCH In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> <4ECAA293.50408@hw.ac.uk> <4ECB9736.90005@hw.ac.uk> Message-ID: <4ECBCBD8.3010105@hw.ac.uk> If I switch to root user I can successfully run my embedded FS app: FS_Embed. but it will fails if I run it as non-root user. One thing I can think of to do is re-install FS as non-root. I tried but it failed to access /usr/lib/python2.7/site-packages/freeswitch.py due to the permissions. On 22/11/11 16:08, Anthony Minessale wrote: > i know you needed apr, not sure, maybe try strace or gdb > > > On Tue, Nov 22, 2011 at 6:36 AM, xl127 > wrote: > > Thanks for the advice! > > I tried it but still got the same error. > > I added the #include and found it from > /usr/local/sr/freeswitch/libs/apr/include > rather than from the installed directory /usr/local/freeswitch. > Hope it is the right one. > > any more clues? > > > > On 22/11/11 01:38, Anthony Minessale wrote: >> try >> >> if (apr_initialize() != SWITCH_STATUS_SUCCESS) { >> fprintf(stderr, "FATAL ERROR! Could not initialize APR\n"); >> return 255; >> } >> >> early on in main() >> >> >> On Mon, Nov 21, 2011 at 1:12 PM, xl127 > > wrote: >> >> Hello, >> >> Following the instructions in >> http://wiki.freeswitch.org/wiki/Embedding_FreeSWITCH >> I am trying to use Embedded FreeSwitch. >> >> I installed FS in Fedora linux as a root user in >> /usr/local/freeswitch. >> Now as non-root user, >> I compiled following codes: >> >> #include >> int main(int argc, char** argv) >> { >> switch_core_flag_t flags = SCF_USE_SQL; >> bool console = true; >> const char *err = NULL; >> switch_core_set_globals(); >> switch_core_init_and_modload(flags, console ? SWITCH_TRUE : >> SWITCH_FALSE, &err); >> switch_core_runtime_loop(!console); >> return 0; >> } >> >> And copy the executable and conf, mod, contents of lib of the >> installed >> FS to my $MyWorkingDir. >> >> My directory structure looks like: >> MyWorkingDir--bin/ (contains my executable: FS_Embed) >> --conf/ >> --mod/ >> --lib/ >> --grammars >> --sounds >> --scripts >> --log >> (I also tried to copy the contents of lib directory into >> MyWorkingDir) >> >> When I run my embedded FS: ./FS_Embed, I got error >> >> 2011-11-21 18:46:26.606783 [INFO] switch_event.c:631 Activate >> Eventing >> Engine. >> 2011-11-21 18:46:26.617941 [DEBUG] switch_event.c:610 Create >> event >> dispatch thread 0 >> FS_Embed: src/switch_xml.c:2225: switch_xml_open_cfg: Assertion >> `MAIN_XML_ROOT != ((void *)0)' failed. >> Aborted (core dumped) >> >> and occasionally on another run, I got error: >> >> 2011-11-21 16:56:51.756040 [INFO] switch_event.c:631 Activate >> Eventing >> Engine. >> 2011-11-21 16:56:51.767198 [DEBUG] switch_event.c:610 Create >> event >> dispatch thread 0 >> Segmentation fault (core dumped) >> >> It looks like there are some things wrong with my Environment >> Setup for >> Embedded FS, >> but after a while of checking/trying/googling I couldn't find >> what's wrong. >> >> Any advice please? >> >> Many thanks! >> Xing >> >> >> >> -- >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> Heriot-Watt University is the Sunday Times >> Scottish University of the Year 2011-2012 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > ------------------------------------------------------------------------ > > MailScanner Signature HW *Heriot-Watt University is the Sunday Times > > Scottish University of the Year 2011-2012 > * > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/e0070b0c/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/e0070b0c/attachment-0001.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/e0070b0c/attachment-0001.jpg From anthony.minessale at gmail.com Tue Nov 22 19:47:55 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Nov 2011 10:47:55 -0600 Subject: [Freeswitch-users] problems in Using Embedded FreeSWITCH In-Reply-To: <4ECBCBD8.3010105@hw.ac.uk> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> <4ECAA293.50408@hw.ac.uk> <4ECB9736.90005@hw.ac.uk> <4ECBCBD8.3010105@hw.ac.uk> Message-ID: maybe your user does not have permissions to read and write the target install dir try to chown -R to your user on all of /usr/local/freeswitch you might want to try things the other way around and embedd your app in FS rather than embed FS in your app. On Tue, Nov 22, 2011 at 10:20 AM, xl127 wrote: > If I switch to root user I can successfully run my embedded FS app: > FS_Embed. > but it will fails if I run it as non-root user. > > One thing I can think of to do is re-install FS as non-root. > I tried but it failed to access > /usr/lib/python2.7/site-packages/freeswitch.py due to the permissions. > > > > On 22/11/11 16:08, Anthony Minessale wrote: > > i know you needed apr, not sure, maybe try strace or gdb > > > On Tue, Nov 22, 2011 at 6:36 AM, xl127 wrote: > >> Thanks for the advice! >> >> I tried it but still got the same error. >> >> I added the #include and found it from >> /usr/local/sr/freeswitch/libs/apr/include >> rather than from the installed directory /usr/local/freeswitch. Hope it >> is the right one. >> >> any more clues? >> >> >> >> On 22/11/11 01:38, Anthony Minessale wrote: >> >> try >> >> if (apr_initialize() != SWITCH_STATUS_SUCCESS) { >> fprintf(stderr, "FATAL ERROR! Could not initialize APR\n"); >> return 255; >> } >> >> early on in main() >> >> >> On Mon, Nov 21, 2011 at 1:12 PM, xl127 wrote: >> >>> Hello, >>> >>> Following the instructions in >>> http://wiki.freeswitch.org/wiki/Embedding_FreeSWITCH >>> I am trying to use Embedded FreeSwitch. >>> >>> I installed FS in Fedora linux as a root user in /usr/local/freeswitch. >>> Now as non-root user, >>> I compiled following codes: >>> >>> #include >>> int main(int argc, char** argv) >>> { >>> switch_core_flag_t flags = SCF_USE_SQL; >>> bool console = true; >>> const char *err = NULL; >>> switch_core_set_globals(); >>> switch_core_init_and_modload(flags, console ? SWITCH_TRUE : >>> SWITCH_FALSE, &err); >>> switch_core_runtime_loop(!console); >>> return 0; >>> } >>> >>> And copy the executable and conf, mod, contents of lib of the installed >>> FS to my $MyWorkingDir. >>> >>> My directory structure looks like: >>> MyWorkingDir--bin/ (contains my executable: FS_Embed) >>> --conf/ >>> --mod/ >>> --lib/ >>> --grammars >>> --sounds >>> --scripts >>> --log >>> (I also tried to copy the contents of lib directory into MyWorkingDir) >>> >>> When I run my embedded FS: ./FS_Embed, I got error >>> >>> 2011-11-21 18:46:26.606783 [INFO] switch_event.c:631 Activate Eventing >>> Engine. >>> 2011-11-21 18:46:26.617941 [DEBUG] switch_event.c:610 Create event >>> dispatch thread 0 >>> FS_Embed: src/switch_xml.c:2225: switch_xml_open_cfg: Assertion >>> `MAIN_XML_ROOT != ((void *)0)' failed. >>> Aborted (core dumped) >>> >>> and occasionally on another run, I got error: >>> >>> 2011-11-21 16:56:51.756040 [INFO] switch_event.c:631 Activate Eventing >>> Engine. >>> 2011-11-21 16:56:51.767198 [DEBUG] switch_event.c:610 Create event >>> dispatch thread 0 >>> Segmentation fault (core dumped) >>> >>> It looks like there are some things wrong with my Environment Setup for >>> Embedded FS, >>> but after a while of checking/trying/googling I couldn't find what's >>> wrong. >>> >>> Any advice please? >>> >>> Many thanks! >>> Xing >>> >>> >>> >>> -- >>> Heriot-Watt University is a Scottish charity >>> registered under charity number SC000278. >>> >>> Heriot-Watt University is the Sunday Times >>> Scottish University of the Year 2011-2012 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> ------------------------------ >> >> [image: MailScanner Signature HW] *Heriot-Watt University is the >> Sunday Times >> >> Scottish University of the Year 2011-2012 >> * >> >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > ------------------------------ > > [image: MailScanner Signature HW] *Heriot-Watt University is the Sunday > Times > Scottish University of the Year 2011-2012* > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/9aae9019/attachment-0003.jpe From anthony.minessale at gmail.com Tue Nov 22 22:22:35 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Nov 2011 13:22:35 -0600 Subject: [Freeswitch-users] using break application after bridge In-Reply-To: References: Message-ID: you can't use break to stop bridging. you can do "uuid_transfer park inline" to put it into an idle state On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: > Hi, > I wanted to cancel the bridge application after issuing it and I tried > using 'break' application. break application didn't work. But, when I tried > using playback application followed by break, it worked. > > connect > > sendmsg > call-command: execute > execute-app-name: answer > > sendmsg > call-command: execute > execute-app-name: bridge > execute-app-arg: user/1001 > > sendmsg > call-command: execute > execute-app-name: break > > Logs: > http://pastebin.freeswitch.org/17843 > > Git version as of 18/11/2011. > > -- > Regards, > Nagalenoj H. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/f1074b27/attachment.html From anthony.minessale at gmail.com Tue Nov 22 23:06:38 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Nov 2011 14:06:38 -0600 Subject: [Freeswitch-users] Playback race condition In-Reply-To: <-5424725111232075662@unknownmsgid> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> <4841539842807372022@unknownmsgid> <-5424725111232075662@unknownmsgid> Message-ID: I just explained it to you for what it's worth. On Mon, Nov 21, 2011 at 10:53 PM, Hynek Cihlar wrote: > I really didn't mean to be arrogant. The analogy was just to simplify the > communication and express my concerns with the protocol, without the > thorough understanding I don't think I can put together a serious esl app. > > I guess I will have to dive into the source code now and try to figure out > by myself. > > Thank you for your inputs! > > > Sent from my mobile device > > On Nov 22, 2011, at 0:30, Anthony Minessale > wrote: > > Your tone has changed to be unsatisfactorily arrogant to warrant my > attention. Please refrain from any more disrespectful analogies. > It would be like you came to a free forum to get help on a free software > then questioned the response from the guy who designed it..... oh wait > that's not an analogy that's what you are doing! > > > If you sent sendevent with execute to send the playback then sent another > one to execute break all (Assuming you did not have event_lock header which > will force it to execute each instruction one at a time) it would indeed > execute in order. > > You asked about using broadcast and uuid_break (broadcast is the same as > sendmsg + execute) uuid_break is instant real time call so if you broadcast > then uuid_break the break will easily beat the queuing. > > BTW I have 3 TV in my house all of which require you to wait for them to > come all the way up to shut them back down. > > > > > On Mon, Nov 21, 2011 at 1:10 PM, Hynek Cihlar < > hynek.cihlar at gmail.com> wrote: > >> I send execute playback at time t0. I send execute break all also at time >> t0. Execute playback is queued in the channel queue. Execute break all is >> also queued in the channel queue. Channel thread picks up the execute >> playback, posts the execute notification, playback is queued for playback. >> Channel thread picks up the execute break all command... >> >> Now, what is the particular reason that channel thread cannot stop the >> initiated playback? >> >> Having to wait for the execute notification to be able to stop the >> playback makes the esl protocol unnecessary complicated. It's like having >> to wait for a TV to come up to be able to turn it off. >> >> Sent from my mobile device >> >> On Nov 21, 2011, at 18:01, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >> execute over event socket does the same thing as broadcast: >> >> The channel is very busy, its doing I/O in a very critical loop in its >> own thread. >> You cannot simply gain access to it and make it start executing your >> application because then it would be doing it inside that other thread and >> possibly stopping it when it's supposed to be doing something else. >> >> Instead the instruction is queued to the channel and the channel does >> periodic calls to see if it has any tasks to execute when its tolerable to >> perform such a task and also ensuring its carried out in it's own dedicated >> session thread. >> >> From the time you send the command there is an expected latency until the >> command is actually executed. From your application you should send the >> command then wait until you receive the channel_execute corresponding to >> the desired command before continuing and having clearance to send the >> uuid_break. >> >> >> >> On Sun, Nov 20, 2011 at 2:07 AM, Hynek Cihlar < >> hynek.cihlar at gmail.com> wrote: >> >>> So why would the bgapi uuid_break all fail if it queues the same way as >>> execute does? >>> >>> Hynek >>> >>> >>> >>> >>> On Sun, Nov 20, 2011 at 4:27 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> Every way you are discussing does it the same way, doing it over bgapi >>>> just wastes an extra thread for no reason. >>>> >>>> All of the above queue the command to the session and it will not >>>> execute it until the next time its convenient for the session to execute >>>> queued instructions. >>>> >>>> So any way you slice it, unless you wait for the execute event telling >>>> you that the playback you tried to start, has actually started then its not >>>> going to help. >>>> >>>> You are most likely trying to had implement some convention that is >>>> dumbed down to one command already in FS so I suggest you re-evaluate what >>>> you are even trying to do. >>>> >>>> >>>> >>>> On Sat, Nov 19, 2011 at 3:37 AM, Hynek Cihlar < >>>> hynek.cihlar at gmail.com> wrote: >>>> >>>>> So I changed the uuid_X-commands to execute and the use case with >>>>> stopping tone playback works like a charm now! >>>>> >>>>> Regarding the blocking, you are right. I'm limiting blocking (and >>>>> context switching) by handling all events and issuing commands always on >>>>> one (and the same) thread per session. This design also allows me to >>>>> simplify the programming model, decrease concurrency bugs, simplify >>>>> debugging, etc. That's why the synthetic delay was giving me troubles since >>>>> the events and commands must fly through as fast as possible. >>>>> >>>>> Thanks again! >>>>> >>>>> Hynek >>>>> >>>>> >>>>> >>>>> >>>>> On Sat, Nov 19, 2011 at 10:28 AM, Peter Olsson < >>>>> peter.olsson at visionutveckling.se> wrote: >>>>> >>>>>> bgapi can of course be used also, and in some cases you might still >>>>>> need it. But the important thing to keep in mind when doing bgapi is to >>>>>> handle the events correctly. When executing bgapi you will get a job-id >>>>>> back, this job id must then be stored internally, so you can link this to >>>>>> the background job event. >>>>>> >>>>>> I try to use the sendMsg/execute as much as possible, you will get >>>>>> CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE events, so you will always know >>>>>> what's going on. >>>>>> >>>>>> I've only tested my system for about 100 concurrent calls, but that >>>>>> was handled without any problems at all. What's really important is to make >>>>>> really sure you never execute something that will block - since that will >>>>>> cause everything to "hang". >>>>>> >>>>>> /Peter >>>>>> ________________________________________ >>>>>> Fr?n: >>>>>> freeswitch-users-bounces at lists.freeswitch.org [ >>>>>> freeswitch-users-bounces at lists.freeswitch.org] för Hynek Cihlar >>>>>> [ >>>>>> hynek.cihlar at gmail.com] >>>>>> Skickat: den 19 november 2011 10:04 >>>>>> Till: FreeSWITCH Users Help >>>>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>>>> >>>>>> Peter, thanks for pointing me to the right direction. I will try the >>>>>> execute command, since queueing is exactly what I would need to preserve >>>>>> consistency across commands (I had an impression that bgapi will give me >>>>>> this consistency). >>>>>> >>>>>> Also I like to hear that it is possible to use a single connection >>>>>> since I have designed the communication with one connection also. May I ask >>>>>> how many concurrent sessions can your system handle? >>>>>> >>>>>> Hynek >>>>>> >>>>>> >>>>>> >>>>>> On Sat, Nov 19, 2011 at 9:35 AM, Peter Olsson < >>>>>> peter.olsson at visionutveckling.se >>>>>> peter.olsson at visionutveckling.se>> wrote: >>>>>> If you think about performance I would say it's better to handle this >>>>>> without spawning lots of extra threads. >>>>>> >>>>>> First of all, most common commands can be queued into the the channel >>>>>> thread using the execute method ( >>>>>> http://wiki.freeswitch.org/wiki/Mod_event_socket#execute), this >>>>>> makes it possible to queue commands directly to the channel's thread, and >>>>>> the ESL session will never wait for the result, it returns immediately. >>>>>> >>>>>> I've never used uuid_broadcast myself, so I'm not really sure how >>>>>> it's handled, but by looking quickly in the code it looks like it shouldn't >>>>>> block either, since it's actually just queuing a playback command to the >>>>>> channel's thread. >>>>>> >>>>>> So basically, keep away from uuid_X-commands, and use execute method >>>>>> as much as possible, and when using uuid_x-methods, make sure that won't >>>>>> block - but if they do, make sure to handle handle the events for the bgapi >>>>>> correctly. >>>>>> >>>>>> I've built a complete IVR system using these methods, with a single >>>>>> ESL connection, and I've never had any performace issues. >>>>>> >>>>>> /Peter >>>>>> >>>>>> ________________________________________ >>>>>> Fr?n: >>>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>>> freeswitch-users-bounces at lists.freeswitch.org> [ >>>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>>> freeswitch-users-bounces at lists.freeswitch.org>] för Hynek >>>>>> Cihlar [ >>>>>> hynek.cihlar at gmail.com >>>>>> hynek.cihlar at gmail.com>] >>>>>> Skickat: den 19 november 2011 09:12 >>>>>> Till: FreeSWITCH Users Help >>>>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>>>> >>>>>> Two reasons I am not using api in this case. >>>>>> >>>>>> 1. Performance >>>>>> 2. Doesn't seem to work >>>>>> >>>>>> 1. Blocking issuing the command will eventually reduce the system >>>>>> throughput. Agree this is only a potential theoretical problem and I >>>>>> haven't gotten into production where I could get the actual results. >>>>>> >>>>>> 2. When issuing api uuid_playback and api uuid_break right after, I'm >>>>>> always getting an error response from the uuid_break. The ESL message body >>>>>> is "-ERR no reply". Nothing in the logs (no idea which log type/level will >>>>>> could generate other useful info). >>>>>> >>>>>> Hynek >>>>>> >>>>>> >>>>>> >>>>>> On Sat, Nov 19, 2011 at 8:48 AM, Peter Olsson < >>>>>> peter.olsson at visionutveckling.se >>>>>> peter.olsson at visionutveckling.se> >>>>>> peter.olsson at visionutveckling.se >>>>>> peter.olsson at visionutveckling.se>>> wrote: >>>>>> Since you're using bgapi, it's impossible to guarantee that >>>>>> uuid_broadcast has actually executed before you exeute uuid_break (they >>>>>> will be executed in two different threads). You will have to do this in a >>>>>> more controlled way (wait for events to show up etc.). And also, do you >>>>>> really need to use bgapi all the time? I'm not sure how uuid_broadcast is >>>>>> handled, but uuid_break is totally safe to use without bgapi, since it will >>>>>> just update a flag and then return immediately. I think uuid_broadcast will >>>>>> do the same thing - so getting rid of bgapi should be a good ebnough >>>>>> solution. >>>>>> >>>>>> /Peter >>>>>> ________________________________________ >>>>>> Fr?n: >>>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>>> freeswitch-users-bounces at lists.freeswitch.org> >>>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>>> freeswitch-users-bounces at lists.freeswitch.org>> [ >>>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>>> freeswitch-users-bounces at lists.freeswitch.org> >>>>>> freeswitch-users-bounces at lists.freeswitch.org >>>>>> freeswitch-users-bounces at lists.freeswitch.org>>] för Hynek >>>>>> Cihlar [ >>>>>> hynek.cihlar at gmail.com >>>>>> hynek.cihlar at gmail.com> >>>>>> hynek.cihlar at gmail.com >>>>>> hynek.cihlar at gmail.com>>] >>>>>> Skickat: den 19 november 2011 08:31 >>>>>> Till: FreeSWITCH Users Help >>>>>> ?mne: Re: [Freeswitch-users] Playback race condition >>>>>> >>>>>> As a workaround I put a synthetic a delay between the start and stop >>>>>> of the playback. I wouldn't want to keep this solution however since it >>>>>> blocks one of the worker threads (decreasing overall system throughput) and >>>>>> second, I really don't know what the synthetic delay should be or whether >>>>>> it could differ during different conditions like during a high load. >>>>>> >>>>>> Was anybody facing similar problem? How did you solve it? Am I using >>>>>> the API wrong and is there another way to invoke playback/stop? >>>>>> >>>>>> Thanks! >>>>>> Hynek >>>>>> >>>>>> >>>>>> >>>>>> On Fri, Nov 18, 2011 at 6:46 PM, Hynek Cihlar < >>>>>> hynek.cihlar at gmail.com >>>>>> hynek.cihlar at gmail.com> >>>>>> hynek.cihlar at gmail.com >>>>>> hynek.cihlar at gmail.com>> >>>>>> hynek.cihlar at gmail.com >>>>>> hynek.cihlar at gmail.com> >>>>>> hynek.cihlar at gmail.com >>>>>> hynek.cihlar at gmail.com>>>> wrote: >>>>>> Hello all, >>>>>> >>>>>> I got on shaky grounds issuing commands to Freeswitch over ESL. >>>>>> >>>>>> For example, issuing the following commands close enough (on my >>>>>> system at around 100 ms and less) causes troubles. >>>>>> >>>>>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>>>>> playback::tone_stream://%(150,4000,425);loops=-1 >>>>>> <100ms appart >>>>>> bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all >>>>>> previous playback is not stopped, the following is queued (not played) >>>>>> bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 >>>>>> playback::tone_stream://%(1000,4000,425);loops=-1 >>>>>> >>>>>> It looks like the first tone playback is not properly initialized >>>>>> when the break arrives. Waiting for the start of the first tone playback >>>>>> (waiting for the right event) is not a solution, I don't want the tone to >>>>>> be played at all if the events come this close. >>>>>> >>>>>> Any ideas? >>>>>> >>>>>> >>>>>> Hynek >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> >>>>>> consulting at freeswitch.org >>>>>> consulting at freeswitch.org> >>>>>> consulting at freeswitch.org >>>>>> consulting at freeswitch.org>> >>>>>> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> <> <> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> http://wiki.freeswitch.org >>>>>> >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> FreeSWITCH-users at lists.freeswitch.org> >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> FreeSWITCH-users at lists.freeswitch.org>> >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> >>>>>> consulting at freeswitch.org >>>>>> consulting at freeswitch.org> >>>>>> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> <> <> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> http://wiki.freeswitch.org >>>>>> >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> FreeSWITCH-users at lists.freeswitch.org> >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> !DSPAM:4ec770ca32761488716642! >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> >>>>>> consulting at freeswitch.org >>>>>> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> <> <> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> http://wiki.freeswitch.org >>>>>> >>>>>> http://www.cluecon.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 >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> >>>>> consulting at freeswitch.org >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> <> <> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> >>>>> http://www.freeswitch.org >>>>> >>>>> http://wiki.freeswitch.org >>>>> >>>>> http://www.cluecon.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> >>>> 888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> <> <> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> >>>> http://www.freeswitch.org >>>> >>>> http://wiki.freeswitch.org >>>> >>>> http://www.cluecon.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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> <> <> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> http://www.cluecon.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> >> 888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> <> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> <> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/de7ac108/attachment-0001.html From msc at freeswitch.org Wed Nov 23 00:06:21 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Nov 2011 13:06:21 -0800 Subject: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on groupcall In-Reply-To: References: Message-ID: On Tue, Nov 22, 2011 at 2:52 AM, Erik Dekkers wrote: > Hi List,**** > > ** ** > > Following my recent message on this list ?P-Asserted-Identity: "Outbound > Call" on internal calls? I?ve managed to get the phones to display the > correct callee name.**** > > Unfortunately adding origination_callee_id_name to the groupcall doesn?t > work since I can?t predict which groupmember will pickup that call.**** > > ** ** > > Is there a way to make this work correctly?**** > > ** > mod_clairvoyance? :P Okay, being serious for a moment: could you restate the problem? It would be easier than asking 2000 people to go fishing through the list history. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/379228c2/attachment.html From royce3 at gmail.com Wed Nov 23 00:08:11 2011 From: royce3 at gmail.com (Royce Mitchell III) Date: Tue, 22 Nov 2011 15:08:11 -0600 Subject: [Freeswitch-users] removing participant from 3-way bridge Message-ID: Can I use "uuid_transfer park inline" to pull a leg out of a 3-way bridge without breaking the bridge? I want to put that leg back into the callcenter_mod queue once it's done connecting the callers. On Tue, Nov 22, 2011 at 1:22 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you can't use break to stop bridging. > you can do "uuid_transfer park inline" to put it into an idle state > > > > On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: > >> Hi, >> I wanted to cancel the bridge application after issuing it and I tried >> using 'break' application. break application didn't work. But, when I tried >> using playback application followed by break, it worked. >> >> connect >> >> sendmsg >> call-command: execute >> execute-app-name: answer >> >> sendmsg >> call-command: execute >> execute-app-name: bridge >> execute-app-arg: user/1001 >> >> sendmsg >> call-command: execute >> execute-app-name: break >> >> Logs: >> http://pastebin.freeswitch.org/17843 >> >> Git version as of 18/11/2011. >> >> -- >> Regards, >> Nagalenoj H. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- There's a fine line between genius and insanity. I like to use it for dental floss. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/57b03485/attachment.html From buscom123+fs at gmail.com Wed Nov 23 00:34:34 2011 From: buscom123+fs at gmail.com (R H) Date: Tue, 22 Nov 2011 14:34:34 -0700 Subject: [Freeswitch-users] Does Mod_Managed support .NET 4.0? If Not, can it be added? Message-ID: Hi, I'm running Mono 2.8.2 which, according to Mono's website, supports several of the API's introduced in 4.0. Unfortunately the moment I compile my libraries with a target runtime of Mono 4.0 Mod_Managed throws the following exception when attempting to load a library: *Missing method System.Threading.Monitor::Enter(object,bool&) in assembly /usr/lib/mono/2.0/mscorlib.dll, referenced in assembly /usr/local/freeswitch/mod/managed/PsiCallCenterAddon.d**ll* >From what I have read, This new method definition was added in 4.0 and is supported by mono 2.8.2 but for some reason using this version of mono with mod_managed is failing. The documentation at http://wiki.freeswitch.org/wiki/Mod_managed seems to explicitly state support for 3.5 but makes no mention of 4.0. So is there current or planned future support for .NET 4.0? What needs to be done to add support of everything mono 2.8.2 has to offer? Any help or information would be appreicated. Thanks! Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/894e01fa/attachment.html From mgg at giagnocavo.net Wed Nov 23 00:43:15 2011 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 22 Nov 2011 16:43:15 -0500 Subject: [Freeswitch-users] Does Mod_Managed support .NET 4.0? If Not, can it be added? In-Reply-To: References: Message-ID: <83FF8D7C9F526E44B77C97DD2891652A2691A0@mse17be1.mse17.exchange.ms> Try editing the Makefile to use dmcs instead of mcs. Although, that might require Mono 2.10. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of R H Sent: Tuesday, November 22, 2011 2:35 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Does Mod_Managed support .NET 4.0? If Not, can it be added? Hi, I'm running Mono 2.8.2 which, according to Mono's website, supports several of the API's introduced in 4.0. Unfortunately the moment I compile my libraries with a target runtime of Mono 4.0 Mod_Managed throws the following exception when attempting to load a library: Missing method System.Threading.Monitor::Enter(object,bool&) in assembly /usr/lib/mono/2.0/mscorlib.dll, referenced in assembly /usr/local/freeswitch/mod/managed/PsiCallCenterAddon.dll >From what I have read, This new method definition was added in 4.0 and is supported by mono 2.8.2 but for some reason using this version of mono with mod_managed is failing. The documentation at http://wiki.freeswitch.org/wiki/Mod_managed seems to explicitly state support for 3.5 but makes no mention of 4.0. So is there current or planned future support for .NET 4.0? What needs to be done to add support of everything mono 2.8.2 has to offer? Any help or information would be appreicated. Thanks! Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/ea1958a4/attachment.html From kerem.erciyes at gmail.com Wed Nov 23 00:44:23 2011 From: kerem.erciyes at gmail.com (Kerem Erciyes) Date: Tue, 22 Nov 2011 23:44:23 +0200 Subject: [Freeswitch-users] Joining a call from Gtalk to a conference on FreeSwitch In-Reply-To: <9438D04074E0DE45A49CD7609982127225B662F2@CORP-MAIL-002.edge.local> References: <9438D04074E0DE45A49CD7609982127225B662F2@CORP-MAIL-002.edge.local> Message-ID: Try turning of VAD and see if it will help. I am sure you should not be setting both proxy_media and bypass_media to true, as you should be transcoding if I remember correctly. http://wiki.freeswitch.org/wiki/Proxy_Media On Mon, Nov 21, 2011 at 12:28 PM, Papineni, Suneel wrote: > Hi,**** > > ** ** > > I got the new GIT version and enabled mod_dingling and compiled. > Everything went through and able to establish call to an extension if I > configure that extension number in ?client? profile.**** > > ** ** > > What I am trying to do is, I want to bridge or join a call coming from > GTalk to an existing conference in FreeSwitch. For this purpose I > configured a different number on ?client profile? and created a dial-plan > for this number to ?park? the call first before trying to join to the > conference.**** > > Then using eventSockets I am trying to join this call to conference and > issued following command. (tried with ?uuid_bridge? command as well)**** > > "api uuid_transfer [Unique-ID] conference:xyz at default inline"**** > > ** ** > > Command is successful and also I can hear a sound that someone joined in > the conference, but I didn?t hear any voice at either side. I couldn?t see > any RTP flow as well (checked wireshark traces at FS). After sometime like > 30 seconds call at GTalk is disconnected automatically.**** > > ** ** > > I am not sure why nothing is heard at both sides and why call got > disconnected. Also tried answering the call first (after Park) and then > bridging to conference, still got the same issue.**** > > ** ** > > Could someone please let me know if I am missing anything or need to > configure in a different way for conferencing.**** > > ** ** > > Thanks & Regards**** > > Suneel**** > > * * > > *Client.xml* > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > *Dial-plan..* > > **** > > **** > > break="never">**** > > data="effective_caller_id_number=$1" />**** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- Kerem Erciyes - Sistem Danismani http://keremerciyes.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/cd5e0b2a/attachment-0001.html From covici at ccs.covici.com Wed Nov 23 00:46:01 2011 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 22 Nov 2011 16:46:01 -0500 Subject: [Freeswitch-users] Does Mod_Managed support .NET 4.0? If Not, can it be added? In-Reply-To: References: Message-ID: <16021.1321998361@ccs.covici.com> I can't answer for the future, but 4.0 does not work as of now -- you have to ask Michael for more details. R H wrote: > Hi, > > I'm running Mono 2.8.2 which, according to Mono's website, supports several > of the API's introduced in 4.0. Unfortunately the moment I compile my > libraries with a target runtime of Mono 4.0 Mod_Managed throws the > following exception when attempting to load a library: > > *Missing method System.Threading.Monitor::Enter(object,bool&) in assembly > /usr/lib/mono/2.0/mscorlib.dll, referenced in assembly > /usr/local/freeswitch/mod/managed/PsiCallCenterAddon.d**ll* > > From what I have read, This new method definition was added in 4.0 and is > supported by mono 2.8.2 but for some reason using this version of mono with > mod_managed is failing. The documentation at > http://wiki.freeswitch.org/wiki/Mod_managed seems to explicitly state > support for 3.5 but makes no mention of 4.0. > > So is there current or planned future support for .NET 4.0? What needs to > be done to add support of everything mono 2.8.2 has to offer? > > Any help or information would be appreicated. Thanks! > > Ryan > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From covici at ccs.covici.com Wed Nov 23 01:00:17 2011 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 22 Nov 2011 17:00:17 -0500 Subject: [Freeswitch-users] Does Mod_Managed support .NET 4.0? If Not, can it be added? In-Reply-To: <83FF8D7C9F526E44B77C97DD2891652A2691A0@mse17be1.mse17.exchange.ms> References: <83FF8D7C9F526E44B77C97DD2891652A2691A0@mse17be1.mse17.exchange.ms> Message-ID: <17971.1321999217@ccs.covici.com> I tried that once and it did not work, but that was before you fixed things. I had a small test, compiled under windows which uses 4.0 by default, and tried to move the .dll over and run it with no success, but I don't remember the exact error at this time. Michael Giagnocavo wrote: > Try editing the Makefile to use dmcs instead of mcs. Although, that might require Mono 2.10. > > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of R H > Sent: Tuesday, November 22, 2011 2:35 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Does Mod_Managed support .NET 4.0? If Not, can it be added? > > Hi, > > I'm running Mono 2.8.2 which, according to Mono's website, supports several of the API's introduced in 4.0. Unfortunately the moment I compile my libraries with a target runtime of Mono 4.0 Mod_Managed throws the following exception when attempting to load a library: > > Missing method System.Threading.Monitor::Enter(object,bool&) in assembly /usr/lib/mono/2.0/mscorlib.dll, referenced in assembly /usr/local/freeswitch/mod/managed/PsiCallCenterAddon.dll > > From what I have read, This new method definition was added in 4.0 and is supported by mono 2.8.2 but for some reason using this version of mono with mod_managed is failing. The documentation at http://wiki.freeswitch.org/wiki/Mod_managed seems to explicitly state support for 3.5 but makes no mention of 4.0. > > So is there current or planned future support for .NET 4.0? What needs to be done to add support of everything mono 2.8.2 has to offer? > > Any help or information would be appreicated. Thanks! > > Ryan > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From mgg at giagnocavo.net Wed Nov 23 01:17:30 2011 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 22 Nov 2011 17:17:30 -0500 Subject: [Freeswitch-users] Does Mod_Managed support .NET 4.0? If Not, can it be added? In-Reply-To: <17971.1321999217@ccs.covici.com> References: <83FF8D7C9F526E44B77C97DD2891652A2691A0@mse17be1.mse17.exchange.ms> <17971.1321999217@ccs.covici.com> Message-ID: <83FF8D7C9F526E44B77C97DD2891652A2691AE@mse17be1.mse17.exchange.ms> I just tried a quick demo. Using gmcs it complained about the method not being found. I switched to dmcs and it worked without complaining. The last round of fixes must have sorted this out. But, I am also on Mono 2.10, and have no plans to support Mono 2.8, unless there's a very compelling reason to do so. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com Sent: Tuesday, November 22, 2011 3:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Does Mod_Managed support .NET 4.0? If Not, can it be added? I tried that once and it did not work, but that was before you fixed things. I had a small test, compiled under windows which uses 4.0 by default, and tried to move the .dll over and run it with no success, but I don't remember the exact error at this time. Michael Giagnocavo wrote: > Try editing the Makefile to use dmcs instead of mcs. Although, that might require Mono 2.10. > > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of R > H > Sent: Tuesday, November 22, 2011 2:35 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Does Mod_Managed support .NET 4.0? If Not, can it be added? > > Hi, > > I'm running Mono 2.8.2 which, according to Mono's website, supports several of the API's introduced in 4.0. Unfortunately the moment I compile my libraries with a target runtime of Mono 4.0 Mod_Managed throws the following exception when attempting to load a library: > > Missing method System.Threading.Monitor::Enter(object,bool&) in > assembly /usr/lib/mono/2.0/mscorlib.dll, referenced in assembly > /usr/local/freeswitch/mod/managed/PsiCallCenterAddon.dll > > From what I have read, This new method definition was added in 4.0 and is supported by mono 2.8.2 but for some reason using this version of mono with mod_managed is failing. The documentation at http://wiki.freeswitch.org/wiki/Mod_managed seems to explicitly state support for 3.5 but makes no mention of 4.0. > > So is there current or planned future support for .NET 4.0? What needs to be done to add support of everything mono 2.8.2 has to offer? > > Any help or information would be appreicated. Thanks! > > Ryan > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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-use > rs > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 admin at blindi.net Wed Nov 23 02:14:09 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Wed, 23 Nov 2011 00:14:09 +0100 (CET) Subject: [Freeswitch-users] Multiple calls fails In-Reply-To: <4EC9FB26.808@tagnet.ru> References: <4EC9DB1F.1000001@tagnet.ru> <20111121133853.3b5e17e8@shadow> <4EC9FB26.808@tagnet.ru> Message-ID: Hello, i make a paralellcall to: 1000 and 1001 For examle: one user 1000 or 1001 or so on is not registered. the transfer to the rest breaks. Is there a possibility that the chain calls anyway? thanks --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From anthony.minessale at gmail.com Wed Nov 23 02:26:13 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Nov 2011 17:26:13 -0600 Subject: [Freeswitch-users] removing participant from 3-way bridge In-Reply-To: References: Message-ID: 3-way via the phone? or FS conference or how? On Tue, Nov 22, 2011 at 3:08 PM, Royce Mitchell III wrote: > Can I use "uuid_transfer park inline" to pull a leg out of a 3-way > bridge without breaking the bridge? I want to put that leg back into the > callcenter_mod queue once it's done connecting the callers. > > On Tue, Nov 22, 2011 at 1:22 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> you can't use break to stop bridging. >> you can do "uuid_transfer park inline" to put it into an idle state >> >> >> >> On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: >> >>> Hi, >>> I wanted to cancel the bridge application after issuing it and I tried >>> using 'break' application. break application didn't work. But, when I tried >>> using playback application followed by break, it worked. >>> >>> connect >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: answer >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: bridge >>> execute-app-arg: user/1001 >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: break >>> >>> Logs: >>> http://pastebin.freeswitch.org/17843 >>> >>> Git version as of 18/11/2011. >>> >>> -- >>> Regards, >>> Nagalenoj H. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > -- > There's a fine line between genius and insanity. I like to use it for > dental floss. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/81be81e2/attachment-0001.html From msc at freeswitch.org Wed Nov 23 03:07:12 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Nov 2011 16:07:12 -0800 Subject: [Freeswitch-users] Multiple calls fails In-Reply-To: References: <4EC9DB1F.1000001@tagnet.ru> <20111121133853.3b5e17e8@shadow> <4EC9FB26.808@tagnet.ru> Message-ID: You can't use the user/xxx method when trying to bridge to more than one user without using the enterprise originate method. See here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#user Hint: use ":_:" instead of "," to separate the endpoints. -MC On Tue, Nov 22, 2011 at 3:14 PM, Thomas Hoellriegel wrote: > Hello, i make a paralellcall to: > 1000 and 1001 > data="user/1000@${domain_name}**, > user/1001@${domain_name},user/**1002@${domain_name}, > user/1003@${domain_name}"/> > > For examle: one user 1000 or 1001 or so on is not registered. the transfer > to the rest breaks. > > Is there a possibility that the chain calls anyway? > thanks > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111122/6e2f4c22/attachment.html From msc at freeswitch.org Wed Nov 23 03:08:57 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Nov 2011 16:08:57 -0800 Subject: [Freeswitch-users] Call to registered user In-Reply-To: <4ECB3CA5.8030705@tagnet.ru> References: <4EC9DB1F.1000001@tagnet.ru> <4ECB21A1.1090600@tagnet.ru> <4ECB3CA5.8030705@tagnet.ru> Message-ID: Is this a multi-domain setup? -MC On Mon, Nov 21, 2011 at 10:09 PM, Boris Kovalenko wrote: > Hello! > > freeswitch at internal> eval ${domain} > rcen-nt.ru > freeswitch at internal> expand sofia_contact 230172@${domain} > error/user_not_registered > > > Boris, > > Do "eval ${domain}" and see what shakes out - IP address or what? Try > this: > > expand sofia_contact 230172@${domain} > > -MC > > On Mon, Nov 21, 2011 at 8:14 PM, Boris Kovalenko wrote: > >> Hello! >> >> Michael, please explain: >> >> sofia status profile internal reg >> Registrations: >> >> ================================================================================================= >> Call-ID: 15020233664170-17093105930554 at 192.168.15.48 >> User: 230172 at rcen-nt.ru >> Contact: 230172 >> Agent: Greenlite ATOM V2.0 >> Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:19) >> EXPSECS(77) >> Host: voip-gw >> IP: 192.168.15.48 >> Port: 5060 >> Auth-User: 230172 >> Auth-Realm: rcen-nt.ru >> MWI-Account: 230172 at rcen-nt.ru >> >> Call-ID: 279492684728797-11326384312512 at 192.168.15.42 >> User: 230175 at rcen-nt.ru >> Contact: 230175 >> Agent: Greenlite ATOM V2.0 >> Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:28) >> EXPSECS(86) >> Host: voip-gw >> IP: 192.168.15.42 >> Port: 5060 >> Auth-User: 230175 >> Auth-Realm: rcen-nt.ru >> MWI-Account: 230175 at rcen-nt.ru >> >> Call-ID: 25095839831609-809787075531 at 192.168.15.46 >> User: 230171 at rcen-nt.ru >> Contact: 230171 >> Agent: Greenlite ATOM V2.0 >> Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:29) >> EXPSECS(87) >> Host: voip-gw >> IP: 192.168.15.46 >> Port: 5060 >> Auth-User: 230171 >> Auth-Realm: rcen-nt.ru >> MWI-Account: 230171 at rcen-nt.ru >> >> Call-ID: 1577286384367-21354577016286 at 192.168.15.45 >> User: 230173 at rcen-nt.ru >> Contact: 230173 >> Agent: Greenlite ATOM V2.0 >> Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:29) >> EXPSECS(87) >> Host: voip-gw >> IP: 192.168.15.45 >> Port: 5060 >> Auth-User: 230173 >> Auth-Realm: rcen-nt.ru >> MWI-Account: 230173 at rcen-nt.ru >> >> Call-ID: 59571353025698-224172903926056 at 192.168.15.43 >> User: 230174 at rcen-nt.ru >> Contact: 230174 >> Agent: Greenlite ATOM V2.0 >> Status: Registered(UDP)(unknown) EXP(2011-11-22 10:18:33) >> EXPSECS(91) >> Host: voip-gw >> IP: 192.168.15.43 >> Port: 5060 >> Auth-User: 230174 >> Auth-Realm: rcen-nt.ru >> MWI-Account: 230174 at rcen-nt.ru >> >> Total items returned: 5 >> >> ================================================================================================= >> >> freeswitch at internal> sofia_contact 230171 >> error/user_not_registered >> >> freeswitch at internal> sofia_contact 230171 at rcen-nt.ru >> error/user_not_registered >> >> freeswitch at internal> sofia_contact internal/230171 at rcent-nt.ru >> error/user_not_registered >> >> freeswitch at internal> sofia_contact internal/230171 >> sofia/internal/sip:230171 at 192.168.15.46:5060 >> >> freeswitch at internal> version >> FreeSWITCH Version 1.0.head (git-cdabd56 2011-11-17 08-23-21 +0100) >> >> >> >> >> On Sun, Nov 20, 2011 at 9:01 PM, Boris Kovalenko wrote: >> >>> Hello! >>> >>> My FS has different sip profiles (internet, local net, dmz etc). >>> There is some type of mobile users that may register with any profile. >>> So, is there a simple way to call registered user without known on which >>> profile it has registered? AFAIK sofia_contact need profile and user/ is >>> looking in profile of incoming call. >>> >> sofia_contact *can* accept a profile argument but it is not required. The >> required argument is "user at domain". If you don't have multiple domains >> then there should not be a reason that you cannot use sofia_contact. >> >> -MC >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> -- >> ? ?????????, >> ????? ????????? >> ??? "??????" >> ???. +7 (3435) 230001 >> ???? +7 (3435) 230005 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > Regards, > Boris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111122/1404697f/attachment-0001.html From philq at qsystemsengineering.com Wed Nov 23 07:04:55 2011 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Tue, 22 Nov 2011 23:04:55 -0500 Subject: [Freeswitch-users] When SSL/TLS enabled on FreeSwitch - sofia profile won't load Message-ID: <030601cca995$120cc5f0$362651d0$@com> I'm relatively new to FreeSwitch and am seriously impressed. That said, I'm having trouble getting TLS/SSL to work. When I enable it for a particular profile, that profile won't load. Running CentOS 5.7 - ModSSL, OpenSSL, and OpenSSL-devel are installed and the latest version. I did a bootstrap/configure/make after verifying that the needed SSL packages were installed but with the same result. Trying to restart the profile with TLS enabled from the FS CLI gives: Invalid Profile [internal]. I should mention that I'm running the FusionPBX front end to make it more convenient to perform the simpler admin tasks, just in case it matters. I did a make current late Sunday night. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f2cf68b 2011-11-20 18-40-41 -0500 Any suggestions as to what the problem might be or ways to go about tracking it down would be greatly appreciated. Thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111122/542d229d/attachment.html From erik.dekkers at certhon.com Wed Nov 23 10:45:45 2011 From: erik.dekkers at certhon.com (Erik Dekkers) Date: Wed, 23 Nov 2011 08:45:45 +0100 Subject: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on groupcall In-Reply-To: References: Message-ID: Hehe, I've asked a while ago why freeswitch is sending a P-Asserted-Identity: "Outbound Call" when a line is bridged. According to Anthony I should put origination_callee_id_name in the bridge. I've did that and it's working perfect when calling a person directly. However, when doing a groupcall this won't work cause you can't predict the person who picks up the call. Regards, Erik Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Michael Collins Verzonden: dinsdag 22 november 2011 22:06 Aan: FreeSWITCH Users Help Onderwerp: Re: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on groupcall On Tue, Nov 22, 2011 at 2:52 AM, Erik Dekkers > wrote: Hi List, Following my recent message on this list "P-Asserted-Identity: "Outbound Call" on internal calls" I've managed to get the phones to display the correct callee name. Unfortunately adding origination_callee_id_name to the groupcall doesn't work since I can't predict which groupmember will pickup that call. Is there a way to make this work correctly? mod_clairvoyance? :P Okay, being serious for a moment: could you restate the problem? It would be easier than asking 2000 people to go fishing through the list history. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/7eff2c05/attachment.html From royce3 at gmail.com Wed Nov 23 16:05:53 2011 From: royce3 at gmail.com (Royce Mitchell III) Date: Wed, 23 Nov 2011 07:05:53 -0600 Subject: [Freeswitch-users] removing participant from 3-way bridge In-Reply-To: References: Message-ID: I'm using uuid_bridge via ESL to create a 3-way, so it's a 3-way inside FS. On Tue, Nov 22, 2011 at 5:26 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > 3-way via the phone? or FS conference or how? > > > On Tue, Nov 22, 2011 at 3:08 PM, Royce Mitchell III wrote: > >> Can I use "uuid_transfer park inline" to pull a leg out of a 3-way >> bridge without breaking the bridge? I want to put that leg back into the >> callcenter_mod queue once it's done connecting the callers. >> >> On Tue, Nov 22, 2011 at 1:22 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> you can't use break to stop bridging. >>> you can do "uuid_transfer park inline" to put it into an idle >>> state >>> >>> >>> >>> On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: >>> >>>> Hi, >>>> I wanted to cancel the bridge application after issuing it and I >>>> tried using 'break' application. break application didn't work. But, when I >>>> tried using playback application followed by break, it worked. >>>> >>>> connect >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: answer >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: bridge >>>> execute-app-arg: user/1001 >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: break >>>> >>>> Logs: >>>> http://pastebin.freeswitch.org/17843 >>>> >>>> Git version as of 18/11/2011. >>>> >>>> -- >>>> Regards, >>>> Nagalenoj H. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> >> >> >> -- >> There's a fine line between genius and insanity. I like to use it for >> dental floss. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- There's a fine line between genius and insanity. I like to use it for dental floss. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/d817f90c/attachment-0001.html From davidwaf at gmail.com Wed Nov 23 16:43:35 2011 From: davidwaf at gmail.com (David Wafula) Date: Wed, 23 Nov 2011 15:43:35 +0200 Subject: [Freeswitch-users] Kernel timer ...those with experience please ? In-Reply-To: References: Message-ID: On Mon, Nov 21, 2011 at 10:35 PM, David Wafula wrote: > Hi all, > I have very recent git (20th Nov 2011) on linode instance on Ubuntu 10.04, > kernel 2.6.39. When alone in a conference, the music plays very neatly and > clear. When another person joins the conference, then everything becomes > distorted, cant even hold a simple conversion. Reading around, everyone > attributes this to kernel timer. Could the kind experienced souls > share their knowledge on kernel timer, how it affects freeswitch, and how > best to correct it please. BTW, am considering during new image build with > the latest kernel, will it help ? > After more reading around, i did what everyone was recommending. Switched of CentOS, the default timer is already 1000 i discovered. Am using CentOS 6 64-bit, from linode.com. No more choppy sound. I still have static, but am sure upping sampling rates should fix some of that. -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/4b20e4af/attachment.html From dujinfang at gmail.com Wed Nov 23 18:20:24 2011 From: dujinfang at gmail.com (Seven Du) Date: Wed, 23 Nov 2011 23:20:24 +0800 Subject: [Freeswitch-users] Wednesday Conf Call - SIP Presence? In-Reply-To: References: Message-ID: <96DA38C129524076A353A8F245668624@gmail.com> I'm happened playing on presence related recently while not enough for a presentation but can share some experiences and learn from others. Will be in the conference. Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) On Tuesday, November 22, 2011 at 11:03 AM, Michael Collins wrote: > Hi all, > > I would like to have someone speak on the subject of SIP presence this coming Wednesday. If you are up for giving the community a brief introduction to the concepts involved with SIP presence please let me know. Also, if you are experienced and/or knowledgeable on the subject but would rather not give a presentation also let me know as you may be in a position to type up some material that could be used by the presenter. > > Please email me off list. > > Thanks all! > > -MC > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/c106fc94/attachment.html From garcia16 at gmail.com Wed Nov 23 17:23:06 2011 From: garcia16 at gmail.com (garcia16 at gmail.com) Date: Wed, 23 Nov 2011 15:23:06 +0100 Subject: [Freeswitch-users] FreeSwitch and BlueBox - cannot connect and no SIP Registration Message-ID: Hello, I have recently installed FreeSwitch with Blue.Box and I have two issues with them : 1. Module SIP Registration does not work, i mean when i click on him, nothing shows up. 2. When i run "sofia status" command on Freeswitch : Name Type Data State ================================================================================================= sipinterface_1 profile sip:mod_sofia at 10.0.2.15:5090 RUNNING (0) sipinterface_3 profile sip:mod_sofia at 10.0.2.15:5080 RUNNING (0) sipinterface_2 profile sip:mod_sofia at 10.0.2.15:5070 RUNNING (0) ================================================================================================= My question is : how connect to my Freeswitch. I would also like to add that i am not able to add any gateway. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/7fc86af1/attachment.html From bellesoft at gmail.com Wed Nov 23 18:29:10 2011 From: bellesoft at gmail.com (bellesoft) Date: Wed, 23 Nov 2011 07:29:10 -0800 (PST) Subject: [Freeswitch-users] Conference and Speex Wideband audio quality problem In-Reply-To: References: Message-ID: <1322062150514-7024739.post@n2.nabble.com> Hi Richard, I know it has been a while since you posted this, but I encounter the exact same issue you descrive (robotic , bad quality audio from the Mic) when connecting to mod_conference/wideband. I am using the same parameters you posted bellow. Did you actually get it to work OK with all flash clients? In my case it works great with PC's, but on my Mac and Ipad the quality connecting to the wide-band conference is VERY bad. Same machine, using a sip client works great. so its not the mic/network. Any help appreciated. -Bellesoft Richard Alam wrote > > OK...figured out what the problem. > > Flash Player by default sends 2 frames per packet with 20ms of audio > (320 samples) per frame. FS expects only a frame every packet. > So setting mic.framesPerPacket = 1; in the flash client worked. > > Here's the complete Actionscript setting in case somebody in the > future wants to work with Flash. > > private function setupMicrophone():void { > mic.setUseEchoSuppression(true); > mic.setLoopBack(false); > mic.setSilenceLevel(0,20000); > mic.codec = SoundCodec.SPEEX; > mic.gain = 60; > mic.framesPerPacket = 1; > mic.rate = 16; // use 8 for Nelly > > LogUtil.debug("codec=SPEEX,gain=60,encodeQuality=10,framesPerPacket=2,rate=16"); > } > > Richard > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Conference-and-Speex-Wideband-audio-quality-problem-tp5356620p7024739.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Wed Nov 23 19:35:29 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Nov 2011 08:35:29 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_23 There have been a number of changes/additions to mod_conference, so we are going to talk about those today. Also, there are several FreeSWITCH updates to talk about. If we have time and if enough subject matter experts are available we will also talk about SIP presence. See you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/4e7db7e0/attachment.html From msc at freeswitch.org Wed Nov 23 19:51:29 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Nov 2011 08:51:29 -0800 Subject: [Freeswitch-users] FreeSwitch and BlueBox - cannot connect and no SIP Registration In-Reply-To: References: Message-ID: You might need to have the 2600hz guys check out this update: http://jira.freeswitch.org/browse/FS-3699 Also, try "sofia status profile sipinterface_1" to get more information about a specific sofia profile. Note: some of these questions are specific to bluebox and not just general FreeSWITCH questions so you might want to ask on their mailing list or IRC channel. -MC On Wed, Nov 23, 2011 at 6:23 AM, wrote: > Hello, > > I have recently installed FreeSwitch with Blue.Box and I have two issues > with them : > > 1. Module SIP Registration does not work, i mean when i click on him, > nothing shows up. > > 2. When i run "sofia status" command on Freeswitch : > > Name Type > Data State > > ================================================================================================= > sipinterface_1 profile > sip:mod_sofia at 10.0.2.15:5090 RUNNING (0) > sipinterface_3 profile > sip:mod_sofia at 10.0.2.15:5080 RUNNING (0) > sipinterface_2 profile > sip:mod_sofia at 10.0.2.15:5070 RUNNING (0) > > ================================================================================================= > > My question is : how connect to my Freeswitch. I would also like to add > that i am not able to add any gateway. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111123/a61604c9/attachment-0001.html From dujinfang at gmail.com Wed Nov 23 20:21:34 2011 From: dujinfang at gmail.com (Seven Du) Date: Thu, 24 Nov 2011 01:21:34 +0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: sorry I calculated time wrong so it will be too late for me and I'll be absent. I created a simple presence page and feel free to update it. http://wiki.freeswitch.org/wiki/Presence Thanks. On Thursday, November 24, 2011 at 12:35 AM, Michael Collins wrote: > Hello all! > > Today's agenda is here: > > http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_23 > > There have been a number of changes/additions to mod_conference, so we are going to talk about those today. Also, there are several FreeSWITCH updates to talk about. If we have time and if enough subject matter experts are available we will also talk about SIP presence. > > See you soon! > > -Michael > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111124/a6ca2966/attachment.html From msc at freeswitch.org Wed Nov 23 20:58:34 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Nov 2011 09:58:34 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: Thanks! -MC On Wed, Nov 23, 2011 at 9:21 AM, Seven Du wrote: > sorry I calculated time wrong so it will be too late for me and I'll be > absent. > > I created a simple presence page and feel free to update it. > > http://wiki.freeswitch.org/wiki/Presence > > Thanks. > > On Thursday, November 24, 2011 at 12:35 AM, Michael Collins wrote: > > Hello all! > > Today's agenda is here: > > http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_23 > > There have been a number of changes/additions to mod_conference, so we are > going to talk about those today. Also, there are several FreeSWITCH updates > to talk about. If we have time and if enough subject matter experts are > available we will also talk about SIP presence. > > See you soon! > > -Michael > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111123/b0a1f845/attachment.html From jerry.richards at teotech.com Wed Nov 23 22:08:39 2011 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 23 Nov 2011 11:08:39 -0800 Subject: [Freeswitch-users] In Bypass Mode Freeswitch Changes SDP? Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D69633169C807C@VA3DIAXVS351.RED001.local> Hello, I noticed an interop issue when using a Bria softphone and inbound-bypass-media is true in the sip_profile. When answering an inbound call, the Bria softphone's SDP Answer includes the list of codecs it supports (audio and video). I noticed that, even in bypass media mode, Freeswitch filters out all of the codecs in the list after the first one. I think this is not right. According to RFC 3264 (Section 7 Offerer Processing of the Answer), it is valid to provide a list of codecs in an SDP Answer and the endpoint SHOULD use the first one (but it might not). Why is Freeswitch altering the SDP in the 200 OK? I think it should send the SDP unmodified. Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/84d522d8/attachment.html From anthony.minessale at gmail.com Wed Nov 23 22:21:34 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Nov 2011 13:21:34 -0600 Subject: [Freeswitch-users] In Bypass Mode Freeswitch Changes SDP? In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D69633169C807C@VA3DIAXVS351.RED001.local> References: <2BF7FB90DF25EA4485949F3AF2B9D69633169C807C@VA3DIAXVS351.RED001.local> Message-ID: Do you have a trace of this taking place on latest GIT? I tested this and it appears to not touch the codecs.... On Wed, Nov 23, 2011 at 1:08 PM, Jerry Richards wrote: > Hello,**** > > ** ** > > I noticed an interop issue when using a Bria softphone and inbound-bypass-media > is true in the sip_profile. When answering an inbound call, the Bria > softphone's SDP Answer includes the list of codecs it supports (audio and > video). I noticed that, even in bypass media mode, Freeswitch filters out > all of the codecs in the list after the first one. I think this is not > right. According to RFC 3264 (Section 7 Offerer Processing of the Answer), > it is valid to provide a list of codecs in an SDP Answer and the endpoint > SHOULD use the first one (but it might not).**** > > ** ** > > Why is Freeswitch altering the SDP in the 200 OK? I think it should send > the SDP unmodified.**** > > ** ** > > Thanks,**** > > Jerry**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/482cc7bd/attachment-0001.html From anthony.minessale at gmail.com Wed Nov 23 22:28:21 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Nov 2011 13:28:21 -0600 Subject: [Freeswitch-users] removing participant from 3-way bridge In-Reply-To: References: Message-ID: I don't understand how you could make a 3 way call with uuid_bridge ? On Wed, Nov 23, 2011 at 7:05 AM, Royce Mitchell III wrote: > I'm using uuid_bridge via ESL to create a 3-way, so it's a 3-way inside FS. > > On Tue, Nov 22, 2011 at 5:26 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> 3-way via the phone? or FS conference or how? >> >> >> On Tue, Nov 22, 2011 at 3:08 PM, Royce Mitchell III wrote: >> >>> Can I use "uuid_transfer park inline" to pull a leg out of a >>> 3-way bridge without breaking the bridge? I want to put that leg back into >>> the callcenter_mod queue once it's done connecting the callers. >>> >>> On Tue, Nov 22, 2011 at 1:22 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> you can't use break to stop bridging. >>>> you can do "uuid_transfer park inline" to put it into an idle >>>> state >>>> >>>> >>>> >>>> On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: >>>> >>>>> Hi, >>>>> I wanted to cancel the bridge application after issuing it and I >>>>> tried using 'break' application. break application didn't work. But, when I >>>>> tried using playback application followed by break, it worked. >>>>> >>>>> connect >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: answer >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: bridge >>>>> execute-app-arg: user/1001 >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: break >>>>> >>>>> Logs: >>>>> http://pastebin.freeswitch.org/17843 >>>>> >>>>> Git version as of 18/11/2011. >>>>> >>>>> -- >>>>> Regards, >>>>> Nagalenoj H. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>>> >>>> >>> >>> >>> -- >>> There's a fine line between genius and insanity. I like to use it for >>> dental floss. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > -- > There's a fine line between genius and insanity. I like to use it for > dental floss. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/7218dca1/attachment.html From msc at freeswitch.org Wed Nov 23 22:33:35 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Nov 2011 11:33:35 -0800 Subject: [Freeswitch-users] removing participant from 3-way bridge In-Reply-To: References: Message-ID: The bridge app only bridges two endpoints. How is the 3rd caller involved? Best thing to do would be to pastebin the console debug log of the 3-way call getting set up. It would also help to see the commands you are sending to create this 3-way. -MC On Wed, Nov 23, 2011 at 5:05 AM, Royce Mitchell III wrote: > I'm using uuid_bridge via ESL to create a 3-way, so it's a 3-way inside FS. > > On Tue, Nov 22, 2011 at 5:26 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> 3-way via the phone? or FS conference or how? >> >> >> On Tue, Nov 22, 2011 at 3:08 PM, Royce Mitchell III wrote: >> >>> Can I use "uuid_transfer park inline" to pull a leg out of a >>> 3-way bridge without breaking the bridge? I want to put that leg back into >>> the callcenter_mod queue once it's done connecting the callers. >>> >>> On Tue, Nov 22, 2011 at 1:22 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> you can't use break to stop bridging. >>>> you can do "uuid_transfer park inline" to put it into an idle >>>> state >>>> >>>> >>>> >>>> On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: >>>> >>>>> Hi, >>>>> I wanted to cancel the bridge application after issuing it and I >>>>> tried using 'break' application. break application didn't work. But, when I >>>>> tried using playback application followed by break, it worked. >>>>> >>>>> connect >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: answer >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: bridge >>>>> execute-app-arg: user/1001 >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: break >>>>> >>>>> Logs: >>>>> http://pastebin.freeswitch.org/17843 >>>>> >>>>> Git version as of 18/11/2011. >>>>> >>>>> -- >>>>> Regards, >>>>> Nagalenoj H. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>>> >>>> >>> >>> >>> -- >>> There's a fine line between genius and insanity. I like to use it for >>> dental floss. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > -- > There's a fine line between genius and insanity. I like to use it for > dental floss. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111123/5815161c/attachment-0001.html From royce3 at gmail.com Wed Nov 23 22:41:37 2011 From: royce3 at gmail.com (Royce Mitchell III) Date: Wed, 23 Nov 2011 13:41:37 -0600 Subject: [Freeswitch-users] removing participant from 3-way bridge In-Reply-To: References: Message-ID: I'm not sure if I've actually tried this, yet, but when you bridge 2 legs, you get a new uuid that describes the bridge, and I would then bridge that with the 3rd leg. Is there a better way to do this? I need to pull two calls together into a conference with the agent, then after the agent makes the introductions, pull the agent out of that conference and back into the callcenter queue. On Wed, Nov 23, 2011 at 1:28 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I don't understand how you could make a 3 way call with uuid_bridge ? > > > On Wed, Nov 23, 2011 at 7:05 AM, Royce Mitchell III wrote: > >> I'm using uuid_bridge via ESL to create a 3-way, so it's a 3-way inside >> FS. >> >> On Tue, Nov 22, 2011 at 5:26 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> 3-way via the phone? or FS conference or how? >>> >>> >>> On Tue, Nov 22, 2011 at 3:08 PM, Royce Mitchell III wrote: >>> >>>> Can I use "uuid_transfer park inline" to pull a leg out of a >>>> 3-way bridge without breaking the bridge? I want to put that leg back into >>>> the callcenter_mod queue once it's done connecting the callers. >>>> >>>> On Tue, Nov 22, 2011 at 1:22 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> you can't use break to stop bridging. >>>>> you can do "uuid_transfer park inline" to put it into an idle >>>>> state >>>>> >>>>> >>>>> >>>>> On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: >>>>> >>>>>> Hi, >>>>>> I wanted to cancel the bridge application after issuing it and I >>>>>> tried using 'break' application. break application didn't work. But, when I >>>>>> tried using playback application followed by break, it worked. >>>>>> >>>>>> connect >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: answer >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: bridge >>>>>> execute-app-arg: user/1001 >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: break >>>>>> >>>>>> Logs: >>>>>> http://pastebin.freeswitch.org/17843 >>>>>> >>>>>> Git version as of 18/11/2011. >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> Nagalenoj H. >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.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 >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:+19193869900 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> There's a fine line between genius and insanity. I like to use it for >>>> dental floss. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> >> >> >> -- >> There's a fine line between genius and insanity. I like to use it for >> dental floss. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- There's a fine line between genius and insanity. I like to use it for dental floss. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/01cba1c7/attachment.html From anthony.minessale at gmail.com Wed Nov 23 22:47:07 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Nov 2011 13:47:07 -0600 Subject: [Freeswitch-users] removing participant from 3-way bridge In-Reply-To: References: Message-ID: no this is not possible but what you can do is use uuid_transfer with the -both arguement to create a conference. say and are bridged normally: uuid_transfer -both conference:@default inline then you can add as many more as you want with uuid_transfer conference:@default inline On Wed, Nov 23, 2011 at 1:41 PM, Royce Mitchell III wrote: > I'm not sure if I've actually tried this, yet, but when you bridge 2 legs, > you get a new uuid that describes the bridge, and I would then bridge that > with the 3rd leg. > > Is there a better way to do this? I need to pull two calls together into a > conference with the agent, then after the agent makes the introductions, > pull the agent out of that conference and back into the callcenter queue. > > On Wed, Nov 23, 2011 at 1:28 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> I don't understand how you could make a 3 way call with uuid_bridge ? >> >> >> On Wed, Nov 23, 2011 at 7:05 AM, Royce Mitchell III wrote: >> >>> I'm using uuid_bridge via ESL to create a 3-way, so it's a 3-way inside >>> FS. >>> >>> On Tue, Nov 22, 2011 at 5:26 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> 3-way via the phone? or FS conference or how? >>>> >>>> >>>> On Tue, Nov 22, 2011 at 3:08 PM, Royce Mitchell III wrote: >>>> >>>>> Can I use "uuid_transfer park inline" to pull a leg out of a >>>>> 3-way bridge without breaking the bridge? I want to put that leg back into >>>>> the callcenter_mod queue once it's done connecting the callers. >>>>> >>>>> On Tue, Nov 22, 2011 at 1:22 PM, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>>> you can't use break to stop bridging. >>>>>> you can do "uuid_transfer park inline" to put it into an idle >>>>>> state >>>>>> >>>>>> >>>>>> >>>>>> On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: >>>>>> >>>>>>> Hi, >>>>>>> I wanted to cancel the bridge application after issuing it and I >>>>>>> tried using 'break' application. break application didn't work. But, when I >>>>>>> tried using playback application followed by break, it worked. >>>>>>> >>>>>>> connect >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name: answer >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name: bridge >>>>>>> execute-app-arg: user/1001 >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name: break >>>>>>> >>>>>>> Logs: >>>>>>> http://pastebin.freeswitch.org/17843 >>>>>>> >>>>>>> Git version as of 18/11/2011. >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> Nagalenoj H. >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.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 >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> pstn:+19193869900 >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.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 >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> There's a fine line between genius and insanity. I like to use it for >>>>> dental floss. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>>> >>>> >>> >>> >>> -- >>> There's a fine line between genius and insanity. I like to use it for >>> dental floss. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > -- > There's a fine line between genius and insanity. I like to use it for > dental floss. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/c5c6a9d9/attachment-0001.html From jeff at jefflenk.com Wed Nov 23 22:49:24 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 23 Nov 2011 11:49:24 -0800 (PST) Subject: [Freeswitch-users] new git - no more 'SIP auth challenge' messages in log In-Reply-To: References: <1320945234630-6982405.post@n2.nabble.com> <18097.1320948567@ccs.covici.com> Message-ID: <1322077764074-7025855.post@n2.nabble.com> This has been changed to log warning messages again with the flag set by default to disabled in the git conf folder. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/new-git-no-more-SIP-auth-challenge-messages-in-log-tp6980582p7025855.html Sent from the freeswitch-users mailing list archive at Nabble.com. From royce3 at gmail.com Wed Nov 23 23:00:44 2011 From: royce3 at gmail.com (Royce Mitchell III) Date: Wed, 23 Nov 2011 14:00:44 -0600 Subject: [Freeswitch-users] removing participant from 3-way bridge In-Reply-To: References: Message-ID: cool, that's enough. how would I "hang up" uuid1 out of the conference in a way that it's transfer_after_bridge setting can take effect? Will "SendMsg uuid1\ncall-command: hangup" do the job, or does that bypass transfer_after_bridge? I will try this when I get a chance later, but I wanted to ask in case that won't work. On Wed, Nov 23, 2011 at 1:47 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > no this is not possible > but what you can do is use uuid_transfer with the -both arguement to > create a conference. > > say and are bridged normally: > > uuid_transfer -both conference:@default inline > > then you can add as many more as you want with > > uuid_transfer conference:@default inline > > > On Wed, Nov 23, 2011 at 1:41 PM, Royce Mitchell III wrote: > >> I'm not sure if I've actually tried this, yet, but when you bridge 2 >> legs, you get a new uuid that describes the bridge, and I would then bridge >> that with the 3rd leg. >> >> Is there a better way to do this? I need to pull two calls together into >> a conference with the agent, then after the agent makes the introductions, >> pull the agent out of that conference and back into the callcenter queue. >> >> On Wed, Nov 23, 2011 at 1:28 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> I don't understand how you could make a 3 way call with uuid_bridge ? >>> >>> >>> On Wed, Nov 23, 2011 at 7:05 AM, Royce Mitchell III wrote: >>> >>>> I'm using uuid_bridge via ESL to create a 3-way, so it's a 3-way inside >>>> FS. >>>> >>>> On Tue, Nov 22, 2011 at 5:26 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> 3-way via the phone? or FS conference or how? >>>>> >>>>> >>>>> On Tue, Nov 22, 2011 at 3:08 PM, Royce Mitchell III wrote: >>>>> >>>>>> Can I use "uuid_transfer park inline" to pull a leg out of a >>>>>> 3-way bridge without breaking the bridge? I want to put that leg back into >>>>>> the callcenter_mod queue once it's done connecting the callers. >>>>>> >>>>>> On Tue, Nov 22, 2011 at 1:22 PM, Anthony Minessale < >>>>>> anthony.minessale at gmail.com> wrote: >>>>>> >>>>>>> you can't use break to stop bridging. >>>>>>> you can do "uuid_transfer park inline" to put it into an idle >>>>>>> state >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: >>>>>>> >>>>>>>> Hi, >>>>>>>> I wanted to cancel the bridge application after issuing it and I >>>>>>>> tried using 'break' application. break application didn't work. But, when I >>>>>>>> tried using playback application followed by break, it worked. >>>>>>>> >>>>>>>> connect >>>>>>>> >>>>>>>> sendmsg >>>>>>>> call-command: execute >>>>>>>> execute-app-name: answer >>>>>>>> >>>>>>>> sendmsg >>>>>>>> call-command: execute >>>>>>>> execute-app-name: bridge >>>>>>>> execute-app-arg: user/1001 >>>>>>>> >>>>>>>> sendmsg >>>>>>>> call-command: execute >>>>>>>> execute-app-name: break >>>>>>>> >>>>>>>> Logs: >>>>>>>> http://pastebin.freeswitch.org/17843 >>>>>>>> >>>>>>>> Git version as of 18/11/2011. >>>>>>>> >>>>>>>> -- >>>>>>>> Regards, >>>>>>>> Nagalenoj H. >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.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 >>>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>>> pstn:+19193869900 >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> There's a fine line between genius and insanity. I like to use it for >>>>>> dental floss. >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.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 >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:+19193869900 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> There's a fine line between genius and insanity. I like to use it for >>>> dental floss. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> >> >> >> -- >> There's a fine line between genius and insanity. I like to use it for >> dental floss. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- There's a fine line between genius and insanity. I like to use it for dental floss. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/91e093b3/attachment-0001.html From anthony.minessale at gmail.com Wed Nov 23 23:07:39 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Nov 2011 14:07:39 -0600 Subject: [Freeswitch-users] removing participant from 3-way bridge In-Reply-To: References: Message-ID: uuid_kill On Wed, Nov 23, 2011 at 2:00 PM, Royce Mitchell III wrote: > cool, that's enough. how would I "hang up" uuid1 out of the conference in > a way that it's transfer_after_bridge setting can take effect? Will > "SendMsg uuid1\ncall-command: hangup" do the job, or does that bypass > transfer_after_bridge? I will try this when I get a chance later, but I > wanted to ask in case that won't work. > > On Wed, Nov 23, 2011 at 1:47 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> no this is not possible >> but what you can do is use uuid_transfer with the -both arguement to >> create a conference. >> >> say and are bridged normally: >> >> uuid_transfer -both conference:@default inline >> >> then you can add as many more as you want with >> >> uuid_transfer conference:@default inline >> >> >> On Wed, Nov 23, 2011 at 1:41 PM, Royce Mitchell III wrote: >> >>> I'm not sure if I've actually tried this, yet, but when you bridge 2 >>> legs, you get a new uuid that describes the bridge, and I would then bridge >>> that with the 3rd leg. >>> >>> Is there a better way to do this? I need to pull two calls together into >>> a conference with the agent, then after the agent makes the introductions, >>> pull the agent out of that conference and back into the callcenter queue. >>> >>> On Wed, Nov 23, 2011 at 1:28 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> I don't understand how you could make a 3 way call with uuid_bridge ? >>>> >>>> >>>> On Wed, Nov 23, 2011 at 7:05 AM, Royce Mitchell III wrote: >>>> >>>>> I'm using uuid_bridge via ESL to create a 3-way, so it's a 3-way >>>>> inside FS. >>>>> >>>>> On Tue, Nov 22, 2011 at 5:26 PM, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>>> 3-way via the phone? or FS conference or how? >>>>>> >>>>>> >>>>>> On Tue, Nov 22, 2011 at 3:08 PM, Royce Mitchell III >>>>> > wrote: >>>>>> >>>>>>> Can I use "uuid_transfer park inline" to pull a leg out of a >>>>>>> 3-way bridge without breaking the bridge? I want to put that leg back into >>>>>>> the callcenter_mod queue once it's done connecting the callers. >>>>>>> >>>>>>> On Tue, Nov 22, 2011 at 1:22 PM, Anthony Minessale < >>>>>>> anthony.minessale at gmail.com> wrote: >>>>>>> >>>>>>>> you can't use break to stop bridging. >>>>>>>> you can do "uuid_transfer park inline" to put it into an >>>>>>>> idle state >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: >>>>>>>> >>>>>>>>> Hi, >>>>>>>>> I wanted to cancel the bridge application after issuing it and I >>>>>>>>> tried using 'break' application. break application didn't work. But, when I >>>>>>>>> tried using playback application followed by break, it worked. >>>>>>>>> >>>>>>>>> connect >>>>>>>>> >>>>>>>>> sendmsg >>>>>>>>> call-command: execute >>>>>>>>> execute-app-name: answer >>>>>>>>> >>>>>>>>> sendmsg >>>>>>>>> call-command: execute >>>>>>>>> execute-app-name: bridge >>>>>>>>> execute-app-arg: user/1001 >>>>>>>>> >>>>>>>>> sendmsg >>>>>>>>> call-command: execute >>>>>>>>> execute-app-name: break >>>>>>>>> >>>>>>>>> Logs: >>>>>>>>> http://pastebin.freeswitch.org/17843 >>>>>>>>> >>>>>>>>> Git version as of 18/11/2011. >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Regards, >>>>>>>>> Nagalenoj H. >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.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 >>>>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>>>> pstn:+19193869900 >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> There's a fine line between genius and insanity. I like to use it >>>>>>> for dental floss. >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.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 >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> pstn:+19193869900 >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.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 >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> There's a fine line between genius and insanity. I like to use it for >>>>> dental floss. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.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 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.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 >>>> >>>> >>> >>> >>> -- >>> There's a fine line between genius and insanity. I like to use it for >>> dental floss. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > -- > There's a fine line between genius and insanity. I like to use it for > dental floss. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/ee2d64f8/attachment-0001.html From freeswitch at peely.com Wed Nov 23 23:52:29 2011 From: freeswitch at peely.com (peely) Date: Wed, 23 Nov 2011 12:52:29 -0800 (PST) Subject: [Freeswitch-users] mod_rtmp: "Read error" when attempting calls from FS to RTMP client. Message-ID: <1322081549623-7026117.post@n2.nabble.com> Hi, I have a strange issue when I attempt to bridge a call from FS out to an RTMP client. As soon as the call is offered up to the client, the call is disconnected immediately i.e at the "ringing" stage. The strange thing is that as soon as I issue an "fsctl loglevel 9" then everything is fine. I don't know whether this is because of any additional load, as nothing above "ERR" is printed in each case. When the call is cleared immediately then I am getting "[ERR] rtmp.c:713 Read error" in the logs. mod_rtmp is compiled from today's git. Is there a possibility something is trying to happen too quickly in the handshake before data has been received from the client? Outbound calls from client to FS work flawlessly and incoming calls too, so long as I seem to have debug logging enabled. I'm not sure if it's relevant, but I'm bridging to the UUID of the session AKA "" Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-rtmp-Read-error-when-attempting-calls-from-FS-to-RTMP-client-tp7026117p7026117.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at peely.com Thu Nov 24 00:23:06 2011 From: freeswitch at peely.com (peely) Date: Wed, 23 Nov 2011 13:23:06 -0800 (PST) Subject: [Freeswitch-users] High load on database server In-Reply-To: References: Message-ID: <1322083386335-7026203.post@n2.nabble.com> I had exactly the same problem with PGSQL and FS, moved to MySQL for the core DB and everything is fine. PGSQL's transaction logging is pretty heavyweight for the amount of transactions you get from FS, but if you are really needing to use PostGres I would look at using unlogged tables. As I say though, MySQL runs like a charm, I've had it at well over 300CPS with 500 registrations/sec. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Re-High-load-on-database-server-tp7008003p7026203.html Sent from the freeswitch-users mailing list archive at Nabble.com. From admin at blindi.net Thu Nov 24 01:22:24 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Wed, 23 Nov 2011 23:22:24 +0100 (CET) Subject: [Freeswitch-users] Question to cancel originate tries In-Reply-To: References: Message-ID: Hi all, I make a originate callback: bgapi originate {ignore_early_media=true,originate_retries=10000, originate_retry_sleep_ms=60000, originate_timeout=900}loopback/08925007676/disa &conference(7400-www.blindi.net at default) Can i break the running originate_retries? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From anthony.minessale at gmail.com Thu Nov 24 01:25:22 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Nov 2011 16:25:22 -0600 Subject: [Freeswitch-users] Question to cancel originate tries In-Reply-To: References: Message-ID: if you create you own uuid and send it in as {origination_uuid=} then you can uuid_kill it On Wed, Nov 23, 2011 at 4:22 PM, Thomas Hoellriegel wrote: > Hi all, > I make a originate callback: > bgapi originate > {ignore_early_media=true,**originate_retries=10000, > originate_retry_sleep_ms=**60000, > originate_timeout=900}**loopback/08925007676/disa > &conference(7400-www.blindi.**net at default) > > Can i break the running originate_retries? > > Thanks. > > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/8f4438e5/attachment.html From admin at blindi.net Thu Nov 24 02:26:36 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Thu, 24 Nov 2011 00:26:36 +0100 (CET) Subject: [Freeswitch-users] Question to cancel originate tries In-Reply-To: References: Message-ID: Hi Anthony, The problem: the call is no answer. i can kill a allready existing call. I want cancel the number of dial attempts prematurely. can you help please? thankx. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From anthony.minessale at gmail.com Thu Nov 24 02:49:08 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Nov 2011 17:49:08 -0600 Subject: [Freeswitch-users] Question to cancel originate tries In-Reply-To: References: Message-ID: Did you miss my instructions? http://wiki.freeswitch.org/wiki/Variable_origination_uuid On Wed, Nov 23, 2011 at 5:26 PM, Thomas Hoellriegel wrote: > Hi Anthony, > The problem: the call is no answer. i can kill a allready existing call. > I want cancel the number of dial attempts prematurely. > can you help please? > thankx. > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111123/538e0689/attachment.html From nagalenoj at gmail.com Thu Nov 24 09:24:57 2011 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Thu, 24 Nov 2011 11:54:57 +0530 Subject: [Freeswitch-users] using break application after bridge In-Reply-To: References: Message-ID: Thanks. I've used hupall and it worked fine for me. On Wed, Nov 23, 2011 at 12:52 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you can't use break to stop bridging. > you can do "uuid_transfer park inline" to put it into an idle state > > > > On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. wrote: > >> Hi, >> I wanted to cancel the bridge application after issuing it and I tried >> using 'break' application. break application didn't work. But, when I tried >> using playback application followed by break, it worked. >> >> connect >> >> sendmsg >> call-command: execute >> execute-app-name: answer >> >> sendmsg >> call-command: execute >> execute-app-name: bridge >> execute-app-arg: user/1001 >> >> sendmsg >> call-command: execute >> execute-app-name: break >> >> Logs: >> http://pastebin.freeswitch.org/17843 >> >> Git version as of 18/11/2011. >> >> -- >> Regards, >> Nagalenoj H. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111124/d0f165df/attachment-0001.html From charlie.orford at attackplan.net Thu Nov 24 13:17:33 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Thu, 24 Nov 2011 11:17:33 +0100 Subject: [Freeswitch-users] Reasons for a SIP/2.0 400 Bad Request ? Message-ID: <4ECE19BD.4050707@attackplan.net> Hi list I have done some googling but cannot find much info about what causes FS to send a SIP/2.0 400 Bad Request response. My phones are able to register but when I try a test call (e.g. to 9196) FS responds to the INVITE with Bad Request. For a short transcript of this in action see: http://pastebin.freeswitch.org/17863 My setup: I have a clean git head install (cloned yesterday) and am running the unmodified "out of the box" configuration (except I have enabled NDLB-force-rport on the internal profile and set a default poassword). FS is on a public server (no nat) and phones (aastra 57i models) are all natted behind an ADSL router. Thanks & Regards, Charlie From x.liu at hw.ac.uk Thu Nov 24 14:35:48 2011 From: x.liu at hw.ac.uk (xl127) Date: Thu, 24 Nov 2011 11:35:48 +0000 Subject: [Freeswitch-users] problems in Using Embedded FreeSWITCH In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68D@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B68E@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C5B2715B690@cooper> <4ECAA293.50408@hw.ac.uk> <4ECB9736.90005@hw.ac.uk> <4ECBCBD8.3010105@hw.ac.uk> Message-ID: <4ECE2C14.5020406@hw.ac.uk> Hi, The problem has been solved. I report it back here in case it might be helpful to others who would run into similar problems. Basically I added switch_core_init(..) and set parameters for the global variable: SWITCH_GLOBAL_dirs Below are the working codes: ====================================================================== #include #include #include int main(int argc, char** argv) { switch_core_flag_t flags = SCF_USE_SQL; bool console = true; const char *err = NULL; switch_core_set_globals(); QDir conf_dir = QDir::current(); SWITCH_GLOBAL_dirs.conf_dir = (char *) malloc(strlen(QString("%1/conf").arg(conf_dir.absolutePath()).toAscii().constData()) + 1); if (!SWITCH_GLOBAL_dirs.conf_dir) { fprintf(stderr, "Cannot allocate memory for conf_dir."); } strcpy(SWITCH_GLOBAL_dirs.conf_dir, QString("%1/conf").arg(conf_dir.absolutePath()).toAscii().constData()); fprintf(stderr, "AAA%s\n",SWITCH_GLOBAL_dirs.conf_dir); SWITCH_GLOBAL_dirs.log_dir = (char *) malloc(strlen(QString("%1/log").arg(conf_dir.absolutePath()).toAscii().constData()) + 1); if (!SWITCH_GLOBAL_dirs.log_dir) { fprintf(stderr,"Cannot allocate memory for log_dir."); } strcpy(SWITCH_GLOBAL_dirs.log_dir, QString("%1/log").arg(conf_dir.absolutePath()).toAscii().constData()); SWITCH_GLOBAL_dirs.run_dir = (char *) malloc(strlen(QString("%1/run").arg(conf_dir.absolutePath()).toAscii().constData()) + 1); if (!SWITCH_GLOBAL_dirs.run_dir) { fprintf(stderr,"Cannot allocate memory for run_dir."); } strcpy(SWITCH_GLOBAL_dirs.run_dir, QString("%1/run").arg(conf_dir.absolutePath()).toAscii().constData()); SWITCH_GLOBAL_dirs.db_dir = (char *) malloc(strlen(QString("%1/db").arg(conf_dir.absolutePath()).toAscii().constData()) + 1); if (!SWITCH_GLOBAL_dirs.db_dir) { fprintf(stderr,"Cannot allocate memory for db_dir."); } strcpy(SWITCH_GLOBAL_dirs.db_dir, QString("%1/db").arg(conf_dir.absolutePath()).toAscii().constData()); SWITCH_GLOBAL_dirs.script_dir = (char *) malloc(strlen(QString("%1/script").arg(conf_dir.absolutePath()).toAscii().constData()) + 1); if (!SWITCH_GLOBAL_dirs.script_dir) { fprintf(stderr,"Cannot allocate memory for script_dir."); } strcpy(SWITCH_GLOBAL_dirs.script_dir, QString("%1/script").arg(conf_dir.absolutePath()).toAscii().constData()); SWITCH_GLOBAL_dirs.htdocs_dir = (char *) malloc(strlen(QString("%1/htdocs").arg(conf_dir.absolutePath()).toAscii().constData()) + 1); if (!SWITCH_GLOBAL_dirs.htdocs_dir) { fprintf(stderr,"Cannot allocate memory for htdocs_dir."); } strcpy(SWITCH_GLOBAL_dirs.htdocs_dir, QString("%1/htdocs").arg(conf_dir.absolutePath()).toAscii().constData()); SWITCH_GLOBAL_dirs.grammar_dir = (char *) malloc(strlen(QString("%1/grammar").arg(conf_dir.absolutePath()).toAscii().constData()) + 1); if (!SWITCH_GLOBAL_dirs.grammar_dir) { fprintf(stderr,"Cannot allocate memory for grammar_dir."); } strcpy(SWITCH_GLOBAL_dirs.grammar_dir, QString("%1/grammar").arg(conf_dir.absolutePath()).toAscii().constData()); /* Initialize the core and load modules, that will startup FS completely */ //if (switch_core_init(flags, console, &err) != SWITCH_STATUS_SUCCESS) { if (switch_core_init(flags, console ? SWITCH_TRUE : SWITCH_FALSE, &err) != SWITCH_STATUS_SUCCESS) { fprintf(stderr, "Failed to initialize FreeSWITCH's core: %s\n", err); } //switch_core_init_and_modload(flags, console ? SWITCH_TRUE : SWITCH_FALSE, &err); switch_core_init_and_modload(flags, SWITCH_FALSE, &err); switch_core_runtime_loop(!console); return 0; } ==================================================================== cheers, Xing On 22/11/11 16:47, Anthony Minessale wrote: > maybe your user does not have permissions to read and write the target > install dir > try to chown -R to your user on all of /usr/local/freeswitch > > you might want to try things the other way around and embedd your app > in FS rather than embed FS in your app. > > > On Tue, Nov 22, 2011 at 10:20 AM, xl127 > wrote: > > If I switch to root user I can successfully run my embedded FS > app: FS_Embed. > but it will fails if I run it as non-root user. > > One thing I can think of to do is re-install FS as non-root. > I tried but it failed to access > /usr/lib/python2.7/site-packages/freeswitch.py due to the > permissions. > > > > On 22/11/11 16:08, Anthony Minessale wrote: >> i know you needed apr, not sure, maybe try strace or gdb >> >> >> On Tue, Nov 22, 2011 at 6:36 AM, xl127 > > wrote: >> >> Thanks for the advice! >> >> I tried it but still got the same error. >> >> I added the #include and found it from >> /usr/local/sr/freeswitch/libs/apr/include >> rather than from the installed directory >> /usr/local/freeswitch. Hope it is the right one. >> >> any more clues? >> >> >> >> On 22/11/11 01:38, Anthony Minessale wrote: >>> try >>> >>> if (apr_initialize() != SWITCH_STATUS_SUCCESS) { >>> fprintf(stderr, "FATAL ERROR! Could not initialize >>> APR\n"); >>> return 255; >>> } >>> >>> early on in main() >>> >>> >>> On Mon, Nov 21, 2011 at 1:12 PM, xl127 >> > wrote: >>> >>> Hello, >>> >>> Following the instructions in >>> http://wiki.freeswitch.org/wiki/Embedding_FreeSWITCH >>> I am trying to use Embedded FreeSwitch. >>> >>> I installed FS in Fedora linux as a root user in >>> /usr/local/freeswitch. >>> Now as non-root user, >>> I compiled following codes: >>> >>> #include >>> int main(int argc, char** argv) >>> { >>> switch_core_flag_t flags = SCF_USE_SQL; >>> bool console = true; >>> const char *err = NULL; >>> switch_core_set_globals(); >>> switch_core_init_and_modload(flags, console ? >>> SWITCH_TRUE : >>> SWITCH_FALSE, &err); >>> switch_core_runtime_loop(!console); >>> return 0; >>> } >>> >>> And copy the executable and conf, mod, contents of lib >>> of the installed >>> FS to my $MyWorkingDir. >>> >>> My directory structure looks like: >>> MyWorkingDir--bin/ (contains my executable: FS_Embed) >>> --conf/ >>> --mod/ >>> --lib/ >>> --grammars >>> --sounds >>> --scripts >>> --log >>> (I also tried to copy the contents of lib directory into >>> MyWorkingDir) >>> >>> When I run my embedded FS: ./FS_Embed, I got error >>> >>> 2011-11-21 18:46:26.606783 [INFO] switch_event.c:631 >>> Activate Eventing >>> Engine. >>> 2011-11-21 18:46:26.617941 [DEBUG] switch_event.c:610 >>> Create event >>> dispatch thread 0 >>> FS_Embed: src/switch_xml.c:2225: switch_xml_open_cfg: >>> Assertion >>> `MAIN_XML_ROOT != ((void *)0)' failed. >>> Aborted (core dumped) >>> >>> and occasionally on another run, I got error: >>> >>> 2011-11-21 16:56:51.756040 [INFO] switch_event.c:631 >>> Activate Eventing >>> Engine. >>> 2011-11-21 16:56:51.767198 [DEBUG] switch_event.c:610 >>> Create event >>> dispatch thread 0 >>> Segmentation fault (core dumped) >>> >>> It looks like there are some things wrong with my >>> Environment Setup for >>> Embedded FS, >>> but after a while of checking/trying/googling I couldn't >>> find what's wrong. >>> >>> Any advice please? >>> >>> Many thanks! >>> Xing >>> >>> >>> >>> -- >>> Heriot-Watt University is a Scottish charity >>> registered under charity number SC000278. >>> >>> Heriot-Watt University is the Sunday Times >>> Scottish University of the Year 2011-2012 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> googletalk:conf+888 at conference.freeswitch.org >>> >>> pstn:+19193869900 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> >> >> ------------------------------------------------------------------------ >> >> MailScanner Signature HW *Heriot-Watt University is the >> Sunday Times >> >> Scottish University of the Year 2011-2012 >> * >> >> Heriot-Watt University is a Scottish charity >> registered under charity number SC000278. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > ------------------------------------------------------------------------ > > MailScanner Signature HW *Heriot-Watt University is the Sunday Times > Scottish University of the Year 2011-2012* > > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111124/dacf5a42/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111124/dacf5a42/attachment-0002.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111124/dacf5a42/attachment-0003.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: hw_uni_of_year.jpg Type: image/jpeg Size: 4803 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111124/dacf5a42/attachment-0001.jpg From Nabble_01394 at slickdeals.endjunk.com Thu Nov 24 14:47:00 2011 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Thu, 24 Nov 2011 03:47:00 -0800 (PST) Subject: [Freeswitch-users] wimax double nat In-Reply-To: <1322057378507-7024457.post@n2.nabble.com> References: <1322057378507-7024457.post@n2.nabble.com> Message-ID: <1322135220548-7027833.post@n2.nabble.com> new_voip wrote > ----------------------------------------------------------------------------------------------------- > *case 1* (does not work) > FS(public static IP) <-- internet <--(public IP ) Nat <-- 10.x.x.x (CPE) > NAT <-- ATA (192.168.x.x) > ----------------------------------------------------------------------------------------------------- Is there a reason for a double NAT/Firewall? If no, then remove the additional layer of NAT/Firewall. ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/wimax-double-nat-tp7024457p7027833.html Sent from the freeswitch-users mailing list archive at Nabble.com. From chrisbware at interfree.it Thu Nov 24 16:29:09 2011 From: chrisbware at interfree.it (chrisbware at interfree.it) Date: 24 Nov 2011 13:29:09 -0000 Subject: [Freeswitch-users] Reasons for a SIP/2.0 400 Bad Request ? Message-ID: <20111124132909.27202.qmail@community17.interfree.it> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111124/8f61e3d6/attachment.html From charlie.orford at attackplan.net Thu Nov 24 16:47:22 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Thu, 24 Nov 2011 14:47:22 +0100 Subject: [Freeswitch-users] Reasons for a SIP/2.0 400 Bad Request ? In-Reply-To: <20111124132909.27202.qmail@community17.interfree.it> References: <20111124132909.27202.qmail@community17.interfree.it> Message-ID: <4ECE4AEA.7080105@attackplan.net> Hi Chris, I've increased the debugging and taken a new trace: http://pastebin.freeswitch.org/17864 And here is the output of tport_sip.log: http://pastebin.freeswitch.org/17865 A person on the #freeswitch IRC channel suggested the problem might be due to packet fragmentation? Charlie On 24/11/2011 14:29, chrisbware at interfree.it wrote: > I suspect problem is on SDP but we can't see it on pastebin. > -----Messaggio originale----- > Da: Charlie Orford > Inviato il: 24 Nov 2011 - 10:17 > A: FreeSWITCH Users Help > > > Hi list > > I have done some googling but cannot find much info about what causes FS > to send a SIP/2.0 400 Bad Request response. > > My phones are able to register but when I try a test call (e.g. to 9196) > FS responds to the INVITE with Bad Request. For a short transcript of > this in action see: http://pastebin.freeswitch.org/17863 > > My setup: > I have a clean git head install (cloned yesterday) and am running the > unmodified "out of the box" configuration (except I have enabled > NDLB-force-rport on the internal profile and set a default poassword). > FS is on a public server (no nat) and phones (aastra 57i models) are all > natted behind an ADSL router. > > Thanks & Regards, > Charlie > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > ------------------------------------------------------------------------------- > Valore legale alle tue mail > InterfreePEC - la tua Posta Elettronica Certificata > http://pec.interfree.it > ------------------------------------------------------------------------------- From chrisbware at interfree.it Thu Nov 24 16:59:20 2011 From: chrisbware at interfree.it (chrisbware at interfree.it) Date: 24 Nov 2011 13:59:20 -0000 Subject: [Freeswitch-users] Reasons for a SIP/2.0 400 Bad Request ? Message-ID: <20111124135920.7315.qmail@community17.interfree.it> Your INVITE is 1672 bytes. Some router split packet over 1500 bytes. To test if this is the problema, disable all codecs leaving only "basic codecs" as Aastra call them (g711a,g711u,g729) and disable ice. Packet should be smaller: if the problem is fixed in this way, packet fragmentation is the cause. -----Messaggio originale----- Da: Charlie Orford Inviato il: 24 Nov 2011 - 13:47 A: FreeSWITCH Users Help Hi Chris, I've increased the debugging and taken a new trace: http://pastebin.freeswitch.org/17864 And here is the output of tport_sip.log: http://pastebin.freeswitch.org/17865 A person on the #freeswitch IRC channel suggested the problem might be due to packet fragmentation? Charlie On 24/11/2011 14:29, chrisbware at interfree.it wrote: > I suspect problem is on SDP but we can't see it on pastebin. > -----Messaggio originale----- > Da: Charlie Orford > Inviato il: 24 Nov 2011 - 10:17 > A: FreeSWITCH Users Help > > > Hi list > > I have done some googling but cannot find much info about what causes FS > to send a SIP/2.0 400 Bad Request response. > > My phones are able to register but when I try a test call (e.g. to 9196) > FS responds to the INVITE with Bad Request. For a short transcript of > this in action see: http://pastebin.freeswitch.org/17863 > > My setup: > I have a clean git head install (cloned yesterday) and am running the > unmodified "out of the box" configuration (except I have enabled > NDLB-force-rport on the internal profile and set a default poassword). > FS is on a public server (no nat) and phones (aastra 57i models) are all > natted behind an ADSL router. > > Thanks & Regards, > Charlie > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > ------------------------------------------------------------------------------- > Valore legale alle tue mail > InterfreePEC - la tua Posta Elettronica Certificata > http://pec.interfree.it > ------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 ------------------------------------------------------------------------------- Valore legale alle tue mail InterfreePEC - la tua Posta Elettronica Certificata http://pec.interfree.it ------------------------------------------------------------------------------- From admin at blindi.net Thu Nov 24 17:03:40 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Thu, 24 Nov 2011 15:03:40 +0100 (CET) Subject: [Freeswitch-users] removing participant from 3-way bridge In-Reply-To: References: Message-ID: Hi Anthony, thanks for your nice help !!! It.s fine!! Best regards from munic Germany. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From charlie.orford at attackplan.net Thu Nov 24 17:27:13 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Thu, 24 Nov 2011 15:27:13 +0100 Subject: [Freeswitch-users] Reasons for a SIP/2.0 400 Bad Request ? In-Reply-To: <20111124135920.7315.qmail@community17.interfree.it> References: <20111124135920.7315.qmail@community17.interfree.it> Message-ID: <4ECE5441.1010309@attackplan.net> Hmm it doesn't look like it is a fragmentation problem. Same "400 Bad Request" response even with the INVITE at 995 bytes: http://pastebin.freeswitch.org/17868 http://pastebin.freeswitch.org/17867 FYI, issuing "ping -f -l 1470 my.fs.server.com" from a windows workstation on the same network as the phone goes through ok (so I think this means as long as the INVITE packet is 1470 bytes or less, it should not get fragmented). Is there anything else I can debug to track this down? Charlie On 24/11/2011 14:59, chrisbware at interfree.it wrote: > Your INVITE is 1672 bytes. Some router split packet over 1500 bytes. To test if this is the problema, disable all codecs leaving only "basic codecs" as Aastra call them (g711a,g711u,g729) and disable ice. Packet should be smaller: if the problem is fixed in this way, packet fragmentation is the cause. > > > -----Messaggio originale----- > Da: Charlie Orford > Inviato il: 24 Nov 2011 - 13:47 > A: FreeSWITCH Users Help > > > Hi Chris, > > I've increased the debugging and taken a new trace: > > http://pastebin.freeswitch.org/17864 > > And here is the output of tport_sip.log: > > http://pastebin.freeswitch.org/17865 > > A person on the #freeswitch IRC channel suggested the problem might be > due to packet fragmentation? > > Charlie > > > On 24/11/2011 14:29, chrisbware at interfree.it wrote: >> I suspect problem is on SDP but we can't see it on pastebin. >> -----Messaggio originale----- >> Da: Charlie Orford >> Inviato il: 24 Nov 2011 - 10:17 >> A: FreeSWITCH Users Help >> >> >> Hi list >> >> I have done some googling but cannot find much info about what causes FS >> to send a SIP/2.0 400 Bad Request response. >> >> My phones are able to register but when I try a test call (e.g. to 9196) >> FS responds to the INVITE with Bad Request. For a short transcript of >> this in action see: http://pastebin.freeswitch.org/17863 >> >> My setup: >> I have a clean git head install (cloned yesterday) and am running the >> unmodified "out of the box" configuration (except I have enabled >> NDLB-force-rport on the internal profile and set a default poassword). >> FS is on a public server (no nat) and phones (aastra 57i models) are all >> natted behind an ADSL router. >> >> Thanks& Regards, >> Charlie >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> ------------------------------------------------------------------------------- >> Valore legale alle tue mail >> InterfreePEC - la tua Posta Elettronica Certificata >> http://pec.interfree.it >> ------------------------------------------------------------------------------- From admin at blindi.net Thu Nov 24 17:45:50 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Thu, 24 Nov 2011 15:45:50 +0100 (CET) Subject: [Freeswitch-users] originate bridge breaks about after 2 seconds In-Reply-To: References: Message-ID: Hi i make a originatebride from a shellscript: #!/bin/bash /usr/local/freeswitch/bin/fs_cli -x "bgapi originate\ {ignore_early_media=true,hangup_after_bridge=false,\ originate_retries=100,origination_uuid=thomas_$1,\ origination_caller_id_name=Callback,originate_retry_sleep_ms=60000,\ originate_timeout=900}loopback/$1/disa\ &conference(7400-sip.blindi.net at default)" for example: i call 10 users in a conference, these works fine. I bridge 2 existing calls for example: uuid_transfer thomas_1 park inline uuid_transfer thomas_2 park inline uuid_bridge thomas_1 thomas_2 The bridge works abbout 2 seconds, and fs hangup the bridge. can your help please? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From charlie.orford at attackplan.net Thu Nov 24 18:34:44 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Thu, 24 Nov 2011 16:34:44 +0100 Subject: [Freeswitch-users] Reasons for a SIP/2.0 400 Bad Request ? [ SOLVED ] In-Reply-To: <4ECE5441.1010309@attackplan.net> References: <20111124135920.7315.qmail@community17.interfree.it> <4ECE5441.1010309@attackplan.net> Message-ID: <4ECE6414.9070903@attackplan.net> Hi Chris I don't know why but replacing the router (a 5 year old Draytek Vigor 2600) with a linux machine + pppoe modem fixed the problem. No change was needed with the phones and they are still behind NAT (but this time its managed by iptables). Buggy router I guess.... On 24/11/2011 15:27, Charlie Orford wrote: > Hmm it doesn't look like it is a fragmentation problem. Same "400 Bad > Request" response even with the INVITE at 995 bytes: > > http://pastebin.freeswitch.org/17868 > http://pastebin.freeswitch.org/17867 > > FYI, issuing "ping -f -l 1470 my.fs.server.com" from a windows > workstation on the same network as the phone goes through ok (so I think > this means as long as the INVITE packet is 1470 bytes or less, it should > not get fragmented). > > Is there anything else I can debug to track this down? > > Charlie > > > On 24/11/2011 14:59, chrisbware at interfree.it wrote: >> Your INVITE is 1672 bytes. Some router split packet over 1500 bytes. To test if this is the problema, disable all codecs leaving only "basic codecs" as Aastra call them (g711a,g711u,g729) and disable ice. Packet should be smaller: if the problem is fixed in this way, packet fragmentation is the cause. >> >> >> -----Messaggio originale----- >> Da: Charlie Orford >> Inviato il: 24 Nov 2011 - 13:47 >> A: FreeSWITCH Users Help >> >> >> Hi Chris, >> >> I've increased the debugging and taken a new trace: >> >> http://pastebin.freeswitch.org/17864 >> >> And here is the output of tport_sip.log: >> >> http://pastebin.freeswitch.org/17865 >> >> A person on the #freeswitch IRC channel suggested the problem might be >> due to packet fragmentation? >> >> Charlie >> >> >> On 24/11/2011 14:29, chrisbware at interfree.it wrote: >>> I suspect problem is on SDP but we can't see it on pastebin. >>> -----Messaggio originale----- >>> Da: Charlie Orford >>> Inviato il: 24 Nov 2011 - 10:17 >>> A: FreeSWITCH Users Help >>> >>> >>> Hi list >>> >>> I have done some googling but cannot find much info about what causes FS >>> to send a SIP/2.0 400 Bad Request response. >>> >>> My phones are able to register but when I try a test call (e.g. to 9196) >>> FS responds to the INVITE with Bad Request. For a short transcript of >>> this in action see: http://pastebin.freeswitch.org/17863 >>> >>> My setup: >>> I have a clean git head install (cloned yesterday) and am running the >>> unmodified "out of the box" configuration (except I have enabled >>> NDLB-force-rport on the internal profile and set a default poassword). >>> FS is on a public server (no nat) and phones (aastra 57i models) are all >>> natted behind an ADSL router. >>> >>> Thanks& Regards, >>> Charlie >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> ------------------------------------------------------------------------------- >>> Valore legale alle tue mail >>> InterfreePEC - la tua Posta Elettronica Certificata >>> http://pec.interfree.it >>> ------------------------------------------------------------------------------- > From avi at avimarcus.net Thu Nov 24 18:50:29 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 24 Nov 2011 17:50:29 +0200 Subject: [Freeswitch-users] Reasons for a SIP/2.0 400 Bad Request ? [ SOLVED ] In-Reply-To: <4ECE6414.9070903@attackplan.net> References: <20111124135920.7315.qmail@community17.interfree.it> <4ECE5441.1010309@attackplan.net> <4ECE6414.9070903@attackplan.net> Message-ID: On a related note, my parents complained about dropped SIP calls on a Linksys ATA. Replacing the 3 yr old Aztech DSL modem, and the verizon actiontec router, with a new d-link modem/router has resulted in no dropped calls. So.. buggy modem/routers affecting calls might be more common than you think. -Avi Marcus On Thu, Nov 24, 2011 at 5:34 PM, Charlie Orford < charlie.orford at attackplan.net> wrote: > Hi Chris > > I don't know why but replacing the router (a 5 year old Draytek Vigor > 2600) with a linux machine + pppoe modem fixed the problem. > > No change was needed with the phones and they are still behind NAT (but > this time its managed by iptables). > > Buggy router I guess.... > > > On 24/11/2011 15:27, Charlie Orford wrote: > > Hmm it doesn't look like it is a fragmentation problem. Same "400 Bad > > Request" response even with the INVITE at 995 bytes: > > > > http://pastebin.freeswitch.org/17868 > > http://pastebin.freeswitch.org/17867 > > > > FYI, issuing "ping -f -l 1470 my.fs.server.com" from a windows > > workstation on the same network as the phone goes through ok (so I think > > this means as long as the INVITE packet is 1470 bytes or less, it should > > not get fragmented). > > > > Is there anything else I can debug to track this down? > > > > Charlie > > > > > > On 24/11/2011 14:59, chrisbware at interfree.it wrote: > >> Your INVITE is 1672 bytes. Some router split packet over 1500 bytes. To > test if this is the problema, disable all codecs leaving only "basic > codecs" as Aastra call them (g711a,g711u,g729) and disable ice. Packet > should be smaller: if the problem is fixed in this way, packet > fragmentation is the cause. > >> > >> > >> -----Messaggio originale----- > >> Da: Charlie Orford > >> Inviato il: 24 Nov 2011 - 13:47 > >> A: FreeSWITCH Users Help > >> > >> > >> Hi Chris, > >> > >> I've increased the debugging and taken a new trace: > >> > >> http://pastebin.freeswitch.org/17864 > >> > >> And here is the output of tport_sip.log: > >> > >> http://pastebin.freeswitch.org/17865 > >> > >> A person on the #freeswitch IRC channel suggested the problem might be > >> due to packet fragmentation? > >> > >> Charlie > >> > >> > >> On 24/11/2011 14:29, chrisbware at interfree.it wrote: > >>> I suspect problem is on SDP but we can't see it on pastebin. > >>> -----Messaggio originale----- > >>> Da: Charlie Orford > >>> Inviato il: 24 Nov 2011 - 10:17 > >>> A: FreeSWITCH Users Help > >>> > >>> > >>> Hi list > >>> > >>> I have done some googling but cannot find much info about what causes > FS > >>> to send a SIP/2.0 400 Bad Request response. > >>> > >>> My phones are able to register but when I try a test call (e.g. to > 9196) > >>> FS responds to the INVITE with Bad Request. For a short transcript of > >>> this in action see: http://pastebin.freeswitch.org/17863 > >>> > >>> My setup: > >>> I have a clean git head install (cloned yesterday) and am running the > >>> unmodified "out of the box" configuration (except I have enabled > >>> NDLB-force-rport on the internal profile and set a default poassword). > >>> FS is on a public server (no nat) and phones (aastra 57i models) are > all > >>> natted behind an ADSL router. > >>> > >>> Thanks& Regards, > >>> Charlie > >>> > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.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 > >>> > >>> > ------------------------------------------------------------------------------- > >>> Valore legale alle tue mail > >>> InterfreePEC - la tua Posta Elettronica Certificata > >>> http://pec.interfree.it > >>> > ------------------------------------------------------------------------------- > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111124/efd46c06/attachment.html From freeswitch at peely.com Thu Nov 24 19:02:52 2011 From: freeswitch at peely.com (peely) Date: Thu, 24 Nov 2011 08:02:52 -0800 (PST) Subject: [Freeswitch-users] Reasons for a SIP/2.0 400 Bad Request ? [ SOLVED ] In-Reply-To: References: <4ECE19BD.4050707@attackplan.net> <20111124135920.7315.qmail@community17.interfree.it> <4ECE5441.1010309@attackplan.net> <4ECE6414.9070903@attackplan.net> Message-ID: <1322150572567-7028646.post@n2.nabble.com> Most SIP routers with in-built ALGs that old are probably poor implementations. It may well have messed about with the SDP offer but not changed the content length in the SIP header. I use Draytek's and frequently find I have to change the local SIP port to anything other than 5060 to stop it messing about with the signaling. FreeSWITCH does a much better job at NAT traversal from the far end than most modems can achieve at the near end. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Reasons-for-a-SIP-2-0-400-Bad-Request-tp7027631p7028646.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sdame at 207me.com Fri Nov 25 00:08:19 2011 From: sdame at 207me.com (Stephen Dame) Date: Thu, 24 Nov 2011 16:08:19 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: <005501cca52f$aa0286c0$fe079440$@com> Message-ID: <021b01ccaaed$31db2bc0$95918340$@com> Fraser, thanks for tips in compiling kernel. I was able to set up a 1000hz ec2 instance ... and run some comparative tests using the same starting 10.04 LTS ami. Here are the results of timing tests.. Not sure how to interpret. Having users test audio right now in voice conferences to see if they notice any difference. Looked in wiki. I'm still not clear on what to look for. Regards, Stephen 100hz kernel time_test 600 10 test 1 sleep 600 8847 test 2 sleep 600 9969 test 3 sleep 600 10006 test 4 sleep 600 9986 test 5 sleep 600 10003 test 6 sleep 600 9977 test 7 sleep 600 10117 test 8 sleep 600 9872 test 9 sleep 600 10014 test 10 sleep 600 10124 timer_test 120 10 2011-11-24 14:53:51.142568 [CONSOLE] mod_commands.c:329 Timer Test: 1 sleep 120 120289 2011-11-24 14:53:51.263610 [CONSOLE] mod_commands.c:329 Timer Test: 2 sleep 120 120703 2011-11-24 14:53:51.382587 [CONSOLE] mod_commands.c:329 Timer Test: 3 sleep 120 118952 2011-11-24 14:53:51.502640 [CONSOLE] mod_commands.c:329 Timer Test: 4 sleep 120 120048 2011-11-24 14:53:51.622851 [CONSOLE] mod_commands.c:329 Timer Test: 5 sleep 120 120213 2011-11-24 14:53:51.742593 [CONSOLE] mod_commands.c:329 Timer Test: 6 sleep 120 119681 2011-11-24 14:53:51.862583 [CONSOLE] mod_commands.c:329 Timer Test: 7 sleep 120 119963 2011-11-24 14:53:51.982568 [CONSOLE] mod_commands.c:329 Timer Test: 8 sleep 120 119975 2011-11-24 14:53:52.133931 [CONSOLE] mod_commands.c:329 Timer Test: 9 sleep 120 151338 2011-11-24 14:53:52.232591 [CONSOLE] mod_commands.c:329 Timer Test: 10 sleep 120 98753 1000hz kernel time_test 600 10 test 1 sleep 600 1591 test 2 sleep 600 1009 test 3 sleep 600 971 test 4 sleep 600 1114 test 5 sleep 600 888 test 6 sleep 600 996 test 7 sleep 600 998 test 8 sleep 600 988 test 9 sleep 600 994 test 10 sleep 600 996 avg 1054 timer_test 120 10 2011-11-24 19:55:27.198564 [CONSOLE] mod_commands.c:562 Timer Test: 1 sleep 120 119956 2011-11-24 19:55:27.318559 [CONSOLE] mod_commands.c:562 Timer Test: 2 sleep 120 119994 2011-11-24 19:55:27.438559 [CONSOLE] mod_commands.c:562 Timer Test: 3 sleep 120 119939 2011-11-24 19:55:27.558559 [CONSOLE] mod_commands.c:562 Timer Test: 4 sleep 120 119965 2011-11-24 19:55:27.678559 [CONSOLE] mod_commands.c:562 Timer Test: 5 sleep 120 119967 2011-11-24 19:55:27.798563 [CONSOLE] mod_commands.c:562 Timer Test: 6 sleep 120 119977 2011-11-24 19:55:27.918559 [CONSOLE] mod_commands.c:562 Timer Test: 7 sleep 120 120000 2011-11-24 19:55:28.038559 [CONSOLE] mod_commands.c:562 Timer Test: 8 sleep 120 119936 2011-11-24 19:55:28.158559 [CONSOLE] mod_commands.c:562 Timer Test: 9 sleep 120 119973 2011-11-24 19:55:28.278558 [CONSOLE] mod_commands.c:562 Timer Test: 10 sleep 120 119968 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fraser Redmond Sent: Saturday, November 19, 2011 7:43 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Timing issues in AWS? Hi Stephen, I spent most of last week working out how to upgrade the kernel timer from 100HZ to 1000HZ on ubuntu on AWS. I documented the steps I took here: http://wiki.freeswitch.org/wiki/Amazon_EC2#Updating_Kernel_Timer_to_1000HZ Making that change gave us a noticeable jump in call quality. Cheers, Fraser On 17 November 2011 08:49, Stephen Dame wrote: I'm running freeswitch on about 40 different m1.small c1.medium's in AWS regions, us-east, us-west, eu-west, and asia.. They are used in videoconf component of BigBlueButton.org . For most part they work great. There are occasional issues with voip quality but the app is 100% voip, with the BBB client all browser based. So the conference is subject to every ones local network connections and most issues "blamed" on the internet instead of freeswitch J We also tie in skype and DID direct to improve latency for some clients. But expectations are set so we meet them. You probably have tougher business conferencing clients that want perfect audio. But these are production deployed and generate revenue. All these are on Ubuntu 10.04 official amis with no mods to kernels. I'm sure there are tweaks that can be made, and bare metal solutions that would work a little better. Does anyone have any ideas how to optimize a Ubuntu instance. I would engage in a few hours of consulting is so. The tests below are interesting but above my paygrade to understand what they mean? I'm running them but don't have a clue how to interpret. If you want a simple of real running data, would be glad to run sample tests on these distributed servers and provide back for analysis. Regards, Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Picher Sent: Thursday, November 17, 2011 8:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Timing issues in AWS? Not having anywhere near the same trouble in a Xen Server in my lab... Although test_time is skewed 180 degrees from what it was in AWS.... freeswitch at internal> timer_test 120 10 Avg: 119.945ms Total Time: 1199.727ms 2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120993 2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 118914 2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119919 2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120032 2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 119802 2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 119999 2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119975 2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 120003 2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 119849 2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119971 freeswitch at internal> time_test 600 10 test 1 sleep 600 1 test 2 sleep 600 1 test 3 sleep 600 0 test 4 sleep 600 1 test 5 sleep 600 0 test 6 sleep 600 1 test 7 sleep 600 1 test 8 sleep 600 0 test 9 sleep 600 0 test 10 sleep 600 1 avg 0 [root at openuc bin]# uname -r 2.6.18-274.7.1.el5 [root at openuc bin]# grep CONFIG_HZ /boot/config-* /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000 Thoughts as to why AWS results are so different from XenServer? Other than not knowing who else is on the AWS box? Thanks, Mike On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher wrote: m1.large I have a c1.xlarge queued up and ready to test... On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen wrote: Just a simple question, what kind of AWS instance are you running your FreeSWITCH? It makes huge difference. Thanks, Chris On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher wrote: Hi guys, Trying to get to the bottom of some conference bridge issues I'm having with running the system in AWS. We're hearing a bunch of snap-crackle-pops in conference bridges and when I tcpdum on the server itself I see them in the RTP and see RTP timestamp problems. I've run the following: timer_test freeswitch at 127.0.0.1@internal> timer_test 120 10 Avg: 120.004ms Total Time: 1200.315ms 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120006 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 120040 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119976 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120007 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 120001 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 120031 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119996 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 119994 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 120005 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119991 test_time freeswitch at 127.0.0.1@internal> time_test 600 10 test 1 sleep 600 1592 test 2 sleep 600 986 test 3 sleep 600 1018 test 4 sleep 600 980 test 5 sleep 600 1005 test 6 sleep 600 1000 test 7 sleep 600 972 test 8 sleep 600 990 test 9 sleep 600 1006 test 10 sleep 600 994 avg 1054 For kernel: [root at openuc bin]# uname -r 2.6.21.7-2.fc8xen CONFIG_HZ: [root at openuc bin]# grep CONFIG_HZ /boot/config-* /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 are the xenU kernel settings screwing me here? Thanks, Mike -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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/20111124/576c6308/attachment-0001.html From fraserredmond at gmail.com Fri Nov 25 03:10:34 2011 From: fraserredmond at gmail.com (Fraser Redmond) Date: Thu, 24 Nov 2011 19:10:34 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: <021b01ccaaed$31db2bc0$95918340$@com> References: <005501cca52f$aa0286c0$fe079440$@com> <021b01ccaaed$31db2bc0$95918340$@com> Message-ID: Yeah, I don't know how to interpret them either - I've seen the tests recommended a few times, but never with much explanation of what to look for. Our users noticed an improvement, be interesting to hear if yours do too. Cheers, Fraser On 24 November 2011 16:08, Stephen Dame wrote: > Fraser, thanks for tips in compiling kernel.**** > > ** ** > > I was able to set up a 1000hz ec2 instance ... and run some comparative > tests using the same starting 10.04 LTS ami.**** > > ** ** > > Here are the results of timing tests.. Not sure how to interpret? > Having users test audio right now in voice conferences to see if they > notice any difference.**** > > ** ** > > Looked in wiki? I?m still not clear on what to look for.**** > > ** ** > > Regards,**** > > Stephen**** > > ** ** > > *100hz kernel* > > ** ** > > *time_test 600 10* > > test 1 sleep 600 8847**** > > test 2 sleep 600 9969**** > > test 3 sleep 600 10006**** > > test 4 sleep 600 9986**** > > test 5 sleep 600 10003**** > > test 6 sleep 600 9977**** > > test 7 sleep 600 10117**** > > test 8 sleep 600 9872**** > > test 9 sleep 600 10014**** > > test 10 sleep 600 10124**** > > ** ** > > *timer_test 120 10* > > 2011-11-24 14:53:51.142568 [CONSOLE] mod_commands.c:329 Timer Test: 1 > sleep 120 120289**** > > 2011-11-24 14:53:51.263610 [CONSOLE] mod_commands.c:329 Timer Test: 2 > sleep 120 120703**** > > 2011-11-24 14:53:51.382587 [CONSOLE] mod_commands.c:329 Timer Test: 3 > sleep 120 118952**** > > 2011-11-24 14:53:51.502640 [CONSOLE] mod_commands.c:329 Timer Test: 4 > sleep 120 120048**** > > 2011-11-24 14:53:51.622851 [CONSOLE] mod_commands.c:329 Timer Test: 5 > sleep 120 120213**** > > 2011-11-24 14:53:51.742593 [CONSOLE] mod_commands.c:329 Timer Test: 6 > sleep 120 119681**** > > 2011-11-24 14:53:51.862583 [CONSOLE] mod_commands.c:329 Timer Test: 7 > sleep 120 119963**** > > 2011-11-24 14:53:51.982568 [CONSOLE] mod_commands.c:329 Timer Test: 8 > sleep 120 119975**** > > 2011-11-24 14:53:52.133931 [CONSOLE] mod_commands.c:329 Timer Test: 9 > sleep 120 151338**** > > 2011-11-24 14:53:52.232591 [CONSOLE] mod_commands.c:329 Timer Test: 10 > sleep 120 98753**** > > ** ** > > ** ** > > *1000hz kernel* > > ** ** > > *time_test 600 10* > > test 1 sleep 600 1591**** > > test 2 sleep 600 1009**** > > test 3 sleep 600 971**** > > test 4 sleep 600 1114**** > > test 5 sleep 600 888**** > > test 6 sleep 600 996**** > > test 7 sleep 600 998**** > > test 8 sleep 600 988**** > > test 9 sleep 600 994**** > > test 10 sleep 600 996**** > > avg 1054**** > > ** ** > > *timer_test 120 10***** > > 2011-11-24 19:55:27.198564 [CONSOLE] mod_commands.c:562 Timer Test: 1 > sleep 120 119956**** > > 2011-11-24 19:55:27.318559 [CONSOLE] mod_commands.c:562 Timer Test: 2 > sleep 120 119994**** > > 2011-11-24 19:55:27.438559 [CONSOLE] mod_commands.c:562 Timer Test: 3 > sleep 120 119939**** > > 2011-11-24 19:55:27.558559 [CONSOLE] mod_commands.c:562 Timer Test: 4 > sleep 120 119965**** > > 2011-11-24 19:55:27.678559 [CONSOLE] mod_commands.c:562 Timer Test: 5 > sleep 120 119967**** > > 2011-11-24 19:55:27.798563 [CONSOLE] mod_commands.c:562 Timer Test: 6 > sleep 120 119977**** > > 2011-11-24 19:55:27.918559 [CONSOLE] mod_commands.c:562 Timer Test: 7 > sleep 120 120000**** > > 2011-11-24 19:55:28.038559 [CONSOLE] mod_commands.c:562 Timer Test: 8 > sleep 120 119936**** > > 2011-11-24 19:55:28.158559 [CONSOLE] mod_commands.c:562 Timer Test: 9 > sleep 120 119973**** > > 2011-11-24 19:55:28.278558 [CONSOLE] mod_commands.c:562 Timer Test: 10 > sleep 120 119968**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Fraser > Redmond > > *Sent:* Saturday, November 19, 2011 7:43 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Timing issues in AWS?**** > > ** ** > > Hi Stephen,**** > > ** ** > > I spent most of last week working out how to upgrade the kernel timer from > 100HZ to 1000HZ on ubuntu on AWS. I documented the steps I took here:**** > > ** ** > > http://wiki.freeswitch.org/wiki/Amazon_EC2#Updating_Kernel_Timer_to_1000HZ > **** > > ** ** > > Making that change gave us a noticeable jump in call quality.**** > > > Cheers, > Fraser > > > > **** > > On 17 November 2011 08:49, Stephen Dame wrote:**** > > I?m running freeswitch on about 40 different m1.small c1.medium?s in AWS > regions, us-east, us-west, eu-west, and asia?. They are used in videoconf > component of BigBlueButton.org . For most part they work great? There are > occasional issues with voip quality but the app is 100% voip, with the BBB > client all browser based. So the conference is subject to every ones local > network connections and most issues ?blamed? on the internet instead of > freeswitch J We also tie in skype and DID direct to improve latency for > some clients. But expectations are set so we meet them.**** > > **** > > You probably have tougher business conferencing clients that want perfect > audio. But these are production deployed and generate revenue. All > these are on Ubuntu 10.04 official amis with no mods to kernels. **** > > **** > > I?m sure there are tweaks that can be made, and bare metal solutions that > would work a little better? Does anyone have any ideas how to optimize a > Ubuntu instance. I would engage in a few hours of consulting is so.**** > > **** > > The tests below are interesting but above my paygrade to understand what > they mean? **** > > **** > > I?m running them but don?t have a clue how to interpret. If you want a > simple of real running data, would be glad to run sample tests on these > distributed servers and provide back for analysis.**** > > **** > > Regards,**** > > Stephen**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Picher > *Sent:* Thursday, November 17, 2011 8:03 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Timing issues in AWS?**** > > **** > > Not having anywhere near the same trouble in a Xen Server in my lab... > > Although test_time is skewed 180 degrees from what it was in AWS.... > > freeswitch at internal> timer_test 120 10**** > > Avg: 119.945ms Total Time: 1199.727ms > > 2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: > samplecount after init: 1 > 2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: > samplecount after first step: 2 > 2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 > sleep 120 120993 > 2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 > sleep 120 118914 > 2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 > sleep 120 119919 > 2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 > sleep 120 120032 > 2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 > sleep 120 119802 > 2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 > sleep 120 119999 > 2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 > sleep 120 119975 > 2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 > sleep 120 120003 > 2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 > sleep 120 119849 > 2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 > sleep 120 119971**** > > > freeswitch at internal> time_test 600 10**** > > test 1 sleep 600 1 > test 2 sleep 600 1 > test 3 sleep 600 0 > test 4 sleep 600 1 > test 5 sleep 600 0 > test 6 sleep 600 1 > test 7 sleep 600 1 > test 8 sleep 600 0 > test 9 sleep 600 0 > test 10 sleep 600 1 > avg 0**** > > > [root at openuc bin]# uname -r**** > > 2.6.18-274.7.1.el5**** > > > [root at openuc bin]# grep CONFIG_HZ /boot/config-***** > > /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 > /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y > /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000**** > > > Thoughts as to why AWS results are so different from XenServer? Other > than not knowing who else is on the AWS box? > > Thanks, > Mike > > **** > > On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher > wrote:**** > > m1.large > > I have a c1.xlarge queued up and ready to test...**** > > **** > > On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen > wrote:**** > > Just a simple question, what kind of AWS instance are you running your > FreeSWITCH?**** > > It makes huge difference.**** > > Thanks,**** > > Chris**** > > On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher > wrote:**** > > Hi guys, > > Trying to get to the bottom of some conference bridge issues I'm having > with running the system in AWS. > > We're hearing a bunch of snap-crackle-pops in conference bridges and when > I tcpdum on the server itself I see them in the RTP and see RTP timestamp > problems. > > I've run the following: > > > timer_test**** > > freeswitch at 127.0.0.1@internal> timer_test 120 10**** > > Avg: 120.004ms Total Time: 1200.315ms**** > > **** > > 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: > samplecount after init: 1 > freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] > mod_commands.c:466 Timer Test: samplecount after first step: 2 > 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 > sleep 120 120006 > 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 > sleep 120 120040 > 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 > sleep 120 119976 > 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 > sleep 120 120007 > 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 > sleep 120 120001 > 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 > sleep 120 120031 > 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 > sleep 120 119996 > 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 > sleep 120 119994 > 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 > sleep 120 120005 > 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 > sleep 120 119991**** > > > test_time**** > > freeswitch at 127.0.0.1@internal> time_test 600 10**** > > > test 1 sleep 600 1592 > test 2 sleep 600 986 > test 3 sleep 600 1018 > test 4 sleep 600 980 > test 5 sleep 600 1005 > test 6 sleep 600 1000 > test 7 sleep 600 972 > test 8 sleep 600 990 > test 9 sleep 600 1006 > test 10 sleep 600 994 > avg 1054**** > > > For kernel:**** > > [root at openuc bin]# uname -r > > 2.6.21.7-2.fc8xen**** > > > CONFIG_HZ:**** > > [root at openuc bin]# grep CONFIG_HZ /boot/config-* > > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set > /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set > /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y > /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set > /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y > /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000**** > > are the xenU kernel settings screwing me here? > > Thanks, > Mike > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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**** > > > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info**** > > > > > -- > There are 10 kinds of people in this world, those who understand binary > and those who don't. > > mpicher at gmail.com > blog: http://www.sipxecs.info > call: sip:mpicher at sipxecs.info**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111124/ce84408f/attachment-0001.html From ahe.sanath at gmail.com Fri Nov 25 05:24:53 2011 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Fri, 25 Nov 2011 07:54:53 +0530 Subject: [Freeswitch-users] playAndGetDigits problem in LUA program Message-ID: Hi all, I used following to get phone number from caller in LUA program. function get_dest() new_number = session:playAndGetDigits(9, 11, 3, 3000, "#", lang_path .. "enter_dest_number.wav", lang_path .. "error.wav", "\\d+"); session:say(new_number, "en", "number", "pronounced"); session:setVariable("dest_numbers",new_number); end -- main -- freeswitch.consoleLog("info", "Voicemail - select_main_menu:\n\r"); reeswitch.consoleLog("info", "Voicemail - Select English :\n\r"); get_dest(); .. .. if caller hang off the phone without entering number(dtmf), I can see lot of same log lines in the freeswitch console log. (around 50000 lines in 1 or 2 sec.Pls see below). *is it bug or just need to ignore. Pls advice. * Br, Sanath 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111125/68b956e3/attachment.html From cliff at develix.com Fri Nov 25 06:59:36 2011 From: cliff at develix.com (Cliff Wells) Date: Thu, 24 Nov 2011 19:59:36 -0800 Subject: [Freeswitch-users] High load on database server In-Reply-To: <1322083386335-7026203.post@n2.nabble.com> References: <1322083386335-7026203.post@n2.nabble.com> Message-ID: <1322193576.2192.24.camel@portable-evil> On Wed, 2011-11-23 at 13:23 -0800, peely wrote: > I had exactly the same problem with PGSQL and FS, moved to MySQL for the core > DB and everything is fine. > > PGSQL's transaction logging is pretty heavyweight for the amount of > transactions you get from FS, but if you are really needing to use PostGres > I would look at using unlogged tables. > > As I say though, MySQL runs like a charm, I've had it at well over 300CPS > with 500 registrations/sec. The OP was using MySQL 5.5. Further, his call load is what I would term "very light". His issue is unlikely to be solved by changing databases. Other people's issues appear to be related to ODBC drivers rather than PostgreSQL itself (my own were solved by upgrading to unixODBC 3.0). Unless you are using MyISAM, MySQL and PostgreSQL aren't drastically different in performance these days. Cliff From cliff at develix.com Fri Nov 25 07:07:57 2011 From: cliff at develix.com (Cliff Wells) Date: Thu, 24 Nov 2011 20:07:57 -0800 Subject: [Freeswitch-users] High load on database server In-Reply-To: <1322193576.2192.24.camel@portable-evil> References: <1322083386335-7026203.post@n2.nabble.com> <1322193576.2192.24.camel@portable-evil> Message-ID: <1322194077.2192.25.camel@portable-evil> On Thu, 2011-11-24 at 19:59 -0800, Cliff Wells wrote: > PostgreSQL itself (my own were solved by upgrading to unixODBC 3.0). I meant unixODBC 2.3. Cliff From ocset at the800group.com Fri Nov 25 12:52:39 2011 From: ocset at the800group.com (ocset) Date: Fri, 25 Nov 2011 17:52:39 +0800 Subject: [Freeswitch-users] Not sure of the correct terminology Message-ID: <4ECF6567.3090501@the800group.com> Hi I am not sure what I should call this topic but I am trying to find out about using Freeswitch in an environment where the caller is prompted with a list of options like "if you know the callers extension, enter it now" etc. Is this possible? What is the correct term for it and where would I find more info on it? Thanks From avi at avimarcus.net Fri Nov 25 13:01:53 2011 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 25 Nov 2011 12:01:53 +0200 Subject: [Freeswitch-users] Not sure of the correct terminology In-Reply-To: <4ECF6567.3090501@the800group.com> References: <4ECF6567.3090501@the800group.com> Message-ID: You want an IVR: http://wiki.freeswitch.org/wiki/IVR_Menu If you specifically want some one to be able to search for a name, you can use: http://wiki.freeswitch.org/wiki/Mod_directory -Avi On Fri, Nov 25, 2011 at 11:52 AM, ocset wrote: > Hi > > I am not sure what I should call this topic but I am trying to find out > about using Freeswitch in an environment where the caller is prompted > with a list of options like "if you know the callers extension, enter it > now" etc. > > Is this possible? What is the correct term for it and where would I find > more info on it? > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111125/59de4941/attachment.html From ocset at the800group.com Fri Nov 25 13:27:48 2011 From: ocset at the800group.com (ocset) Date: Fri, 25 Nov 2011 18:27:48 +0800 Subject: [Freeswitch-users] Not sure of the correct terminology In-Reply-To: References: <4ECF6567.3090501@the800group.com> Message-ID: <4ECF6DA4.5000201@the800group.com> Thanks Avi On 11/25/2011 06:01 PM, Avi Marcus wrote: > You want an IVR: http://wiki.freeswitch.org/wiki/IVR_Menu > > If you specifically want some one to be able to search for a name, you > can use: http://wiki.freeswitch.org/wiki/Mod_directory > > > -Avi > > > On Fri, Nov 25, 2011 at 11:52 AM, ocset > wrote: > > Hi > > I am not sure what I should call this topic but I am trying to > find out > about using Freeswitch in an environment where the caller is prompted > with a list of options like "if you know the callers extension, > enter it > now" etc. > > Is this possible? What is the correct term for it and where would > I find > more info on it? > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111125/c5be18e3/attachment-0001.html From davidwaf at gmail.com Fri Nov 25 13:44:56 2011 From: davidwaf at gmail.com (David Wafula) Date: Fri, 25 Nov 2011 12:44:56 +0200 Subject: [Freeswitch-users] g++: Internal error: Segmentation fault (program cc1plus Message-ID: Am on CentOS 6.0, on linode.com. I did 'make current' few minutes ago, and ran into this: Please help ? mv -f .deps/pcregrep.Tpo .deps/pcregrep.Po /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o pcregrep pcregrep.o libpcreposix.la libtool: link: gcc -g -O2 -o pcregrep pcregrep.o ./.libs/libpcreposix.a /usr/local/src/freeswitch/libs/pcre/.libs/libpcre.a g++ -DHAVE_CONFIG_H -I. -g -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See for instructions. make[3]: *** [pcrecpp_unittest.o] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/libs/pcre' make[2]: *** [all] Error 2 make[2]: Leaving directory `/usr/local/src/freeswitch/libs/pcre' make[1]: *** [libs/pcre/libpcre.la] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111125/4c7f72ee/attachment.html From peter.olsson at visionutveckling.se Fri Nov 25 14:01:48 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 25 Nov 2011 12:01:48 +0100 Subject: [Freeswitch-users] g++: Internal error: Segmentation fault (program cc1plus In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B2768B696@cooper> Make sure you've not set ulimit -s... I had a similar issue once, and after exiting the shell, and logging in again (with default limit's) everything worked. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r David Wafula Skickat: den 25 november 2011 11:45 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] g++: Internal error: Segmentation fault (program cc1plus Am on CentOS 6.0, on linode.com. I did 'make current' few minutes ago, and ran into this: Please help ? mv -f .deps/pcregrep.Tpo .deps/pcregrep.Po /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o pcregrep pcregrep.o libpcreposix.la libtool: link: gcc -g -O2 -o pcregrep pcregrep.o ./.libs/libpcreposix.a /usr/local/src/freeswitch/libs/pcre/.libs/libpcre.a g++ -DHAVE_CONFIG_H -I. -g -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See for instructions. make[3]: *** [pcrecpp_unittest.o] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/libs/pcre' make[2]: *** [all] Error 2 make[2]: Leaving directory `/usr/local/src/freeswitch/libs/pcre' make[1]: *** [libs/pcre/libpcre.la] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 -- David Wafula !DSPAM:4ecf721c32761750112969! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111125/f876f326/attachment.html From davidwaf at gmail.com Fri Nov 25 15:38:38 2011 From: davidwaf at gmail.com (David Wafula) Date: Fri, 25 Nov 2011 14:38:38 +0200 Subject: [Freeswitch-users] g++: Internal error: Segmentation fault (program cc1plus In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B2768B696@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B2768B696@cooper> Message-ID: Thanks Peter. I had indeed set ulimit -s on the shell. It's building fine now. david On Fri, Nov 25, 2011 at 1:01 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Make sure you?ve not set ulimit ?s... I had a similar issue once, and > after exiting the shell, and logging in again (with default limit?s) > everything worked.**** > > ** ** > > /Peter**** > > ** ** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *David Wafula > *Skickat:* den 25 november 2011 11:45 > *Till:* FreeSWITCH Users Help > *?mne:* [Freeswitch-users] g++: Internal error: Segmentation fault > (program cc1plus**** > > ** ** > > Am on CentOS 6.0, on linode.com. I did 'make current' few minutes ago, > and ran into this:**** > > ** ** > > Please help ?**** > > ** ** > > mv -f .deps/pcregrep.Tpo .deps/pcregrep.Po**** > > /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o pcregrep > pcregrep.o libpcreposix.la **** > > libtool: link: gcc -g -O2 -o pcregrep pcregrep.o ./.libs/libpcreposix.a > /usr/local/src/freeswitch/libs/pcre/.libs/libpcre.a**** > > g++ -DHAVE_CONFIG_H -I. -g -O2 -MT pcrecpp_unittest.o -MD -MP -MF > .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc*** > * > > g++: Internal error: Segmentation fault (program cc1plus)**** > > Please submit a full bug report.**** > > See for instructions.**** > > make[3]: *** [pcrecpp_unittest.o] Error 1**** > > make[3]: Leaving directory `/usr/local/src/freeswitch/libs/pcre'**** > > make[2]: *** [all] Error 2**** > > make[2]: Leaving directory `/usr/local/src/freeswitch/libs/pcre'**** > > make[1]: *** [libs/pcre/libpcre.la] Error 2**** > > make[1]: Leaving directory `/usr/local/src/freeswitch'**** > > make: *** [current] Error 2**** > > ** ** > > ** ** > > -- > David Wafula > !DSPAM:4ecf721c32761750112969! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111125/cfa17f90/attachment.html From dm03514 at gmail.com Thu Nov 24 17:53:10 2011 From: dm03514 at gmail.com (daniel mican) Date: Thu, 24 Nov 2011 09:53:10 -0500 Subject: [Freeswitch-users] Help Making a Call through External Sip Provider Message-ID: Greetings, I am having difficulty making external phone calls through freeswitch. I successfully receive calls but cannot sned them. I believe the problem is somewhere in here, I am not resolving contact with my sip gateway. It just keeps trying to invite until it times out. I have double checked all config files, and spent time reading the fs wiki's in hope of find what I am doing wrong. Below is the log and a packet. I have been fumbling with this for days and any help at all would be greatly appreciated. Ty 2011-11-24 09:35:13.274794 [DEBUG] sofia.c:5366 Channel sofia/external/+14102944410 entering state [calling][0] nua: nua_handle_magic: entering nta: timer A fired, retransmit INVITE (20736592) tport_release(0x7f7b2c052c00): 0x7f7b2c10b940 by 0x7f7b2c074b10 with (nil) tport_tsend(0x7f7b2c052c00) tpn = UDP/75.101.162.138:5060 tport_resolve addrinfo = 75.101.162.138:5060 tport_by_addrinfo(0x7f7b2c052c00): not found by name UDP/75.101.162.138:5060 tport_vsend(0x7f7b2c052c00): 1089 bytes of 1089 to udp/75.101.162.138:5060 tport_vsend returned 1089 nta: resent INVITE (20736592) to UDP/75.101.162.138:5060 tport_pend(0x7f7b2c052c00): pending 0x7f7b2c10b940 for udp/10.0.0.4:5080(already 0) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/1 term, 0/2 free nta: timer set next to 2000 ms nta: timer A fired, retransmit INVITE (20736592) tport_release(0x7f7b2c052c00): 0x7f7b2c10b940 by 0x7f7b2c074b10 with (nil) tport_tsend(0x7f7b2c052c00) tpn = UDP/75.101.162.138:5060 tport_resolve addrinfo = 75.101.162.138:5060 tport_by_addrinfo(0x7f7b2c052c00): not found by name UDP/75.101.162.138:5060 tport_vsend(0x7f7b2c052c00): 1089 bytes of 1089 to udp/75.101.162.138:5060 tport_vsend returned 1089 nta: resent INVITE (20736592) to UDP/75.101.162.138:5060 tport_pend(0x7f7b2c052c00): pending 0x7f7b2c10b940 for udp/10.0.0.4:5080(already 0) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/1 term, 0/2 free nta: timer set next to 4000 ms nta: timer A fired, retransmit INVITE (20736592) tport_release(0x7f7b2c052c00): 0x7f7b2c10b940 by 0x7f7b2c074b10 with (nil) tport_tsend(0x7f7b2c052c00) tpn = UDP/75.101.162.138:5060 tport_resolve addrinfo = 75.101.162.138:5060 tport_by_addrinfo(0x7f7b2c052c00): not found by name UDP/75.101.162.138:5060 tport_vsend(0x7f7b2c052c00): 1089 bytes of 1089 to udp/75.101.162.138:5060 tport_vsend returned 1089 nta: resent INVITE (20736592) to UDP/75.101.162.138:5060 tport_pend(0x7f7b2c052c00): pending 0x7f7b2c10b940 for udp/10.0.0.4:5080(already 0) 09:35:28.281385 IP (tos 0x0, ttl 64, id 12974, offset 0, flags [none], proto UDP (17), length 1117) 10.0.0.4.5080 > 75.101.162.138.5060: [udp sum ok] UDP, length 1089 E..]2... at .K. ...Ke.......I..INVITE sip:+14102944410 at sip1.teltechsys.com SIP/2.0 Via: SIP/2.0/UDP 68.48.6.208:5080;rport;branch=z9hG4bKZ2Q3Km0KNKUcK Max-Forwards: 70 From: "" ;tag=3pmg8NZt4SHNQ To: Call-ID: 5c4496eb-914c-122f-9d94-0025646e9c00 CSeq: 20736592 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 13-52-28 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-Carrier: sip.flowroute.com X-Carrier-Prefix: 74278017 X-FS-Support: update_display Remote-Party-ID: ;party=calling;screen=yes;privacy=off -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111124/064accf6/attachment-0001.html From Giovanni.Visciano at italtel.it Fri Nov 25 09:27:17 2011 From: Giovanni.Visciano at italtel.it (Visciano Giovanni) Date: Fri, 25 Nov 2011 07:27:17 +0100 Subject: [Freeswitch-users] Sofia late-negotiation on re-INVITE (codec-modification) References: Message-ID: Basic call SIP vs SIP with: This help to avoid FS transcoding, the SIP endpoints negotiate codec end-to-end. FS still in the media path but no transcoding applied. INVITE [pcma,g729] ---> FS ---> INVITE [pcma,g729] 200 OK [pcma] <--- FS ---> 200 OK [pcma] If a SIP ep send a re-INVITE for codec modification, sofia negotiate the new codec on that leg without "asking" the remote party. This result in transcoding and since I have mod_g729 only in passthrough mode the call is released by the core. re-INVITE [g729] ---> FS 200 OK [g729] <--- FS ************* It seems that "late_negotiation + inherit_codec" doesn't apply to a codec modification. Do I miss some configuration options in the SIP profile? (Note: I cannot use proxy-media/no-media mode in my configuration) ************* Regards Giovanni Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111125/34309778/attachment.html From brian at freeswitch.org Sat Nov 26 01:39:39 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 25 Nov 2011 16:39:39 -0600 Subject: [Freeswitch-users] Sofia late-negotiation on re-INVITE (codec-modification) In-Reply-To: References: Message-ID: <3E991E54-B2A4-4B97-B74D-3B0EC2227B29@freeswitch.org> You do realize that those param's are sofia profile params and not dialplan XML right? /b On Nov 25, 2011, at 12:27 AM, Visciano Giovanni wrote: > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111125/3942ee1a/attachment.html From nbhatti at gmail.com Sat Nov 26 08:42:07 2011 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Sat, 26 Nov 2011 10:42:07 +0500 Subject: [Freeswitch-users] control variables in xml_curl Message-ID: Hello, I am using xml_curl to fetch directory from the DB. When a request comes in, I can see the following variables being sent by FS, (along with a few others) [user] => '298192' [domain] => '192.168.169.128' [ip] => '192.168.169.1' This works, fine, I can pull the user directory from the DB and user is able to register fine. The issue is that if I have two profiles, both running on same IP but on different ports, they both are able to register. Theoretically each profile should handle it's own settings/users. I don't want profile A's user to register on profile B. If I am able to send the port on which the user is trying to register (Profile port), It solves the problem. Can I control the variables sent in by FS to xml_curl for fetching user directory? Or is there any other way to fix the issue? -B From nbhatti at gmail.com Sat Nov 26 09:49:16 2011 From: nbhatti at gmail.com (nbhatti) Date: Fri, 25 Nov 2011 22:49:16 -0800 (PST) Subject: [Freeswitch-users] control variables in xml_curl In-Reply-To: References: Message-ID: <1322290156339-7033167.post@n2.nabble.com> Well to be exact, sip_request_port is not being shown. All I can see is, http://pastebin.freeswitch.org/17876 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/control-variables-in-xml-curl-tp7033104p7033167.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mpicher at gmail.com Sat Nov 26 18:06:41 2011 From: mpicher at gmail.com (Michael Picher) Date: Sat, 26 Nov 2011 10:06:41 -0500 Subject: [Freeswitch-users] Timing issues in AWS? In-Reply-To: References: <005501cca52f$aa0286c0$fe079440$@com> <021b01ccaaed$31db2bc0$95918340$@com> Message-ID: I managed to get my own CentOS 5.7 box built and loaded up as an AMI (for those interested http://sipxecs.blogspot.com/2011/11/building-centos-57-images-for-amazon.html ). Kernel timing is 1000 Hz but still can't get acceptable performance on a m1.large instance for conference bridge (running sipXecs 4.4.0 latest patch). There is of course a lot of extra stuff going on with sipXecs beyond FS... so it may be something else sucking down CPU. Maybe I'll try moving conf bridge onto its own virtual in a cluster and see what happens... I am able to get acceptable (unloaded) bridge performance with a c1.xlarge. For the price of a c1.xlarge though you may as well purchase a 1/3 of a rack and load up your own servers. Mike On Thu, Nov 24, 2011 at 7:10 PM, Fraser Redmond wrote: > Yeah, I don't know how to interpret them either - I've seen the tests > recommended a few times, but never with much explanation of what to look > for. > > Our users noticed an improvement, be interesting to hear if yours do too. > > Cheers, > Fraser > > > > > > On 24 November 2011 16:08, Stephen Dame wrote: > >> Fraser, thanks for tips in compiling kernel.**** >> >> ** ** >> >> I was able to set up a 1000hz ec2 instance ... and run some comparative >> tests using the same starting 10.04 LTS ami.**** >> >> ** ** >> >> Here are the results of timing tests.. Not sure how to interpret? >> Having users test audio right now in voice conferences to see if they >> notice any difference.**** >> >> ** ** >> >> Looked in wiki? I?m still not clear on what to look for.**** >> >> ** ** >> >> Regards,**** >> >> Stephen**** >> >> ** ** >> >> *100hz kernel* >> >> ** ** >> >> *time_test 600 10* >> >> test 1 sleep 600 8847**** >> >> test 2 sleep 600 9969**** >> >> test 3 sleep 600 10006**** >> >> test 4 sleep 600 9986**** >> >> test 5 sleep 600 10003**** >> >> test 6 sleep 600 9977**** >> >> test 7 sleep 600 10117**** >> >> test 8 sleep 600 9872**** >> >> test 9 sleep 600 10014**** >> >> test 10 sleep 600 10124**** >> >> ** ** >> >> *timer_test 120 10* >> >> 2011-11-24 14:53:51.142568 [CONSOLE] mod_commands.c:329 Timer Test: 1 >> sleep 120 120289**** >> >> 2011-11-24 14:53:51.263610 [CONSOLE] mod_commands.c:329 Timer Test: 2 >> sleep 120 120703**** >> >> 2011-11-24 14:53:51.382587 [CONSOLE] mod_commands.c:329 Timer Test: 3 >> sleep 120 118952**** >> >> 2011-11-24 14:53:51.502640 [CONSOLE] mod_commands.c:329 Timer Test: 4 >> sleep 120 120048**** >> >> 2011-11-24 14:53:51.622851 [CONSOLE] mod_commands.c:329 Timer Test: 5 >> sleep 120 120213**** >> >> 2011-11-24 14:53:51.742593 [CONSOLE] mod_commands.c:329 Timer Test: 6 >> sleep 120 119681**** >> >> 2011-11-24 14:53:51.862583 [CONSOLE] mod_commands.c:329 Timer Test: 7 >> sleep 120 119963**** >> >> 2011-11-24 14:53:51.982568 [CONSOLE] mod_commands.c:329 Timer Test: 8 >> sleep 120 119975**** >> >> 2011-11-24 14:53:52.133931 [CONSOLE] mod_commands.c:329 Timer Test: 9 >> sleep 120 151338**** >> >> 2011-11-24 14:53:52.232591 [CONSOLE] mod_commands.c:329 Timer Test: 10 >> sleep 120 98753**** >> >> ** ** >> >> ** ** >> >> *1000hz kernel* >> >> ** ** >> >> *time_test 600 10* >> >> test 1 sleep 600 1591**** >> >> test 2 sleep 600 1009**** >> >> test 3 sleep 600 971**** >> >> test 4 sleep 600 1114**** >> >> test 5 sleep 600 888**** >> >> test 6 sleep 600 996**** >> >> test 7 sleep 600 998**** >> >> test 8 sleep 600 988**** >> >> test 9 sleep 600 994**** >> >> test 10 sleep 600 996**** >> >> avg 1054**** >> >> ** ** >> >> *timer_test 120 10***** >> >> 2011-11-24 19:55:27.198564 [CONSOLE] mod_commands.c:562 Timer Test: 1 >> sleep 120 119956**** >> >> 2011-11-24 19:55:27.318559 [CONSOLE] mod_commands.c:562 Timer Test: 2 >> sleep 120 119994**** >> >> 2011-11-24 19:55:27.438559 [CONSOLE] mod_commands.c:562 Timer Test: 3 >> sleep 120 119939**** >> >> 2011-11-24 19:55:27.558559 [CONSOLE] mod_commands.c:562 Timer Test: 4 >> sleep 120 119965**** >> >> 2011-11-24 19:55:27.678559 [CONSOLE] mod_commands.c:562 Timer Test: 5 >> sleep 120 119967**** >> >> 2011-11-24 19:55:27.798563 [CONSOLE] mod_commands.c:562 Timer Test: 6 >> sleep 120 119977**** >> >> 2011-11-24 19:55:27.918559 [CONSOLE] mod_commands.c:562 Timer Test: 7 >> sleep 120 120000**** >> >> 2011-11-24 19:55:28.038559 [CONSOLE] mod_commands.c:562 Timer Test: 8 >> sleep 120 119936**** >> >> 2011-11-24 19:55:28.158559 [CONSOLE] mod_commands.c:562 Timer Test: 9 >> sleep 120 119973**** >> >> 2011-11-24 19:55:28.278558 [CONSOLE] mod_commands.c:562 Timer Test: 10 >> sleep 120 119968**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Fraser >> Redmond >> >> *Sent:* Saturday, November 19, 2011 7:43 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Timing issues in AWS?**** >> >> ** ** >> >> Hi Stephen,**** >> >> ** ** >> >> I spent most of last week working out how to upgrade the kernel timer >> from 100HZ to 1000HZ on ubuntu on AWS. I documented the steps I took here: >> **** >> >> ** ** >> >> http://wiki.freeswitch.org/wiki/Amazon_EC2#Updating_Kernel_Timer_to_1000HZ >> **** >> >> ** ** >> >> Making that change gave us a noticeable jump in call quality.**** >> >> >> Cheers, >> Fraser >> >> >> >> **** >> >> On 17 November 2011 08:49, Stephen Dame wrote:**** >> >> I?m running freeswitch on about 40 different m1.small c1.medium?s in AWS >> regions, us-east, us-west, eu-west, and asia?. They are used in videoconf >> component of BigBlueButton.org . For most part they work great? There are >> occasional issues with voip quality but the app is 100% voip, with the BBB >> client all browser based. So the conference is subject to every ones local >> network connections and most issues ?blamed? on the internet instead of >> freeswitch J We also tie in skype and DID direct to improve latency for >> some clients. But expectations are set so we meet them.**** >> >> **** >> >> You probably have tougher business conferencing clients that want perfect >> audio. But these are production deployed and generate revenue. All >> these are on Ubuntu 10.04 official amis with no mods to kernels. **** >> >> **** >> >> I?m sure there are tweaks that can be made, and bare metal solutions that >> would work a little better? Does anyone have any ideas how to optimize a >> Ubuntu instance. I would engage in a few hours of consulting is so.**** >> >> **** >> >> The tests below are interesting but above my paygrade to understand what >> they mean? **** >> >> **** >> >> I?m running them but don?t have a clue how to interpret. If you want a >> simple of real running data, would be glad to run sample tests on these >> distributed servers and provide back for analysis.**** >> >> **** >> >> Regards,**** >> >> Stephen**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael >> Picher >> *Sent:* Thursday, November 17, 2011 8:03 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Timing issues in AWS?**** >> >> **** >> >> Not having anywhere near the same trouble in a Xen Server in my lab... >> >> Although test_time is skewed 180 degrees from what it was in AWS.... >> >> freeswitch at internal> timer_test 120 10**** >> >> Avg: 119.945ms Total Time: 1199.727ms >> >> 2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: >> samplecount after init: 1 >> 2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: >> samplecount after first step: 2 >> 2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 >> sleep 120 120993 >> 2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 >> sleep 120 118914 >> 2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 >> sleep 120 119919 >> 2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 >> sleep 120 120032 >> 2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 >> sleep 120 119802 >> 2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 >> sleep 120 119999 >> 2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 >> sleep 120 119975 >> 2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 >> sleep 120 120003 >> 2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 >> sleep 120 119849 >> 2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 >> sleep 120 119971**** >> >> >> freeswitch at internal> time_test 600 10**** >> >> test 1 sleep 600 1 >> test 2 sleep 600 1 >> test 3 sleep 600 0 >> test 4 sleep 600 1 >> test 5 sleep 600 0 >> test 6 sleep 600 1 >> test 7 sleep 600 1 >> test 8 sleep 600 0 >> test 9 sleep 600 0 >> test 10 sleep 600 1 >> avg 0**** >> >> >> [root at openuc bin]# uname -r**** >> >> 2.6.18-274.7.1.el5**** >> >> >> [root at openuc bin]# grep CONFIG_HZ /boot/config-***** >> >> /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y >> /boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 >> /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y >> /boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 >> /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y >> /boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 >> /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y >> /boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 >> /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y >> /boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000**** >> >> >> Thoughts as to why AWS results are so different from XenServer? Other >> than not knowing who else is on the AWS box? >> >> Thanks, >> Mike >> >> **** >> >> On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher >> wrote:**** >> >> m1.large >> >> I have a c1.xlarge queued up and ready to test...**** >> >> **** >> >> On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen >> wrote:**** >> >> Just a simple question, what kind of AWS instance are you running your >> FreeSWITCH?**** >> >> It makes huge difference.**** >> >> Thanks,**** >> >> Chris**** >> >> On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher >> wrote:**** >> >> Hi guys, >> >> Trying to get to the bottom of some conference bridge issues I'm having >> with running the system in AWS. >> >> We're hearing a bunch of snap-crackle-pops in conference bridges and when >> I tcpdum on the server itself I see them in the RTP and see RTP timestamp >> problems. >> >> I've run the following: >> >> >> timer_test**** >> >> freeswitch at 127.0.0.1@internal> timer_test 120 10**** >> >> Avg: 120.004ms Total Time: 1200.315ms**** >> >> **** >> >> 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: >> samplecount after init: 1 >> freeswitch at 127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] >> mod_commands.c:466 Timer Test: samplecount after first step: 2 >> 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 >> sleep 120 120006 >> 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 >> sleep 120 120040 >> 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 >> sleep 120 119976 >> 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 >> sleep 120 120007 >> 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 >> sleep 120 120001 >> 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 >> sleep 120 120031 >> 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 >> sleep 120 119996 >> 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 >> sleep 120 119994 >> 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 >> sleep 120 120005 >> 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 >> sleep 120 119991**** >> >> >> test_time**** >> >> freeswitch at 127.0.0.1@internal> time_test 600 10**** >> >> >> test 1 sleep 600 1592 >> test 2 sleep 600 986 >> test 3 sleep 600 1018 >> test 4 sleep 600 980 >> test 5 sleep 600 1005 >> test 6 sleep 600 1000 >> test 7 sleep 600 972 >> test 8 sleep 600 990 >> test 9 sleep 600 1006 >> test 10 sleep 600 994 >> avg 1054**** >> >> >> For kernel:**** >> >> [root at openuc bin]# uname -r >> >> 2.6.21.7-2.fc8xen**** >> >> >> CONFIG_HZ:**** >> >> [root at openuc bin]# grep CONFIG_HZ /boot/config-* >> >> /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y >> /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set >> /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set >> /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 >> /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set >> /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set >> /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y >> /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 >> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set >> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set >> /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set >> /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y >> /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000**** >> >> are the xenU kernel settings screwing me here? >> >> Thanks, >> Mike >> >> >> -- >> There are 10 kinds of people in this world, those who understand binary >> and those who don't. >> >> mpicher at gmail.com >> blog: http://www.sipxecs.info >> call: sip:mpicher at sipxecs.info**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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**** >> >> **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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**** >> >> >> >> >> -- >> There are 10 kinds of people in this world, those who understand binary >> and those who don't. >> >> mpicher at gmail.com >> blog: http://www.sipxecs.info >> call: sip:mpicher at sipxecs.info**** >> >> >> >> >> -- >> There are 10 kinds of people in this world, those who understand binary >> and those who don't. >> >> mpicher at gmail.com >> blog: http://www.sipxecs.info >> call: sip:mpicher at sipxecs.info**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher at gmail.com blog: http://www.sipxecs.info call: sip:mpicher at sipxecs.info -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111126/2b1c928a/attachment-0001.html From avi at avimarcus.net Sat Nov 26 18:40:05 2011 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 26 Nov 2011 17:40:05 +0200 Subject: [Freeswitch-users] Sofia late-negotiation on re-INVITE (codec-modification) In-Reply-To: <3E991E54-B2A4-4B97-B74D-3B0EC2227B29@freeswitch.org> References: <3E991E54-B2A4-4B97-B74D-3B0EC2227B29@freeswitch.org> Message-ID: Visciano: they have to be sofia profile params, because otherwise, the codec is already negotiated by the time it gets to the dialplan. -Avi On Sat, Nov 26, 2011 at 12:39 AM, Brian West wrote: > You do realize that those param's are sofia profile params and not > dialplan XML right? > > /b > > On Nov 25, 2011, at 12:27 AM, Visciano Giovanni wrote: > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111126/41cf2743/attachment.html From davidwaf at gmail.com Sat Nov 26 21:45:12 2011 From: davidwaf at gmail.com (David Wafula) Date: Sat, 26 Nov 2011 20:45:12 +0200 Subject: [Freeswitch-users] capture not working on freeswitch.org? Message-ID: I wanted to create an account on freeswitch.org, because i need to use pastebin. That is how to get the credentials for pastebin, right? But i am unable register there, the capture is not working. regards, -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111126/0d766c08/attachment.html From lloyd.aloysius at gmail.com Sat Nov 26 21:53:43 2011 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Sat, 26 Nov 2011 13:53:43 -0500 Subject: [Freeswitch-users] capture not working on freeswitch.org? In-Reply-To: References: Message-ID: David, for http://pastebin.freeswitch.org/ ... Please read the login screen message. The user name and password in the login screen. Thanks Lloyd On Sat, Nov 26, 2011 at 1:45 PM, David Wafula wrote: > I wanted to create an account on freeswitch.org, because i need to use > pastebin. That is how to get the credentials for pastebin, right? But i am > unable register there, the capture is not working. > > regards, > -- > David Wafula > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111126/5aa9e509/attachment.html From brian at freeswitch.org Sat Nov 26 22:33:29 2011 From: brian at freeswitch.org (Brian West) Date: Sat, 26 Nov 2011 13:33:29 -0600 Subject: [Freeswitch-users] capture not working on freeswitch.org? In-Reply-To: References: Message-ID: <12B8C135-3FFD-437F-8182-A1DB676CCE51@freeswitch.org> You gave away the test answer. :P Thats one of the things I use to gauge if someone pays attention! /b On Nov 26, 2011, at 12:53 PM, Lloyd Aloysius wrote: > David, > > for http://pastebin.freeswitch.org/ ... Please read the login screen > message. The user name and password in the login screen. > > > > Thanks > Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111126/70172780/attachment.html From curriegrad2004 at gmail.com Sat Nov 26 22:36:35 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 26 Nov 2011 11:36:35 -0800 Subject: [Freeswitch-users] capture not working on freeswitch.org? In-Reply-To: <12B8C135-3FFD-437F-8182-A1DB676CCE51@freeswitch.org> References: <12B8C135-3FFD-437F-8182-A1DB676CCE51@freeswitch.org> Message-ID: Or to see if they are using a browser that isn't broken On Sat, Nov 26, 2011 at 11:33 AM, Brian West wrote: > You gave away the test answer. ?:P ?Thats one of the things I use to gauge > if someone pays attention! > /b > On Nov 26, 2011, at 12:53 PM, Lloyd Aloysius wrote: > > David, > > for?http://pastebin.freeswitch.org/??... Please read the login screen > message. The user name and password in the login screen. > > > > Thanks > Lloyd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 davidwaf at gmail.com Sat Nov 26 22:42:50 2011 From: davidwaf at gmail.com (David Wafula) Date: Sat, 26 Nov 2011 21:42:50 +0200 Subject: [Freeswitch-users] capture not working on freeswitch.org? In-Reply-To: <12B8C135-3FFD-437F-8182-A1DB676CCE51@freeswitch.org> References: <12B8C135-3FFD-437F-8182-A1DB676CCE51@freeswitch.org> Message-ID: On Sat, Nov 26, 2011 at 9:33 PM, Brian West wrote: > You gave away the test answer. :P Thats one of the things I use to gauge > if someone pays attention! > > > That was a good one :) ...am in ! -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111126/eba589a9/attachment.html From brian at freeswitch.org Sat Nov 26 23:05:43 2011 From: brian at freeswitch.org (Brian West) Date: Sat, 26 Nov 2011 14:05:43 -0600 Subject: [Freeswitch-users] capture not working on freeswitch.org? In-Reply-To: References: <12B8C135-3FFD-437F-8182-A1DB676CCE51@freeswitch.org> Message-ID: <3D9076B6-2454-43E4-B89B-63376B4298E7@freeswitch.org> Sigh... Broken technology is only the fault of the user for putting up with it. (But I can totally understand that sometimes you kinda have too!) /b On Nov 26, 2011, at 1:36 PM, curriegrad2004 wrote: > Or to see if they are using a browser that isn't broken > > On Sat, Nov 26, 2011 at 11:33 AM, Brian West wrote: >> You gave away the test answer. :P Thats one of the things I use to gauge >> if someone pays attention! >> /b >> On Nov 26, 2011, at 12:53 PM, Lloyd Aloysius wrote: >> >> David, >> >> for http://pastebin.freeswitch.org/ ... Please read the login screen >> message. The user name and password in the login screen. >> >> >> >> Thanks >> Lloyd >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111126/7278d221/attachment-0001.html From jesse at mactechs.com Sun Nov 27 03:47:48 2011 From: jesse at mactechs.com (Jesse Peterson) Date: Sat, 26 Nov 2011 16:47:48 -0800 Subject: [Freeswitch-users] No shared modules built, linking problems on NetBSD Message-ID: <9AC01FAC-5F9B-42CF-A1E2-DD9789FBE03C@mactechs.com> Hello, I'm having some problems building FreeSWITCH head from Git on NetBSD. There are some known NetBSD issue in the source that need to be corrected for but once corrected then the compilation happens successfully. However once installed FreeSWITCH has only installed .a and .la files into /usr/local/freeswitch/mod/ instead of .so files. Thus when FreeSWITCH starts it cannot load any of the modules because there's no dynamic libraries. I don't know if it's related but when building modules and then linking modules many of them have: *** Warning: This system can not link to static lib archive libs/apr-util/libaprutil-1.la. *** I have the capability to make that library automatically link in when *** you link to this library. But I can only do this if you have a *** shared version of the library, which you do not appear to have. *** Warning: This system can not link to static lib archive /home/local-src/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la. *** I have the capability to make that library automatically link in when *** you link to this library. But I can only do this if you have a *** shared version of the library, which you do not appear to have. *** Warning: Trying to link with static lib archive libs/libedit/src/.libs/libedit.a. *** I have the capability to make that library automatically link in when *** you link to this library. But I can only do this if you have a *** shared version of the library, which you do not appear to have *** because the file extensions .a of this argument makes me believe *** that it is just a static archive that I should not use here. These vary by the specific module. This system is NetBSD/i386 5.1 with LibTool 2.2.6b with autoconf 2.68 and automake 1.11.1 (all from pkgsrc). It appears there was one shared library built: ./libs/libfreeswitch.so so it seems it *can* build shared libraries, just none of the modules perhaps? Any ideas or help would be appreciated. Thanks! - Jesse From ahe.sanath at gmail.com Sun Nov 27 06:02:04 2011 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Sun, 27 Nov 2011 08:32:04 +0530 Subject: [Freeswitch-users] playAndGetDigits problem in LUA program In-Reply-To: References: Message-ID: Hi all, I used following to get phone number from caller in LUA program. function get_dest() new_number = session:playAndGetDigits(9, 11, 3, 3000, "#", lang_path .. "enter_dest_number.wav", lang_path .. "error.wav", "\\d+"); session:say(new_number, "en", "number", "pronounced"); session:setVariable("dest_numbers",new_number); end -- main -- freeswitch.consoleLog("info", "Voicemail - select_main_menu:\n\r"); reeswitch.consoleLog("info", "Voicemail - Select English :\n\r"); get_dest(); .. .. if caller hang off the phone without entering number(dtmf), I can see lot of same log lines in the freeswitch console log. (around 50000 lines in 1 or 2 sec.Pls see below). *is it bug or just need to ignore. Pls advice. * Br, Sanath 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111127/9eb2789d/attachment.html From Giovanni.Visciano at italtel.it Sat Nov 26 09:54:53 2011 From: Giovanni.Visciano at italtel.it (Visciano Giovanni) Date: Sat, 26 Nov 2011 07:54:53 +0100 Subject: [Freeswitch-users] Sofia late-negotiation on re-INVITE(codec-modification) References: <3E991E54-B2A4-4B97-B74D-3B0EC2227B29@freeswitch.org> Message-ID: > You do realize that those param's are sofia profile params and not dialplan XML right? Hi Brian, sure I know that. So: In sofia SIP profile I have: and in my dialplan XML I hit: I repeated my test scenario, sorry what I described was with inherit_codec=false. If I set inherit_codec=true (this is my purpose) sofia behavior is this: INVITE [pcma,g729] ---> FS ---> INVITE [pcma,g729] 200 OK [pcma] <--- FS ---> 200 OK [pcma] re-INVITE [g729] ---> FS Reinvite Codec Error! <--- FS Here the call is release and from the log I see that sofia just negotiatiate localy and compare the codec list in the new offer only with PCMA. With inherit_codec=false it compare the codec list with the codec loaded in the core so it match G729 (but damn it's a passthrought codec and I cannot do transcoding!). Here is an extract off the log, that show the error. Now I do not have with me the complete session log, I will send it tomorrow. Do you think it's a bug? 1970-01-01 23:10:28.600056 [DEBUG] switch_core_session.c:857 Send signal sofia/external/abcd at xyz.com [BREAK] 1970-01-01 23:10:28.600056 [DEBUG] switch_core_session.c:857 Send signal sofia/external/abcd at xyz.com [BREAK] 1970-01-01 23:10:28.620894 [DEBUG] sofia.c:5143 Channel sofia/external/abcd at xyz.com entering state [received][100] 1970-01-01 23:10:28.620894 [DEBUG] sofia.c:5154 Remote SDP: v=0 o=- 0 1 IN IP4 10.50.185.16 s=IMSS c=IN IP4 10.50.185.16 t=0 0 m=audio 10992 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 1970-01-01 23:10:28.620894 [DEBUG] sofia_glue.c:4731 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 1970-01-01 23:10:28.620894 [DEBUG] sofia_glue.c:4731 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 1970-01-01 23:10:28.620894 [DEBUG] sofia_glue.c:4845 Set 2833 dtmf send/recv payload to 101 1970-01-01 23:10:28.620894 [ERR] sofia.c:5611 Reinvite Codec Error! 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This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 3935 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111126/c7faca2e/attachment-0001.bin From josuputa at yahoo.es Sun Nov 27 15:51:13 2011 From: josuputa at yahoo.es (JoZu) Date: Sun, 27 Nov 2011 04:51:13 -0800 (PST) Subject: [Freeswitch-users] Skypeopen dont send caller audio Message-ID: <1322398273500-7036124.post@n2.nabble.com> Hi, i have a little problem i have FS installed, skypeopen_oss client and skypeopen.ko oss sound driver installed. I configure in public a DID number bridge to skypopen/RR/skype22, the call going from DID to SKYPE1 and call SKYPE22, here is the LOG: http://pastebin.freeswitch.org/17877 http://pastebin.freeswitch.org/17877 All work OK, but the sound from DID caller to Skype22 dont reach, no sound, Skype22 to DID caller works ok. Questions: Skype_oss and the skypeopen.ko require a sound card to work? What is the problem? Codecs? Please helpme, thanks and sorry my english. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skypeopen-dont-send-caller-audio-tp7036124p7036124.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/20111127/1ef424a9/attachment-0001.html From josuputa at yahoo.es Sun Nov 27 16:00:58 2011 From: josuputa at yahoo.es (asdfa asdfasdf) Date: Sun, 27 Nov 2011 13:00:58 +0000 (GMT) Subject: [Freeswitch-users] Skypopen audio problems Message-ID: <1322398858.57178.YahooMailNeo@web27607.mail.ukl.yahoo.com> Hi, i have a little problem i have FS installed, skypeopen_oss client and skypeopen.ko oss sound driver installed. I configure in public a DID number bridge to skypopen/RR/skype22, the call going from DID to SKYPE1 and call SKYPE22, here is the LOG: http://pastebin.freeswitch.org/17877 All work OK, but the sound from DID caller to Skype22 dont reach, no sound, Skype22 to DID caller works ok. Questions: Skype_oss and the skypeopen.ko require a sound card to work? What is the problem? Codecs? Please helpme, thanks and sorry my english. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111127/c4fad312/attachment-0001.html From henrikaagaardsorensen at gmail.com Sun Nov 27 18:52:19 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Sun, 27 Nov 2011 16:52:19 +0100 Subject: [Freeswitch-users] Mod_cdr_csv, no cdr_csv.conf.xml in standard installation. Message-ID: I've installed a very basic installation of FreeSwitch, following this guide: http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide Everything works fine, I also get CDR files in my freeswitch/log/cdr-csv directory. But I cannot find the file cdr_csv.conf.xml in the conf/ directory (anywhere, including autoload_configs/). Reading this wiki-page: http://wiki.freeswitch.org/wiki/Mod_cdr_csv says that it requires the cdr_csv.conf.xml file. Does FreeSwitch have a standard Mod_cdr_csv settings, which are used in case the conf-file is missing or...? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111127/fa3dcaeb/attachment.html From gmaruzz at gmail.com Sun Nov 27 19:27:00 2011 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 27 Nov 2011 17:27:00 +0100 Subject: [Freeswitch-users] Skypopen audio problems In-Reply-To: <1322398858.57178.YahooMailNeo@web27607.mail.ukl.yahoo.com> References: <1322398858.57178.YahooMailNeo@web27607.mail.ukl.yahoo.com> Message-ID: No, skypopen does not need a soundcard. >From your log nothing strange in skypopen. Let me recap if I have understood: 1) you have a SIP DID 2) this SIP DID is going to an extension that bridges to a skypopen outbound call to SKYPE_22 3) the SIP incoming call is correctly bridged to SKYPE_22 4) DID caller can hear SKYPE_22 talking 5) SKYPE_22 do not hear DID caller talking. This is the problem. I have understood correctly? Can you post the skypopen.conf.xml and the dialplan snippet that do the bridge? -giovanni On Sun, Nov 27, 2011 at 2:00 PM, asdfa asdfasdf wrote: > Hi, i have a little problem i have FS installed, skypeopen_oss client and > skypeopen.ko oss sound driver installed. > I configure in public a DID number bridge to skypopen/RR/skype22, the call > going from DID to SKYPE1 and call SKYPE22, here is the LOG: > http://pastebin.freeswitch.org/17877 > All work OK, but the sound from DID caller to Skype22 dont reach, no sound, > Skype22 to DID caller works ok. Questions: > Skype_oss and the skypeopen.ko require a sound card to work? > What is the problem? Codecs? > Please helpme, thanks and sorry my english. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 philq at qsystemsengineering.com Sun Nov 27 21:10:01 2011 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Sun, 27 Nov 2011 13:10:01 -0500 Subject: [Freeswitch-users] Sofia late-negotiation on re-INVITE (codec-modification) Message-ID: <00ad01ccad2f$cae6c010$60b44030$@com> I think I might be experiencing a similar problem here... A friend of mine dialed into one of my conference rooms with only the GSM codec enabled on his mobile SIP client (Linphone). Despite the fact that I offered GSM as one of the codecs, FS only appeared to offer G711u and the call failed. When he set G711u as an option on his end, he was able to connect. Late negotiation is enabled. Some relevant settings: internal.xml: vars.xml: SIP transaction info follows: ------------------------------------------------------------------------ recv 765 bytes from udp/[68.47.xx.xx]:5060 at 06:59:23.381436: ------------------------------------------------------------------------ INVITE sip:3030 at myswitch.org:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.15:5060;rport;branch=z9hG4bK1252688933 From: "resist0r" ;tag=539248952 To: Call-ID: 1476867094 CSeq: 20 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.4.3 (eXosip2/3.1.0) Subject: Phone call Expires: 120 Content-Length: 234 v=0 o=resist0r 1856 1856 IN IP4 68.47.xx.xx s=Talk c=IN IP4 68.47.xx.xx t=0 0 m=audio 9003 RTP/AVP 0 0 3 101 a=rtpmap:0 GSM/22050 a=rtpmap:0 GSM/11025 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 ------------------------------------------------------------------------ send 354 bytes to udp/[68.47.xx.xx]:5060 at 06:59:23.381660: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.15:5060;rport=5060;branch=z9hG4bK1252688933;received=68.47.xx.xx From: "resist0r" ;tag=539248952 To: Call-ID: 1476867094 CSeq: 20 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f2cf68b 2011-11-20 18-40-41 -0500 Content-Length: 0 ------------------------------------------------------------------------ 2011-11-25 01:59:23.365421 [NOTICE] switch_channel.c:920 New Channel sofia/external/resist0r at 68.47.xx.xx [b9d2afd7-e0ba-4168-acee-29ccd2dc4791] 2011-11-25 01:59:23.365421 [DEBUG] sofia.c:5356 Channel sofia/external/resist0r at 68.47.xx.xx entering state [received][100] 2011-11-25 01:59:23.365421 [DEBUG] sofia.c:5367 Remote SDP: v=0 o=resist0r 1856 1856 IN IP4 68.47.xx.xx s=Talk c=IN IP4 68.47.xx.xx t=0 0 m=audio 9003 RTP/AVP 0 0 3 101 a=rtpmap:0 GSM/11025 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[PCMU:0:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[PCMU:0:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[PCMA:8:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[GSM:3:8000:20:13200] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[iLBC:97:8000:30:13330] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:2869 Set Codec sofia/external/resist0r at 68.47.xx.xx PCMU/8000 20 ms 160 samples 64000 bits Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111127/7150e722/attachment.html From paul at cupis.co.uk Sun Nov 27 22:43:38 2011 From: paul at cupis.co.uk (Paul Cupis) Date: Sun, 27 Nov 2011 19:43:38 +0000 Subject: [Freeswitch-users] Sofia late-negotiation on re-INVITE (codec-modification) In-Reply-To: <00ad01ccad2f$cae6c010$60b44030$@com> References: <00ad01ccad2f$cae6c010$60b44030$@com> Message-ID: <4ED292EA.5040501@cupis.co.uk> On 27/11/11 18:10, Phil Quesinberry wrote: > A friend of mine dialed into one of my conference rooms with only the GSM > codec enabled on his mobile SIP client (Linphone). Despite the fact that I > offered GSM as one of the codecs, FS only appeared to offer G711u and the > call failed. When he set G711u as an option on his end, he was able to > connect. Late negotiation is enabled. The SDP from the mobile SIP client is bogus, it offers codec 0 (PCMU) but tries to redefine it as GSM. FreeSWITCH accepts the offer of PCMU and negotiates it - but then the callers client sends/expects invalid media: m=audio 9003 RTP/AVP 0 0 3 101 a=rtpmap:0 GSM/22050 a=rtpmap:0 GSM/11025 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 http://www.iana.org/assignments/rtp-parameters 0 PCMU/8000 3 GSM/8000 It should be using 96-126 for GSM at other bitrates... Regards, From ayobami at programmer.net Sun Nov 27 23:10:15 2011 From: ayobami at programmer.net (ayobami) Date: Sun, 27 Nov 2011 12:10:15 -0800 (PST) Subject: [Freeswitch-users] Soft phones could not register to my Freeswitch instance Message-ID: <1322424615934-7036985.post@n2.nabble.com> I have my Freeswitch box connected to some other computer through wireless, FS automatically detects the IP address of the wireless adapter and things work fine every soft phone could register and calls were routed appropriately. But when I connected the Freeswitch box to a switch with the other computers many issues came up. At first freeswitch does not bind to the ethernet adapter, it indicates that its binding to the loopback IP address 127.0.0.1, so I tried restarting several times, no luck, so I decided to do some tweaking in Vars.xml I changed this to and in the internal.xml in sip_profile folder I changed these two settings from default values to so after all these changes, FS now binds to the ethernet adapter. But the problem now is no softphone on other computers could register again, they are all giving error 408 which means they could not register, I have googled and looked for solutions, but couldnot find any, please people help me out, I am really frustrated -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Soft-phones-could-not-register-to-my-Freeswitch-instance-tp7036985p7036985.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ayobami at programmer.net Sun Nov 27 23:57:09 2011 From: ayobami at programmer.net (ayobami) Date: Sun, 27 Nov 2011 12:57:09 -0800 (PST) Subject: [Freeswitch-users] Soft phones could not register to my Freeswitch instance Message-ID: <1322427429430-7037047.post@n2.nabble.com> I have my Freeswitch box connected to some other computer through wireless, FS automatically detects the IP address of the wireless adapter and things work fine every soft phone could register and calls were routed appropriately. But when I connected the Freeswitch box to a switch with the other computers many issues came up. At first freeswitch does not bind to the ethernet adapter, it indicates that its binding to the loopback IP address 127.0.0.1, so I tried restarting several times, no luck, so I decided to do some tweaking in Vars.xml I changed this to and in the internal.xml in sip_profile folder I changed these two settings from default values to so after all these changes, FS now binds to the ethernet adapter. But the problem now is no softphone on other computers could register again, they are all giving error 408 which means they could not register, I have googled and looked for solutions, but couldnot find any -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Soft-phones-could-not-register-to-my-Freeswitch-instance-tp7037047p7037047.html Sent from the freeswitch-users mailing list archive at Nabble.com. From valery.kalinin at gmail.com Mon Nov 28 04:45:08 2011 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Mon, 28 Nov 2011 07:45:08 +0600 Subject: [Freeswitch-users] FreeTDM does not work Message-ID: I upgraded freeswitch from git. Nothing changed in the configuration! Suddenly stopped working freetdm. Why? my configs: freetdm.conf [span zt pri] name => pri trunk_type => E1 group => e1group b-channel => 1-15 d-channel => 16 b-channel => 17-31 wanpipe.conf [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 freetdm.conf.xml log: 2011-11-27 14:35:12.284837 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 3 2011-11-27 14:35:12.284847 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 4 2011-11-27 14:35:12.284898 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 1 2011-11-27 14:35:12.284906 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 2 2011-11-27 14:35:12.285290 [WARNING] ftmod_zt.c:333 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 1 fd 50 (Invalid argument)] 2011-11-27 14:35:12.285523 [WARNING] ftmod_zt.c:333 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 3 fd 50 (Invalid argument)] 2011-11-27 14:35:12.285634 [ERR] mod_freetdm.c:2465 Error finding FreeTDM span id:1 name:pri Card Digium TE121 Please HELP! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/9fadfd9f/attachment.html From trever.adams at gmail.com Mon Nov 28 05:55:00 2011 From: trever.adams at gmail.com (Trever L. Adams) Date: Sun, 27 Nov 2011 19:55:00 -0700 Subject: [Freeswitch-users] FreeTDM does not work In-Reply-To: References: Message-ID: <4ED2F804.1020707@gmail.com> On 11/27/2011 06:45 PM, Valery Kalinin wrote: > I upgraded freeswitch from git. > Nothing changed in the configuration! > Suddenly stopped working freetdm. > Why? Just out of curiosity, did you do a reboot and get a kernel change? I had this happen to me and it was because only part of the dahdi stuff (which I recompile on every reboot to catch this now) got compiled and things broke that way. If not, sorry for the noise. Trever -- "As soon as you know ogg vorbis, you know that mp3 sucks!" -- Unknown -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111127/4672f070/attachment.bin From valery.kalinin at gmail.com Mon Nov 28 08:55:17 2011 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Mon, 28 Nov 2011 11:55:17 +0600 Subject: [Freeswitch-users] FreeTDM does not work Message-ID: > Just out of curiosity, did you do a reboot and get a kernel change? I > had this happen to me and it was because only part of the dahdi stuff > (which I recompile on every reboot to catch this now) got compiled and > things broke that way. I did not understand a bit. You offer to recompile dahdi? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/6e3d2ba5/attachment.html From trever.adams at gmail.com Mon Nov 28 13:40:01 2011 From: trever.adams at gmail.com (Trever L. Adams) Date: Mon, 28 Nov 2011 03:40:01 -0700 Subject: [Freeswitch-users] [SPAM] Re: FreeTDM does not work In-Reply-To: References: Message-ID: <4ED36501.5080009@gmail.com> On 11/27/2011 10:55 PM, Valery Kalinin wrote: > > Just out of curiosity, did you do a reboot and get a kernel change? I > > had this happen to me and it was because only part of the dahdi stuff > > (which I recompile on every reboot to catch this now) got compiled and > > things broke that way. > > I did not understand a bit. > You offer to recompile dahdi? > I had a problem similar to what you described. My problem ended up being an incompatibility between a kernel I had installed and rebooted with and dahdi, not with FreeSwitch. There was also another time where I failed to start dahdi correctly. If this doesn't help, I am sorry for the waste. Trever -- "Those willing to give up a little liberty for a little security deserve neither security nor liberty." -- Benjamin Franklin -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/bebacdfd/attachment.bin From mchlmll at gmail.com Mon Nov 28 16:57:46 2011 From: mchlmll at gmail.com (Michele M) Date: Mon, 28 Nov 2011 14:57:46 +0100 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 Message-ID: Hi there, it is my intention to buy 100 g729 sw codecs from FS to setup a mee -me conference system.Which are the minimal hw requirements to support 100 conference legs where all the codecs implied are G729 ? ( meaning no transcoding from pcmu->G729 and viceversa). Thx in advance Michele -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/7f495531/attachment.html From curriegrad2004 at gmail.com Mon Nov 28 18:16:05 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 28 Nov 2011 07:16:05 -0800 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: References: Message-ID: In this case there is no need to purchase any G729 software licenses at all if you are doing all of this in passthrough mode (meaning both ends are g729 codecs and FS is just simply passing them through). The default G729 codec that is included with FS is passthrough. You could also look into this: http://wiki.freeswitch.org/wiki/Proxy_Media or this http://wiki.freeswitch.org/wiki/Bypass_Media On Mon, Nov 28, 2011 at 5:57 AM, Michele M wrote: > Hi there, > > it is my intention to buy 100 g729 sw codecs from FS to setup a mee -me > conference?system.Which are the minimal hw requirements to support 100 > conference legs where all the codecs implied?are G729 ? ( meaning no > transcoding from pcmu->G729 and viceversa). > > Thx in advance > > Michele > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 curriegrad2004 at gmail.com Mon Nov 28 18:17:47 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 28 Nov 2011 07:17:47 -0800 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: References: Message-ID: Whoops, I didn't read the question throughly... Yeah, for about 100 calls or so, I'd personally just get a hardware transcoder card instead of purchasing 100 software g729 licenses to do the job with the conference calls. iirc, g729 is quite CPU and resource demanding. On Mon, Nov 28, 2011 at 7:16 AM, curriegrad2004 wrote: > In this case there is no need to purchase any G729 software licenses > at all if you are doing all of this in passthrough mode (meaning both > ends are g729 codecs and FS is just simply passing them through). The > default G729 codec that is included with FS is passthrough. > > You could also look into this: > http://wiki.freeswitch.org/wiki/Proxy_Media or this > http://wiki.freeswitch.org/wiki/Bypass_Media > > On Mon, Nov 28, 2011 at 5:57 AM, Michele M wrote: >> Hi there, >> >> it is my intention to buy 100 g729 sw codecs from FS to setup a mee -me >> conference?system.Which are the minimal hw requirements to support 100 >> conference legs where all the codecs implied?are G729 ? ( meaning no >> transcoding from pcmu->G729 and viceversa). >> >> Thx in advance >> >> Michele >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 Nabble_01394 at slickdeals.endjunk.com Mon Nov 28 18:22:07 2011 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Mon, 28 Nov 2011 07:22:07 -0800 (PST) Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: References: Message-ID: <1322493727673-7039072.post@n2.nabble.com> Michele M wrote > ( meaning no transcoding from pcmu->G729 and viceversa). If you don't do a transcoding, then perhaps you can just use the free G729 pass-thru CoDec and no need to pay for any licenses. ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/MInimal-Hardware-requirements-G729-tp7038829p7039072.html Sent from the freeswitch-users mailing list archive at Nabble.com. From curriegrad2004 at gmail.com Mon Nov 28 18:36:32 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 28 Nov 2011 07:36:32 -0800 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: <1322493727673-7039072.post@n2.nabble.com> References: <1322493727673-7039072.post@n2.nabble.com> Message-ID: mazilo: that's a little bit on the tricky side to do it with conference calls... I just realized it when I started to drink my first coffee of the day lol... On Mon, Nov 28, 2011 at 7:22 AM, mazilo wrote: > > Michele M wrote >> ( meaning no transcoding from pcmu->G729 and viceversa). > If you don't do a transcoding, then perhaps you can just use the free G729 > pass-thru CoDec and no need to pay for any licenses. > > ----- > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/MInimal-Hardware-requirements-G729-tp7038829p7039072.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 Mon Nov 28 19:05:34 2011 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Nov 2011 10:05:34 -0600 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: References: <1322493727673-7039072.post@n2.nabble.com> Message-ID: <6AF0D454-D126-46BA-80BA-CBEC25D78F3B@freeswitch.org> So sugar isn't the only thing in the coffee! :P /b On Nov 28, 2011, at 9:36 AM, curriegrad2004 wrote: > mazilo: that's a little bit on the tricky side to do it with > conference calls... I just realized it when I started to drink my > first coffee of the day lol... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/70469381/attachment.html From gustavomarsico at gmail.com Sun Nov 27 23:20:09 2011 From: gustavomarsico at gmail.com (=?iso-8859-1?Q?Gustavo_M=E1rsico?=) Date: Sun, 27 Nov 2011 17:20:09 -0300 Subject: [Freeswitch-users] Soft phones could not register to my Freeswitch instance In-Reply-To: <1322424615934-7036985.post@n2.nabble.com> References: <1322424615934-7036985.post@n2.nabble.com> Message-ID: <28D54CA1-1F1C-48CA-9716-2C9AADC39B40@gmail.com> Don't know if 0.0.0.0 is valid, but it should be. On Nov 27, 2011, at 5:10 PM, ayobami wrote: > I have my Freeswitch box connected to some other computer through wireless, > FS automatically detects the IP address of the wireless adapter and things > work fine every soft phone could register and calls were routed > appropriately. But when I connected the Freeswitch box to a switch with the > other computers many issues came up. At first freeswitch does not bind to > the ethernet adapter, it indicates that its binding to the loopback IP > address 127.0.0.1, so I tried restarting several times, no luck, so I > decided to do some tweaking in Vars.xml > I changed this to > > and in the internal.xml in sip_profile folder > I changed these two settings from default values to > > > > so after all these changes, FS now binds to the ethernet adapter. > But the problem now is no softphone on other computers could register again, > they are all giving error 408 which means they could not register, I have > googled and looked for solutions, but couldnot find any, please people help > me out, I am really frustrated > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Soft-phones-could-not-register-to-my-Freeswitch-instance-tp7036985p7036985.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 josuputa at yahoo.es Sun Nov 27 20:12:53 2011 From: josuputa at yahoo.es (JoZu) Date: Sun, 27 Nov 2011 09:12:53 -0800 (PST) Subject: [Freeswitch-users] Skypopen audio problems In-Reply-To: References: <1322398858.57178.YahooMailNeo@web27607.mail.ukl.yahoo.com> Message-ID: <1322413960.23145.YahooMailNeo@web27606.mail.ukl.yahoo.com> ? Thanks Giovanni for you respose, you understandme perfectly. A frew aditional information: My SIP (DID) provider does not requiere a password,? and the configuration does not require any specific parameter. I put SIP/5001 at xxxxx.xxxxx.net:5080 at provider mapping. In Freeswitch: I made change only in two files: acl.conf.xml - for include in domains list the providers IP'S public.xml - for include this: ????? ????? ?????? ????? ? <--- skype user ????? ??? skypopen.conf.xml, dont change anything at this moment the file are same as source. ________________________________ De: Giovanni Maruzzelli-2 [via freeswitch-users] Para: JoZu Enviado: domingo 27 de noviembre de 2011 17:31 Asunto: Re: Skypopen audio problems No, skypopen does not need a soundcard. >From your log nothing strange in skypopen. Let me recap if I have understood: 1) you have a SIP DID 2) this SIP DID is going to an extension that bridges to a skypopen outbound call to SKYPE_22 3) the SIP incoming call is correctly bridged to SKYPE_22 4) DID caller can hear SKYPE_22 talking 5) SKYPE_22 do not hear DID caller talking. This is the problem. I have understood correctly? Can you post the skypopen.conf.xml and the dialplan snippet that do the bridge? -giovanni On Sun, Nov 27, 2011 at 2:00 PM, asdfa asdfasdf <[hidden email]> wrote: > Hi, i have a little problem i have FS installed, skypeopen_oss client and > skypeopen.ko oss sound driver installed. > I configure in public a DID number bridge to skypopen/RR/skype22, the call > going from DID to SKYPE1 and call SKYPE22, here is the LOG: > http://pastebin.freeswitch.org/17877 > All work OK, but the sound from DID caller to Skype22 dont reach, no sound, > Skype22 to DID caller works ok. Questions: > Skype_oss and the skypeopen.ko require a sound card to work? > What is the problem? Codecs? > Please helpme, thanks and sorry my english. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: [hidden email] http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 ________________________________ If you reply to this email, your message will be added to the discussion below:http://freeswitch-users.2379917.n2.nabble.com/Skypopen-audio-problems-tp7036246p7036606.html To start a new topic under freeswitch-users, email ml-node+s2379917n2379917h85 at n2.nabble.com To unsubscribe from freeswitch-users, click here. NAML -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skypopen-audio-problems-tp7036246p7036703.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/20111127/ec799ad6/attachment.html From brian at freeswitch.org Mon Nov 28 19:26:32 2011 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Nov 2011 10:26:32 -0600 Subject: [Freeswitch-users] Soft phones could not register to my Freeswitch instance In-Reply-To: <28D54CA1-1F1C-48CA-9716-2C9AADC39B40@gmail.com> References: <1322424615934-7036985.post@n2.nabble.com> <28D54CA1-1F1C-48CA-9716-2C9AADC39B40@gmail.com> Message-ID: 0.0.0.0 is NOT valid. We won't bind to INADDR_ANY go read the spec it has some obscure details about this! :P /b On Nov 27, 2011, at 2:20 PM, Gustavo M?rsico wrote: > Don't know if 0.0.0.0 is valid, but it should be. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/c2d76cc1/attachment-0001.html From krice at freeswitch.org Mon Nov 28 19:30:22 2011 From: krice at freeswitch.org (Ken Rice) Date: Mon, 28 Nov 2011 10:30:22 -0600 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: Message-ID: Keep in Mind that Mod_conference (and anyone elses conference software for that matter) take everything to slinear for mixing. Unfortunately you cant mix encoded auto with encoded audio very well. So far 100 Channels in a conf bridge, he would require 100 licenses. K On 11/28/11 9:16 AM, "curriegrad2004" wrote: > In this case there is no need to purchase any G729 software licenses at all if > you are doing all of this in passthrough mode (meaning both ends are g729 > codecs and FS is just simply passing them through). The default G729 codec > that is included with FS is passthrough. You could also look into > this: http://wiki.freeswitch.org/wiki/Proxy_Media or > this http://wiki.freeswitch.org/wiki/Bypass_Media >> On Mon, Nov 28, 2011 at 5:57 AM, Michele M wrote: >> Hi there, it is my intention to buy 100 g729 sw codecs from FS to setup a >> meet-me conference?system.Which are the minimal hw requirements to support >> 100 conference legs where all the codecs implied?are G729 ? ( meaning no >> transcoding from pcmu->G729 and viceversa). > > Thx in advance > > Michele > From jeff at jefflenk.com Mon Nov 28 21:20:31 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 28 Nov 2011 10:20:31 -0800 (PST) Subject: [Freeswitch-users] Help Making a Call through External Sip Provider In-Reply-To: References: Message-ID: <1322504431411-7039775.post@n2.nabble.com> You did not provide enough information here, do this and attach the resulting log with the combined debug and siptrace. fsctl loglevel 7 sofia profile global siptrace on -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-Making-a-Call-through-External-Sip-Provider-tp7032525p7039775.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at jefflenk.com Mon Nov 28 21:56:22 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 28 Nov 2011 10:56:22 -0800 (PST) Subject: [Freeswitch-users] Mod_cdr_csv, no cdr_csv.conf.xml in standard installation. In-Reply-To: References: Message-ID: <1322506582708-7039930.post@n2.nabble.com> The file is definitely there: conf\autoload_configs -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Mod-cdr-csv-no-cdr-csv-conf-xml-in-standard-installation-tp7036532p7039930.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jkomar at jbox.ca Mon Nov 28 21:11:55 2011 From: jkomar at jbox.ca (Komar, Jason) Date: Mon, 28 Nov 2011 11:11:55 -0700 Subject: [Freeswitch-users] FXS doesn't stop ringing Message-ID: Hello Everyone, My current setup uses a Sangoma A400 card with 2 FXO and 2 FXS ports. I am working toward having my analog lines ported over to SIP trunks, but in the meantime I am working with what I have. When a call comes in over the POTS, my public dialplan transfers the call to extension 2000 which is setup in my default dialplan. This extension is setup like the Local_Extension example, but instead of bridging with this line: I bridge to my FXS port: It seems to work well, but the issue I have is that the FXS port keeps ringing the analog phone even after the call bridges to voicemail. I also tested doing a call in and hangup after 1 ring and the FXS keeps ringing and does not detect the hangup. I appreciate any help that someone can offer. Thanks, Jason Komar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/b8219d97/attachment.html From trever.adams at gmail.com Mon Nov 28 23:02:09 2011 From: trever.adams at gmail.com (Trever L. Adams) Date: Mon, 28 Nov 2011 13:02:09 -0700 Subject: [Freeswitch-users] [SPAM] FXS doesn't stop ringing In-Reply-To: References: Message-ID: <4ED3E8C1.40603@gmail.com> On 11/28/2011 11:11 AM, Komar, Jason wrote: > Hello Everyone, > > My current setup uses a Sangoma A400 card with 2 FXO and 2 FXS ports. > I am working toward having my analog lines ported over to SIP trunks, > but in the meantime I am working with what I have. > > When a call comes in over the POTS, my public dialplan transfers the > call to extension 2000 which is setup in my default dialplan. This > extension is setup like the Local_Extension example, but instead of > bridging with this line: > > data="user/${dialed_extension}@${domain_name}"/> > > I bridge to my FXS port: > > > > It seems to work well, but the issue I have is that the FXS port keeps > ringing the analog phone even after the call bridges to voicemail. I > also tested doing a call in and hangup after 1 ring and the FXS keeps > ringing and does not detect the hangup. > > I appreciate any help that someone can offer. > > Thanks, > Jason Komar > Please, feel free to add to my bug: http://jira.freeswitch.org/browse/OPENZAP-173 I have been unable to find a solution. Trever -- "Ultimately, it comes down to taste... it's comes down to trying to expose yourself to the best things that humans have done, and then bringing those things into what you are doing." -- Steve Jobs: Co-founder of Apple -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/e19795d2/attachment.bin From msc at freeswitch.org Mon Nov 28 23:03:31 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Nov 2011 12:03:31 -0800 Subject: [Freeswitch-users] Informal survey: calling multiple endpoints sequentially Message-ID: Hello all, I'm hoping to get some community feedback as I work on the FreeSWITCH Cookbook. The issue at hand is how you handle the dialing of sequential endpoints. For those of you who do this, would you mind replying to the list and letting us know what you do? I'm curious what methods you use, such as: pipe-sep list of endpoints in a bridge/originate continue_on_fail=true and multiple individual bridge apps mod_lcr or similar Jay Binks' transfer_on_fail method etc. I am particularly interested in the problems that you are trying to solve. In other words - how did the method you chose help you solve your problem? Your perspectives will help me very much. Thanks all! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/a07f82d0/attachment.html From henrikaagaardsorensen at gmail.com Mon Nov 28 23:50:54 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Mon, 28 Nov 2011 21:50:54 +0100 Subject: [Freeswitch-users] Mod_cdr_csv, no cdr_csv.conf.xml in standard installation. In-Reply-To: <1322506582708-7039930.post@n2.nabble.com> References: <1322506582708-7039930.post@n2.nabble.com> Message-ID: Dear Jeff. I'm pretty sure the file is not there. Otherwise my question would be rather odd I guess :) This is output from the terminal as proof: # /usr/local/freeswitch/conf/autoload_configs# ls acl.conf.xml mongo.conf.xml alsa.conf.xml nibblebill.conf.xml blacklist.conf.xml opal.conf.xml cdr_mongodb.conf.xml osp.conf.xml cdr_pg_csv.conf.xml perl.conf.xml cdr_sqlite.conf.xml pocketsphinx.conf.xml cepstral.conf.xml portaudio.conf.xml cidlookup.conf.xml post_load_modules.conf.xml console.conf.xml presence_map.conf.xml db.conf.xml python.conf.xml dialplan_directory.conf.xml redis.conf.xml dingaling.conf.xml rss.conf.xml easyroute.conf.xml rtmp.conf.xml enum.conf.xml sangoma_codec.conf.xml erlang_event.conf.xml shout.conf.xml event_multicast.conf.xml skinny.conf.xml event_socket.conf.xml sofia.conf.xml fax.conf.xml spandsp.conf.xml fifo.conf.xml spidermonkey.conf.xml hash.conf.xml switch.conf.xml http_cache.conf.xml syslog.conf.xml java.conf.xml timezones.conf.xml lcr.conf.xml tts_commandline.conf.xml local_stream.conf.xml unicall.conf.xml locations.conf.xml unimrcp.conf.xml logfile.conf.xml voicemail.conf.xml lua.conf.xml xml_curl.conf.xml memcache.conf.xml xml_rpc.conf.xml modules.conf.xml zeroconf.conf.xml On Mon, Nov 28, 2011 at 7:56 PM, Jeff Lenk wrote: > The file is definitely there: > > conf\autoload_configs > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Mod-cdr-csv-no-cdr-csv-conf-xml-in-standard-installation-tp7036532p7039930.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111128/a7056aad/attachment.html From valery.kalinin at gmail.com Tue Nov 29 05:39:06 2011 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Tue, 29 Nov 2011 08:39:06 +0600 Subject: [Freeswitch-users] Where is mod_sms? Message-ID: I update Freeswitch from git, but mod_sms not found. modules.conf - not found src/mod/applications/ - not found How to download mod_sms? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/eb794599/attachment-0001.html From curriegrad2004 at gmail.com Tue Nov 29 06:06:30 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 28 Nov 2011 19:06:30 -0800 Subject: [Freeswitch-users] Where is mod_sms? In-Reply-To: References: Message-ID: make mod_sms-all On Mon, Nov 28, 2011 at 6:39 PM, Valery Kalinin wrote: > I update Freeswitch from git, but mod_sms not found. > modules.conf - not found > src/mod/applications/ - not found > > How to download mod_sms? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 valter at fastway.com.br Tue Nov 29 02:22:55 2011 From: valter at fastway.com.br (Valter Nogueira) Date: Mon, 28 Nov 2011 21:22:55 -0200 Subject: [Freeswitch-users] error compiling 1.0.6 on ubuntu 10.10 64 bits Message-ID: after ./configure, make ends with following message gcc -E /home/fastway/freeswitch-1.0.6/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null | grep -v "^#" > src/include/switch_swigable_cpp.h make: *** [src/include/switch_swigable_cpp.h] Error 1 thanks Valter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/ec25a178/attachment.html From timtitov at hotmail.com Tue Nov 29 05:38:26 2011 From: timtitov at hotmail.com (Tim Titov) Date: Mon, 28 Nov 2011 18:38:26 -0800 Subject: [Freeswitch-users] tiff file affects transmission over T.38 In-Reply-To: References: Message-ID: First I converted a pdf to tiff with magick and no compression. Then I used tiffcp (3.9.2) to compress it as g3. The file could not be transmitted over T.38 Gafachi or flowroute. It was frustrating to figure this out. I hope others don't run into this problem. P.S.: ghostscript works fine. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/eb566a88/attachment.html From brian at freeswitch.org Tue Nov 29 06:19:18 2011 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Nov 2011 21:19:18 -0600 Subject: [Freeswitch-users] tiff file affects transmission over T.38 In-Reply-To: References: Message-ID: see http://www.soft-switch.org/spandsp_faq/ar01s14.html You're trying to way too hard! /b On Nov 28, 2011, at 8:38 PM, Tim Titov wrote: > > First I converted a pdf to tiff with magick and no compression. Then I used tiffcp (3.9.2) to compress it as g3. The file could not be transmitted over T.38 Gafachi or flowroute. It was frustrating to figure this out. I hope others don't run into this problem. > P.S.: ghostscript works fine. _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 fraserredmond at gmail.com Tue Nov 29 06:45:48 2011 From: fraserredmond at gmail.com (Fraser Redmond) Date: Mon, 28 Nov 2011 22:45:48 -0500 Subject: [Freeswitch-users] xml_curl directory - doing authentication in cgi, how to recreate user's password? Message-ID: I am setting up a connection to a database of users, whose passwords have been saved as a one-way hash. That means that my xml_curl php/sql will need to perform the authentication, and return a user without any password. (According to Anthony, back in 2008: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/029882.html ) Only thing is I can't find any mention anywhere of how to re-generate the user's password from the sip_auth variables in order to run it through my one-way hash for comparison to the database. It's got to be something to do with these: sip_auth_nonce = 4d95dd9f-2247-474a-8496-aa7c08700fe7 sip_auth_cnonce = a088c6b6ba18d1387a45998b6bfa842d sip_auth_nc = 0000000a sip_auth_response = 9edefab216a46ed75f1ed1297dd9c9d3 Any ideas how to rebuild the original user's password? Or is there a way to send the password through as part of the post? (maybe using enable-post-var) Cheers, Fraser -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111128/b4903402/attachment.html From timtitov at hotmail.com Tue Nov 29 07:14:02 2011 From: timtitov at hotmail.com (Tim Titov) Date: Mon, 28 Nov 2011 20:14:02 -0800 Subject: [Freeswitch-users] tiff file affects transmission over T.38 In-Reply-To: EFED3B54-A9D3-4DA0-940A-3C42FD6A8455@freeswitch.org Message-ID: Brian, I was doing that because the quality of ghostscript black-and-white conversion isn't that great. I wanted to do some image processing to improve the result. From rendyfrx at gmail.com Tue Nov 29 07:59:19 2011 From: rendyfrx at gmail.com (Rendy) Date: Tue, 29 Nov 2011 12:59:19 +0800 Subject: [Freeswitch-users] xml_curl directory - doing authentication in cgi, how to recreate user's password? In-Reply-To: References: Message-ID: Hi, Why don't you let your user authenticate using hashed password then in php you return the user xml with the hashed password that is stored. In that way, you will not have any issue. I don't think you can rebuild the original password as what hash function is meant to be one way only. On Tue, Nov 29, 2011 at 11:45 AM, Fraser Redmond wrote: > I am setting up a connection to a database of users, whose passwords have > been saved as a one-way hash. > That means that my xml_curl php/sql will need to perform the authentication, > and return a user without any password. > (According to Anthony, back in > 2008:?http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/029882.html?) > Only thing is I can't find any mention anywhere of how to re-generate the > user's password from the sip_auth variables in order to run it through my > one-way hash for comparison to the database. > It's got to be something to do with these: > sip_auth_nonce = 4d95dd9f-2247-474a-8496-aa7c08700fe7 > sip_auth_cnonce = a088c6b6ba18d1387a45998b6bfa842d > sip_auth_nc = 0000000a > sip_auth_response = 9edefab216a46ed75f1ed1297dd9c9d3 > Any ideas how to rebuild the original user's password? > Or is there a way to send the password through as part of the post? (maybe > using?enable-post-var) > Cheers, > Fraser > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 daveh at beachdognet.com Tue Nov 29 06:22:25 2011 From: daveh at beachdognet.com (Dave Horton) Date: Mon, 28 Nov 2011 22:22:25 -0500 Subject: [Freeswitch-users] Anyone got experience with how limit scales with db back end? Message-ID: <641E4A06-08A3-4E6E-92DF-8219337C13FE@beachdognet.com> I am using limit with db (sqlite) backend to limit all calls into my server. I have 72 inbound channels on gateways sending to my freeswitch, which is running on a single processor dual core 3GHz server, with the sqlite call limits database on the local disk, and at times I am seeing a delay of up to 3 seconds between the arrival of an invite and FS sending a 180 Ringing from the ring ready application. Other delays in signaling I also see intermittently. The only real processing going on before the ring_ready application is executed is the limit application. I see from the code there is a global mutex that looks like will result in all limit checks being single threaded through the code, and I am wondering if I am experience slight intermittent bottlenecks at peak call times. I realize the hash backend is faster and I may switch to that (decision pre-dated my arrival), but I am wondering if anyone out there has any good data about how many cps the limit application with the db backend can scale to. Obviously its dependent on hardware, but I'd still like to get any real world data (or suggestions) that are out there. From msc at freeswitch.org Tue Nov 29 09:35:50 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Nov 2011 22:35:50 -0800 Subject: [Freeswitch-users] error compiling 1.0.6 on ubuntu 10.10 64 bits In-Reply-To: References: Message-ID: 1.0.6 is old and unsupported. Your absolute best bet is to grab the latest git and install. Check this page: http://wiki.freeswitch.org/wiki/Installation_Guide#Recommended:_Git It's actually not all that difficult to do... -MC On Mon, Nov 28, 2011 at 3:22 PM, Valter Nogueira wrote: > after ./configure, make ends with following message > > gcc -E /home/fastway/freeswitch-1.0.6/src/include/switch_cpp.h > -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x > -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null | > grep -v "^#" > src/include/switch_swigable_cpp.h > make: *** [src/include/switch_swigable_cpp.h] Error 1 > > > thanks > > Valter > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111128/b92b263b/attachment-0001.html From avi at avimarcus.net Tue Nov 29 09:46:06 2011 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 29 Nov 2011 08:46:06 +0200 Subject: [Freeswitch-users] Anyone got experience with how limit scales with db back end? In-Reply-To: <641E4A06-08A3-4E6E-92DF-8219337C13FE@beachdognet.com> References: <641E4A06-08A3-4E6E-92DF-8219337C13FE@beachdognet.com> Message-ID: Is there any reason it's being stored on disk rather than ram? A waste of i/o of there's not.. -Avi Marcus On Tue, Nov 29, 2011 at 5:22 AM, Dave Horton wrote: > I am using limit with db (sqlite) backend to limit all calls into my > server. I have 72 inbound channels on gateways sending to my freeswitch, > which is running on a single processor dual core 3GHz server, with the > sqlite call limits database on the local disk, and at times I am seeing a > delay of up to 3 seconds between the arrival of an invite and FS sending a > 180 Ringing from the ring ready application. Other delays in signaling I > also see intermittently. The only real processing going on before the > ring_ready application is executed is the limit application. I see from > the code there is a global mutex that looks like will result in all limit > checks being single threaded through the code, and I am wondering if I am > experience slight intermittent bottlenecks at peak call times. I realize > the hash backend is faster and I may switch to that (decision pre-dated my > arrival), but I am wondering if anyone out there has any good data about > how many cps the limit application with the db backend can scale to. > Obviously its dependent on hardware, but I'd still like to get any real > world data (or suggestions) that are out there. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/b2cc6f02/attachment.html From mchlmll at gmail.com Tue Nov 29 11:55:48 2011 From: mchlmll at gmail.com (Michele M) Date: Tue, 29 Nov 2011 09:55:48 +0100 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: References: Message-ID: Ok, but again , hardware requirements? (CPU, RAM, etc, etc) 2011/11/28 Ken Rice > Keep in Mind that Mod_conference (and anyone elses conference software for > that matter) take everything to slinear for mixing. Unfortunately you cant > mix encoded auto with encoded audio very well. > > So far 100 Channels in a conf bridge, he would require 100 licenses. > > K > > > On 11/28/11 9:16 AM, "curriegrad2004" wrote: > > > In this case there is no need to purchase any G729 software licenses > at all if > > you are doing all of this in passthrough mode (meaning both > ends are g729 > > codecs and FS is just simply passing them through). The > default G729 codec > > that is included with FS is passthrough. > > You could also look into > > this: > http://wiki.freeswitch.org/wiki/Proxy_Media or > > this > http://wiki.freeswitch.org/wiki/Bypass_Media > > > > > >> On Mon, Nov 28, 2011 at 5:57 AM, Michele M wrote: > > >> Hi there, > it is my intention to buy 100 g729 sw codecs from FS to setup a > >> meet-me > conference system.Which are the minimal hw requirements to support > >> 100 conference legs where all the codecs implied are G729 ? ( meaning no > >> transcoding from pcmu->G729 and viceversa). > > > > Thx in advance > > > > Michele > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/4ccfd666/attachment.html From roland at haenel.me Tue Nov 29 16:11:30 2011 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Tue, 29 Nov 2011 14:11:30 +0100 Subject: [Freeswitch-users] Detecting fax on an incoming call leg (direct T.38 case) Message-ID: I'm trying to reliably detect whether there is a fax machine on an incoming channel. Right now, it works nicely with the following dial plan: answer spandsp_start_fax_detect sleep 5000 spandsp_stop_fax_detect [... no fax there, so now do stuff that has nothing to do with fax ...] If the remote end is a PSTN fax, spandsp will fire an event upon detecting the CNG tone, and I will react accordingly. However, if the remove end is a T.38 fax, which will immediately re-invite after answer, then the following problem occurs: FreeSWITCH does not answer the re-invite (just says 100 Trying and nothing follows). Even though the remove end fax transmits the CNG tone afterwards, it is not detected by spandsp. I have recorded the call to make sure the CGN tone is there, and it is there. I have even played this tone into another phone call, where it is successfully detected by spandsp. All I can guess is that the non-answered re-invite might block spandsp from detecting the tone? Greetings, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/323dd464/attachment.html From roland at haenel.me Tue Nov 29 16:17:01 2011 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Tue, 29 Nov 2011 14:17:01 +0100 Subject: [Freeswitch-users] Detecting fax on an incoming call leg (direct T.38 case) In-Reply-To: References: Message-ID: one more question: what would also solve my problem if it was possible to know about the re-invite on the event socket (inbound) in some way - then I could switch to rxfax without doing any tone detection at all. But it seems that there is no way that FreeSWITCH lets me know about the re-invite... greetings, roland 2011/11/29 Roland H?nel > I'm trying to reliably detect whether there is a fax machine on an > incoming channel. > > Right now, it works nicely with the following dial plan: > > answer > spandsp_start_fax_detect > sleep 5000 > spandsp_stop_fax_detect > [... no fax there, so now do stuff that has nothing to do with fax > ...] > > If the remote end is a PSTN fax, spandsp will fire an event upon detecting > the CNG tone, and I will react accordingly. > > However, if the remove end is a T.38 fax, which will immediately re-invite > after answer, then the following problem occurs: > > FreeSWITCH does not answer the re-invite (just says 100 Trying and nothing > follows). Even though the remove end fax transmits the CNG tone afterwards, > it is not detected by spandsp. I have recorded the call to make sure the > CGN tone is there, and it is there. I have even played this tone into > another phone call, where it is successfully detected by spandsp. All I can > guess is that the non-answered re-invite might block spandsp from detecting > the tone? > > Greetings, > Roland > > -- Gru?, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/14bf813b/attachment.html From brian at freeswitch.org Tue Nov 29 17:47:23 2011 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Nov 2011 08:47:23 -0600 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: References: Message-ID: Please deposit twenty five cents for the next 10 seconds. You know this is outlined on the wiki... YOU will still need to benchmark YOUR application for YOUR needs to ensure that YOU do not exceed YOUR platforms capabilities because its different for EVERYONE! /b On Nov 29, 2011, at 2:55 AM, Michele M wrote: > Ok, but again > , hardware requirements? (CPU, RAM, etc, etc) > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/66d8a432/attachment-0001.html From gmaruzz at gmail.com Tue Nov 29 17:49:49 2011 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 29 Nov 2011 15:49:49 +0100 Subject: [Freeswitch-users] Skypopen audio problems In-Reply-To: <1322413960.23145.YahooMailNeo@web27606.mail.ukl.yahoo.com> References: <1322398858.57178.YahooMailNeo@web27607.mail.ukl.yahoo.com> <1322413960.23145.YahooMailNeo@web27606.mail.ukl.yahoo.com> Message-ID: Hi JoZu 1) can you delete from the dialplan and see if that makes a difference? 2) are you using the skypopen.conf.xml made by install.pl ? 3) can you please post your skytpopen.conf.xml ? 4) if we don't find out the problem, can you prepare for me to ssh into your machine, and investigate? -giovanni On Sun, Nov 27, 2011 at 6:12 PM, JoZu wrote: > > > Thanks Giovanni for you respose, you understandme perfectly. > A frew aditional information: > My SIP (DID) provider does not requiere a password,? and the configuration > does not require any specific parameter. I put SIP/[hidden email]:5080 at > provider mapping. > In Freeswitch: > I made change only in two files: > acl.conf.xml - for include in domains list the providers IP'S > public.xml - for include this: > > ????? > ?????? > ????? ? <--- skype > user > ????? > ??? > skypopen.conf.xml, dont change anything at this moment the file are same as > source. > > ________________________________ > De: Giovanni Maruzzelli-2 [via freeswitch-users] <[hidden email]> > Para: JoZu <[hidden email]> > Enviado: domingo 27 de noviembre de 2011 17:31 > Asunto: Re: Skypopen audio problems > > No, skypopen does not need a soundcard. > > >From your log nothing strange in skypopen. > > Let me recap if I have understood: > > 1) you have a SIP DID > 2) this SIP DID is going to an extension that bridges to a skypopen > outbound call to SKYPE_22 > 3) the SIP incoming call is correctly bridged to SKYPE_22 > 4) DID caller can hear SKYPE_22 talking > 5) SKYPE_22 do not hear DID caller talking. This is the problem. > > I have understood correctly? > > Can you post the skypopen.conf.xml and the dialplan snippet that do the > bridge? > > -giovanni > > On Sun, Nov 27, 2011 at 2:00 PM, asdfa asdfasdf <[hidden email]> wrote: > >> Hi, i have a little problem i have FS installed, skypeopen_oss client and >> skypeopen.ko oss sound driver installed. >> I configure in public a DID number bridge to skypopen/RR/skype22, the call >> going from DID to SKYPE1 and call SKYPE22, here is the LOG: >> http://pastebin.freeswitch.org/17877 >> All work OK, but the sound from DID caller to Skype22 dont reach, no >> sound, >> Skype22 to DID caller works ok. Questions: >> Skype_oss and the skypeopen.ko require a sound card to work? >> What is the problem? Codecs? >> Please helpme, thanks and sorry my english. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> [hidden email] >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > > > ________________________________ > If you reply to this email, your message will be added to the discussion > below: > http://freeswitch-users.2379917.n2.nabble.com/Skypopen-audio-problems-tp7036246p7036606.html > To start a new topic under freeswitch-users, email [hidden email] > To unsubscribe from freeswitch-users, click here. > NAML > > > > ________________________________ > View this message in context: Re: Skypopen audio problems > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 kris at kriskinc.com Tue Nov 29 18:09:13 2011 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 29 Nov 2011 10:09:13 -0500 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: References: Message-ID: You're going to pay $1000 for G729 software licenses. Sangoma cards include the license in the cost of the card and the transcoding frees up CPU resources for other things. I'm not sure of the pricing but you can probably get a 100 channel Sangoma card for less than you're going to pay in software licenses. When you start getting in the > 100 channel range you should really, really be using a Sangoma card. There's very little reason not to. On Tue, Nov 29, 2011 at 3:55 AM, Michele M wrote: > Ok, but again > , hardware requirements? (CPU, RAM, etc, etc) > > 2011/11/28 Ken Rice >> >> Keep in Mind that Mod_conference (and anyone elses conference software for >> that matter) take everything to slinear for mixing. Unfortunately you cant >> mix encoded auto with encoded audio very well. >> >> So far 100 Channels in a conf bridge, he would require 100 licenses. >> >> K >> >> >> On 11/28/11 9:16 AM, "curriegrad2004" wrote: >> >> > In this case there is no need to purchase any G729 software licenses >> at all if >> > you are doing all of this in passthrough mode (meaning both >> ends are g729 >> > codecs and FS is just simply passing them through). The >> default G729 codec >> > that is included with FS is passthrough. >> >> You could also look into >> > this: >> http://wiki.freeswitch.org/wiki/Proxy_Media or >> > this >> http://wiki.freeswitch.org/wiki/Bypass_Media >> >> >> >> >> >> On Mon, Nov 28, 2011 at 5:57 AM, Michele M wrote: >> >> >> Hi there, >> it is my intention to buy 100 g729 sw codecs from FS to setup a >> >> meet-me >> conference?system.Which are the minimal hw requirements to support >> >> 100 conference legs where all the codecs implied?are G729 ? ( meaning >> >> no >> >> transcoding from pcmu->G729 and viceversa). >> > >> > Thx in advance >> > >> > Michele >> > >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 From curriegrad2004 at gmail.com Tue Nov 29 18:18:02 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 29 Nov 2011 07:18:02 -0800 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: References: Message-ID: Not to mention at a conference call type of setup, you'd need extra processing done on the machine that's handling approx. 100 users at any given time on G729, as I already said above, start considering with using a hardware transcoding card from sangoma or etc. As funny as it may seem, your CPU in your box wasn't really designed to behave or act like a DSP, although Intel tried to :P On Tue, Nov 29, 2011 at 7:09 AM, Kristian Kielhofner wrote: > You're going to pay $1000 for G729 software licenses. > > Sangoma cards include the license in the cost of the card and the > transcoding frees up CPU resources for other things. ?I'm not sure of > the pricing but you can probably get a 100 channel Sangoma card for > less than you're going to pay in software licenses. > > When you start getting in the > 100 channel range you should really, > really be using a Sangoma card. ?There's very little reason not to. > > On Tue, Nov 29, 2011 at 3:55 AM, Michele M wrote: >> Ok, but again >> , hardware requirements? (CPU, RAM, etc, etc) >> >> 2011/11/28 Ken Rice >>> >>> Keep in Mind that Mod_conference (and anyone elses conference software for >>> that matter) take everything to slinear for mixing. Unfortunately you cant >>> mix encoded auto with encoded audio very well. >>> >>> So far 100 Channels in a conf bridge, he would require 100 licenses. >>> >>> K >>> >>> >>> On 11/28/11 9:16 AM, "curriegrad2004" wrote: >>> >>> > In this case there is no need to purchase any G729 software licenses >>> at all if >>> > you are doing all of this in passthrough mode (meaning both >>> ends are g729 >>> > codecs and FS is just simply passing them through). The >>> default G729 codec >>> > that is included with FS is passthrough. >>> >>> You could also look into >>> > this: >>> http://wiki.freeswitch.org/wiki/Proxy_Media or >>> > this >>> http://wiki.freeswitch.org/wiki/Bypass_Media >>> >>> >>> >>> >>> >> On Mon, Nov 28, 2011 at 5:57 AM, Michele M wrote: >>> >>> >> Hi there, >>> it is my intention to buy 100 g729 sw codecs from FS to setup a >>> >> meet-me >>> conference?system.Which are the minimal hw requirements to support >>> >> 100 conference legs where all the codecs implied?are G729 ? ( meaning >>> >> no >>> >> transcoding from pcmu->G729 and viceversa). >>> > >>> > Thx in advance >>> > >>> > Michele >>> > >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 fraserredmond at gmail.com Tue Nov 29 19:19:08 2011 From: fraserredmond at gmail.com (Fraser Redmond) Date: Tue, 29 Nov 2011 11:19:08 -0500 Subject: [Freeswitch-users] xml_curl directory - doing authentication in cgi, how to recreate user's password? In-Reply-To: References: Message-ID: Thanks Randy... but I think either I don't understand you, or you don't understand me... The password stored in the database has been hashed using mysql's ENCRYPT function with a seed (because it's not good security policy to store a password in any recoverable format.) I think you're saying that the nonce is also a hashed version of the password that also can't be reverted back to the original password - is that right? Which means that I now have two hashes which have been generated using different methods, so there's no way to compare them - cant compare within the cgi, and can't send the Freeswitch format back for Freeswitch to compare. If that's the case (and I'd still like to be clear on that), is it possible to pass through the password in addition? (I'll be using https, so sending without hashing is ok.) Cheers, Fraser On 28 November 2011 23:59, Rendy wrote: > Hi, > Why don't you let your user authenticate using hashed password then in > php you return the user xml with the hashed password that is stored. > In that way, you will not have any issue. I don't think you can > rebuild the original password as what hash function is meant to be one > way only. > > > On Tue, Nov 29, 2011 at 11:45 AM, Fraser Redmond > wrote: > > I am setting up a connection to a database of users, whose passwords have > > been saved as a one-way hash. > > That means that my xml_curl php/sql will need to perform the > authentication, > > and return a user without any password. > > (According to Anthony, back in > > 2008: > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/029882.html > ) > > Only thing is I can't find any mention anywhere of how to re-generate the > > user's password from the sip_auth variables in order to run it through my > > one-way hash for comparison to the database. > > It's got to be something to do with these: > > sip_auth_nonce = 4d95dd9f-2247-474a-8496-aa7c08700fe7 > > sip_auth_cnonce = a088c6b6ba18d1387a45998b6bfa842d > > sip_auth_nc = 0000000a > > sip_auth_response = 9edefab216a46ed75f1ed1297dd9c9d3 > > Any ideas how to rebuild the original user's password? > > Or is there a way to send the password through as part of the post? > (maybe > > using enable-post-var) > > Cheers, > > Fraser > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/a4dd8526/attachment.html From daemonserj at gmail.com Tue Nov 29 10:09:48 2011 From: daemonserj at gmail.com (daemonserj TVC) Date: Tue, 29 Nov 2011 14:09:48 +0700 Subject: [Freeswitch-users] Fwd: FS compile errors on Solaris 10 sparc In-Reply-To: References: Message-ID: Hello all! I'm newbie :) System is SunOS tornado 5.10 Generic_144488-17 sun4u sparc SUNW,Sun-Fire-V240 Solaris SunStudio 12, CBE 1.7 I tried to compile concerning http://wiki.freeswitch.org/wiki/Installation_Guide#Solaris And I got : CC: Warning: Option -ggdb passed to ld, if ld is invoked, ignored otherwise CC: Warning: Option -fPIC passed to ld, if ld is invoked, ignored otherwise "/export/home/freeswitch/freeswitch/libs/esl/src/include/esl.h", line 485: Error: "," expected instead of "int". "/export/home/freeswitch/freeswitch/libs/esl/src/include/esl.h", line 491: Error: A declaration was expected instead of "return". "/export/home/freeswitch/freeswitch/libs/esl/src/include/esl.h", line 491: Error: s1 is not defined. "src/esl_oop.cpp", line 295: Error: __func__ is not defined. "src/esl_oop.cpp", line 300: Error: __func__ is not defined. "src/esl_oop.cpp", line 364: Error: __func__ is not defined. "src/esl_oop.cpp", line 391: Error: __func__ is not defined. "src/esl_oop.cpp", line 397: Error: __func__ is not defined. "src/esl_oop.cpp", line 404: Error: __func__ is not defined. "src/esl_oop.cpp", line 409: Error: __func__ is not defined. "src/esl_oop.cpp", line 416: Error: __func__ is not defined. "src/esl_oop.cpp", line 421: Error: __func__ is not defined. "src/esl_oop.cpp", line 429: Error: __func__ is not defined. "src/esl_oop.cpp", line 434: Error: __func__ is not defined. "src/esl_oop.cpp", line 442: Error: __func__ is not defined. "src/esl_oop.cpp", line 447: Error: __func__ is not defined. "src/esl_oop.cpp", line 455: Error: __func__ is not defined. "src/esl_oop.cpp", line 460: Error: __func__ is not defined. "src/esl_oop.cpp", line 469: Error: __func__ is not defined. "src/esl_oop.cpp", line 474: Error: __func__ is not defined. "src/esl_oop.cpp", line 483: Error: __func__ is not defined. "src/esl_oop.cpp", line 488: Error: __func__ is not defined. "src/esl_oop.cpp", line 496: Error: __func__ is not defined. "src/esl_oop.cpp", line 501: Error: __func__ is not defined. 24 Error(s) detected. make[5]: *** [src/esl_oop.o] Error 24 make[4]: *** [/export/home/freeswitch/freeswitch/libs/esl/libesl.so] Error 2 make[3]: *** [mod_hash-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 -bash-3.00$ -bash-3.00$ uname SunOS -bash-3.00$ uname -a PS. ?XXXPP configured to /usr/bin/cc -E it seems /libs/esl/src/include/esl.h has not nothing criminal From dflawal at hotmail.com Tue Nov 29 18:34:05 2011 From: dflawal at hotmail.com (David Lawal) Date: Tue, 29 Nov 2011 08:34:05 -0700 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: References: , , , , Message-ID: I currently run freeswith in a VM environment in the cloud, does this mean for G729 license I will run into issues with anything above 100 channels? Would have never imagined that... Any input to this will help with FS and G729 trans-coding requirements. Thanks David > Date: Tue, 29 Nov 2011 07:18:02 -0800 > From: curriegrad2004 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] MInimal Hardware requirements - G729 > > Not to mention at a conference call type of setup, you'd need extra > processing done on the machine that's handling approx. 100 users at > any given time on G729, as I already said above, start considering > with using a hardware transcoding card from sangoma or etc. > > As funny as it may seem, your CPU in your box wasn't really designed > to behave or act like a DSP, although Intel tried to :P > > On Tue, Nov 29, 2011 at 7:09 AM, Kristian Kielhofner wrote: > > You're going to pay $1000 for G729 software licenses. > > > > Sangoma cards include the license in the cost of the card and the > > transcoding frees up CPU resources for other things. I'm not sure of > > the pricing but you can probably get a 100 channel Sangoma card for > > less than you're going to pay in software licenses. > > > > When you start getting in the > 100 channel range you should really, > > really be using a Sangoma card. There's very little reason not to. > > > > On Tue, Nov 29, 2011 at 3:55 AM, Michele M wrote: > >> Ok, but again > >> , hardware requirements? (CPU, RAM, etc, etc) > >> > >> 2011/11/28 Ken Rice > >>> > >>> Keep in Mind that Mod_conference (and anyone elses conference software for > >>> that matter) take everything to slinear for mixing. Unfortunately you cant > >>> mix encoded auto with encoded audio very well. > >>> > >>> So far 100 Channels in a conf bridge, he would require 100 licenses. > >>> > >>> K > >>> > >>> > >>> On 11/28/11 9:16 AM, "curriegrad2004" wrote: > >>> > >>> > In this case there is no need to purchase any G729 software licenses > >>> at all if > >>> > you are doing all of this in passthrough mode (meaning both > >>> ends are g729 > >>> > codecs and FS is just simply passing them through). The > >>> default G729 codec > >>> > that is included with FS is passthrough. > >>> > >>> You could also look into > >>> > this: > >>> http://wiki.freeswitch.org/wiki/Proxy_Media or > >>> > this > >>> http://wiki.freeswitch.org/wiki/Bypass_Media > >>> > >>> > >>> > >>> > >>> >> On Mon, Nov 28, 2011 at 5:57 AM, Michele M wrote: > >>> > >>> >> Hi there, > >>> it is my intention to buy 100 g729 sw codecs from FS to setup a > >>> >> meet-me > >>> conference system.Which are the minimal hw requirements to support > >>> >> 100 conference legs where all the codecs implied are G729 ? ( meaning > >>> >> no > >>> >> transcoding from pcmu->G729 and viceversa). > >>> > > >>> > Thx in advance > >>> > > >>> > Michele > >>> > > >>> > >>> > >>> > >>> _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.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 > >> > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/bbc7dad7/attachment.html From msc at freeswitch.org Tue Nov 29 21:37:31 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Nov 2011 10:37:31 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Tomorrow: Special Guest Message-ID: Hello all! I just wanted to give everyone a heads up about tomorrow's conference call. We have a special guest scheduled to join us: Jean-Marc Valin from Mozilla! Jean-Marc is a codec guru who is working diligently on the OPUS codec. He will be coming to talk to us about the codec and to answer any questions that you may have. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_30 We look forward to having you on our call tomorrow! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/752c0b03/attachment.html From anthony.minessale at gmail.com Tue Nov 29 21:49:35 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Nov 2011 12:49:35 -0600 Subject: [Freeswitch-users] Where is mod_sms? In-Reply-To: References: Message-ID: it certainly is: you are clearly not actually updating if you can't see it. http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_sms/mod_sms.c On Mon, Nov 28, 2011 at 8:39 PM, Valery Kalinin wrote: > I update Freeswitch from git, but mod_sms not found. > modules.conf - not found > src/mod/applications/ - not found > > How to download mod_sms? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/b09255c4/attachment.html From msc at freeswitch.org Tue Nov 29 22:08:18 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Nov 2011 11:08:18 -0800 Subject: [Freeswitch-users] originate bridge breaks about after 2 seconds In-Reply-To: References: Message-ID: What does the console show leading up to the bridge being disconnected. -MC On Thu, Nov 24, 2011 at 6:45 AM, Thomas Hoellriegel wrote: > Hi i make a originatebride from a shellscript: > #!/bin/bash > /usr/local/freeswitch/bin/fs_**cli -x "bgapi originate\ > {ignore_early_media=true,**hangup_after_bridge=false,\ > originate_retries=100,**origination_uuid=thomas_$1,\ > origination_caller_id_name=**Callback,originate_retry_**sleep_ms=60000,\ > originate_timeout=900}**loopback/$1/disa\ > &conference(7400-sip.blindi.**net at default)" > > for example: i call 10 users in a conference, > these works fine. > > I bridge 2 existing calls for example: > uuid_transfer thomas_1 park inline > uuid_transfer thomas_2 park inline > uuid_bridge thomas_1 thomas_2 > > The bridge works abbout 2 seconds, and fs hangup the bridge. > > can your help please? > Thanks. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/e3f3b32c/attachment-0001.html From msc at freeswitch.org Tue Nov 29 22:21:45 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Nov 2011 11:21:45 -0800 Subject: [Freeswitch-users] playAndGetDigits problem in LUA program In-Reply-To: References: Message-ID: Do you have a while(session:ready) wrapped around this? -MC On Thu, Nov 24, 2011 at 6:24 PM, Sanath Prasanna wrote: > Hi all, > I used following to get phone number from caller in LUA program. > > function get_dest() > new_number = session:playAndGetDigits(9, 11, 3, 3000, "#", lang_path .. > "enter_dest_number.wav", lang_path .. "error.wav", "\\d+"); > session:say(new_number, "en", "number", "pronounced"); > session:setVariable("dest_numbers",new_number); > end > > -- main -- > freeswitch.consoleLog("info", "Voicemail - select_main_menu:\n\r"); > reeswitch.consoleLog("info", "Voicemail - Select English :\n\r"); > get_dest(); > .. > .. > > if caller hang off the phone without entering number(dtmf), I can see lot > of same log lines in the freeswitch console log. (around 50000 lines in 1 > or 2 sec.Pls see below). > *is it bug or just need to ignore. Pls advice. * > > Br, > Sanath > > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > select_main_menu: > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select > English > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > ./vm/en/ > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > select_main_menu: > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select > English > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > ./vm/en/ > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > select_main_menu: > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select > English > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > ./vm/en/ > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > select_main_menu: > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select > English > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > ./vm/en/ > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > select_main_menu: > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select > English > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > ./vm/en/ > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > select_main_menu: > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select > English > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > ./vm/en/ > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > select_main_menu: > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select > English > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > ./vm/en/ > 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - > select_main_menu: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/9bb8fd53/attachment.html From msc at freeswitch.org Tue Nov 29 22:34:08 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Nov 2011 11:34:08 -0800 Subject: [Freeswitch-users] Mod_cdr_csv, no cdr_csv.conf.xml in standard installation. In-Reply-To: References: <1322506582708-7039930.post@n2.nabble.com> Message-ID: Something seems to have happened to it. Go to your source directory and try this: "make samples" That should restore the default cdr_csv.conf.xml -MC 2011/11/28 Henrik Aagaard S?rensen > Dear Jeff. > > I'm pretty sure the file is not there. Otherwise my question would be > rather odd I guess :) > > This is output from the terminal as proof: > > # /usr/local/freeswitch/conf/autoload_configs# ls > acl.conf.xml mongo.conf.xml > alsa.conf.xml nibblebill.conf.xml > blacklist.conf.xml opal.conf.xml > cdr_mongodb.conf.xml osp.conf.xml > cdr_pg_csv.conf.xml perl.conf.xml > cdr_sqlite.conf.xml pocketsphinx.conf.xml > cepstral.conf.xml portaudio.conf.xml > cidlookup.conf.xml post_load_modules.conf.xml > console.conf.xml presence_map.conf.xml > db.conf.xml python.conf.xml > dialplan_directory.conf.xml redis.conf.xml > dingaling.conf.xml rss.conf.xml > easyroute.conf.xml rtmp.conf.xml > enum.conf.xml sangoma_codec.conf.xml > erlang_event.conf.xml shout.conf.xml > event_multicast.conf.xml skinny.conf.xml > event_socket.conf.xml sofia.conf.xml > fax.conf.xml spandsp.conf.xml > fifo.conf.xml spidermonkey.conf.xml > hash.conf.xml switch.conf.xml > http_cache.conf.xml syslog.conf.xml > java.conf.xml timezones.conf.xml > lcr.conf.xml tts_commandline.conf.xml > local_stream.conf.xml unicall.conf.xml > locations.conf.xml unimrcp.conf.xml > logfile.conf.xml voicemail.conf.xml > lua.conf.xml xml_curl.conf.xml > memcache.conf.xml xml_rpc.conf.xml > modules.conf.xml zeroconf.conf.xml > > > On Mon, Nov 28, 2011 at 7:56 PM, Jeff Lenk wrote: > >> The file is definitely there: >> >> conf\autoload_configs >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Mod-cdr-csv-no-cdr-csv-conf-xml-in-standard-installation-tp7036532p7039930.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/0b8b9e9c/attachment.html From kris at kriskinc.com Tue Nov 29 22:36:38 2011 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 29 Nov 2011 14:36:38 -0500 Subject: [Freeswitch-users] MInimal Hardware requirements - G729 In-Reply-To: References: Message-ID: Having over 100 G729 channels in software is fine but as I said there are very few cases where you'd want to do that because G729 hardware offers many advantages. The case you pointed out (VM in the cloud) would certainly be an exception. While there are inherent problems with any VoIP in VM environments I'll ignore them for the purposes of this discussion. Look back through the archives and see how many threads have started with "I'm running in a VM and I'm having RTP problems"... VM environments work well for moderate to low loads as hardware has gotten faster (combine eight moderately loaded web servers on one large server, etc). As you start using 100 channels of G729 (and conferencing, etc) you start leaving less room for other hosts on a VM. This begins to destroy the economic advantage of a VM. Additionally, a 120 channel Sangoma card costs about $1000. That's 20 more G729 channels that you'd get with software AND it won't generate as much heat or use as much power (or your CPU can be free for other things). I'd be interested to see the math on hosted VM + 100 software G729 channels (and CPU usage to go along with it) vs. Relatively low end dedicated server with Sangoma card. On Tue, Nov 29, 2011 at 10:34 AM, David Lawal wrote: > I currently run freeswith in a VM environment in the cloud, does this mean > for G729 license I will run into issues > with anything above 100 channels? Would have never imagined that... > Any input to this will help with FS and G729 trans-coding requirements. > Thanks > David > >> Date: Tue, 29 Nov 2011 07:18:02 -0800 >> From: curriegrad2004 at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] MInimal Hardware requirements - G729 >> >> Not to mention at a conference call type of setup, you'd need extra >> processing done on the machine that's handling approx. 100 users at >> any given time on G729, as I already said above, start considering >> with using a hardware transcoding card from sangoma or etc. >> >> As funny as it may seem, your CPU in your box wasn't really designed >> to behave or act like a DSP, although Intel tried to :P >> >> On Tue, Nov 29, 2011 at 7:09 AM, Kristian Kielhofner >> wrote: >> > You're going to pay $1000 for G729 software licenses. >> > >> > Sangoma cards include the license in the cost of the card and the >> > transcoding frees up CPU resources for other things. ?I'm not sure of >> > the pricing but you can probably get a 100 channel Sangoma card for >> > less than you're going to pay in software licenses. >> > >> > When you start getting in the > 100 channel range you should really, >> > really be using a Sangoma card. ?There's very little reason not to. >> > >> > On Tue, Nov 29, 2011 at 3:55 AM, Michele M wrote: >> >> Ok, but again >> >> , hardware requirements? (CPU, RAM, etc, etc) >> >> >> >> 2011/11/28 Ken Rice >> >>> >> >>> Keep in Mind that Mod_conference (and anyone elses conference software >> >>> for >> >>> that matter) take everything to slinear for mixing. Unfortunately you >> >>> cant >> >>> mix encoded auto with encoded audio very well. >> >>> >> >>> So far 100 Channels in a conf bridge, he would require 100 licenses. >> >>> >> >>> K >> >>> >> >>> >> >>> On 11/28/11 9:16 AM, "curriegrad2004" >> >>> wrote: >> >>> >> >>> > In this case there is no need to purchase any G729 software licenses >> >>> at all if >> >>> > you are doing all of this in passthrough mode (meaning both >> >>> ends are g729 >> >>> > codecs and FS is just simply passing them through). The >> >>> default G729 codec >> >>> > that is included with FS is passthrough. >> >>> >> >>> You could also look into >> >>> > this: >> >>> http://wiki.freeswitch.org/wiki/Proxy_Media or >> >>> > this >> >>> http://wiki.freeswitch.org/wiki/Bypass_Media >> >>> >> >>> >> >>> >> >>> >> >>> >> On Mon, Nov 28, 2011 at 5:57 AM, Michele M >> >>> >> wrote: >> >>> >> >>> >> Hi there, >> >>> it is my intention to buy 100 g729 sw codecs from FS to setup a >> >>> >> meet-me >> >>> conference?system.Which are the minimal hw requirements to support >> >>> >> 100 conference legs where all the codecs implied?are G729 ? ( >> >>> >> meaning >> >>> >> no >> >>> >> transcoding from pcmu->G729 and viceversa). >> >>> > >> >>> > Thx in advance >> >>> > >> >>> > Michele >> >>> > >> >>> >> >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.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 >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.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 >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 From henrikaagaardsorensen at gmail.com Tue Nov 29 23:15:54 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Tue, 29 Nov 2011 21:15:54 +0100 Subject: [Freeswitch-users] Mod_cdr_csv, no cdr_csv.conf.xml in standard installation. In-Reply-To: References: <1322506582708-7039930.post@n2.nabble.com> Message-ID: Hi Michael. It did. What does make samples else do? I do not which to have FreeSwitch installed with various sample-data, which I do not use. And that's what I though/think make samples does. On Tue, Nov 29, 2011 at 8:34 PM, Michael Collins wrote: > Something seems to have happened to it. Go to your source directory and > try this: > "make samples" > > That should restore the default cdr_csv.conf.xml > > -MC > > 2011/11/28 Henrik Aagaard S?rensen > >> Dear Jeff. >> >> I'm pretty sure the file is not there. Otherwise my question would be >> rather odd I guess :) >> >> This is output from the terminal as proof: >> >> # /usr/local/freeswitch/conf/autoload_configs# ls >> acl.conf.xml mongo.conf.xml >> alsa.conf.xml nibblebill.conf.xml >> blacklist.conf.xml opal.conf.xml >> cdr_mongodb.conf.xml osp.conf.xml >> cdr_pg_csv.conf.xml perl.conf.xml >> cdr_sqlite.conf.xml pocketsphinx.conf.xml >> cepstral.conf.xml portaudio.conf.xml >> cidlookup.conf.xml post_load_modules.conf.xml >> console.conf.xml presence_map.conf.xml >> db.conf.xml python.conf.xml >> dialplan_directory.conf.xml redis.conf.xml >> dingaling.conf.xml rss.conf.xml >> easyroute.conf.xml rtmp.conf.xml >> enum.conf.xml sangoma_codec.conf.xml >> erlang_event.conf.xml shout.conf.xml >> event_multicast.conf.xml skinny.conf.xml >> event_socket.conf.xml sofia.conf.xml >> fax.conf.xml spandsp.conf.xml >> fifo.conf.xml spidermonkey.conf.xml >> hash.conf.xml switch.conf.xml >> http_cache.conf.xml syslog.conf.xml >> java.conf.xml timezones.conf.xml >> lcr.conf.xml tts_commandline.conf.xml >> local_stream.conf.xml unicall.conf.xml >> locations.conf.xml unimrcp.conf.xml >> logfile.conf.xml voicemail.conf.xml >> lua.conf.xml xml_curl.conf.xml >> memcache.conf.xml xml_rpc.conf.xml >> modules.conf.xml zeroconf.conf.xml >> >> >> On Mon, Nov 28, 2011 at 7:56 PM, Jeff Lenk wrote: >> >>> The file is definitely there: >>> >>> conf\autoload_configs >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/Mod-cdr-csv-no-cdr-csv-conf-xml-in-standard-installation-tp7036532p7039930.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/3bf19c01/attachment.html From brian at freeswitch.org Tue Nov 29 23:19:20 2011 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Nov 2011 14:19:20 -0600 Subject: [Freeswitch-users] Mod_cdr_csv, no cdr_csv.conf.xml in standard installation. In-Reply-To: References: <1322506582708-7039930.post@n2.nabble.com> Message-ID: <5E154E5A-23E9-40FE-81EB-CE05D09330C5@freeswitch.org> This is the reason you didn't have the config file.. you would have to manually do that yourself or have us re-install all the missing sample data. /b On Nov 29, 2011, at 2:15 PM, Henrik Aagaard S?rensen wrote: > Hi Michael. > > It did. What does make samples else do? I do not which to have FreeSwitch > installed with various sample-data, which I do not use. And that's what I > though/think make samples does. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/85b9a338/attachment.html From henrikaagaardsorensen at gmail.com Tue Nov 29 23:24:36 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Tue, 29 Nov 2011 21:24:36 +0100 Subject: [Freeswitch-users] Mod_cdr_csv, no cdr_csv.conf.xml in standard installation. In-Reply-To: <5E154E5A-23E9-40FE-81EB-CE05D09330C5@freeswitch.org> References: <1322506582708-7039930.post@n2.nabble.com> <5E154E5A-23E9-40FE-81EB-CE05D09330C5@freeswitch.org> Message-ID: Hi Brian. I was just wondering what else it installed, as I wish to have the minimal installation of FS. Also, how did it work before without the conf-file? On Tue, Nov 29, 2011 at 9:19 PM, Brian West wrote: > This is the reason you didn't have the config file.. you would have to > manually do that yourself or have us re-install all the missing sample > data. > > /b > > On Nov 29, 2011, at 2:15 PM, Henrik Aagaard S?rensen wrote: > > Hi Michael. > > It did. What does make samples else do? I do not which to have FreeSwitch > installed with various sample-data, which I do not use. And that's what I > though/think make samples does. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/6fcc267a/attachment.html From vetali100 at gmail.com Wed Nov 30 01:29:09 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Tue, 29 Nov 2011 14:29:09 -0800 Subject: [Freeswitch-users] xml_curl directory - doing authentication in cgi, how to recreate user's password? In-Reply-To: References: Message-ID: This might solve your problem: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#a1-hash In short, you should hash not only the "password", but the concatenation of "username:domain:password" Then use xml_curl to return this hashed value and FS will do the authentication for you. Please let me know if this helps. Regards, Vitalie 2011/11/29 Fraser Redmond > Thanks Randy... but I think either I don't understand you, or you don't > understand me... > > The password stored in the database has been hashed using mysql's ENCRYPT > function with a seed (because it's not good security policy to store a > password in any recoverable format.) > > I think you're saying that the nonce is also a hashed version of the > password that also can't be reverted back to the original password - is > that right? > > Which means that I now have two hashes which have been generated using > different methods, so there's no way to compare them - cant compare within > the cgi, and can't send the Freeswitch format back for Freeswitch to > compare. > > If that's the case (and I'd still like to be clear on that), is it > possible to pass through the password in addition? (I'll be using https, so > sending without hashing is ok.) > > Cheers, > Fraser > > > > > > On 28 November 2011 23:59, Rendy wrote: > >> Hi, >> Why don't you let your user authenticate using hashed password then in >> php you return the user xml with the hashed password that is stored. >> In that way, you will not have any issue. I don't think you can >> rebuild the original password as what hash function is meant to be one >> way only. >> >> >> On Tue, Nov 29, 2011 at 11:45 AM, Fraser Redmond >> wrote: >> > I am setting up a connection to a database of users, whose passwords >> have >> > been saved as a one-way hash. >> > That means that my xml_curl php/sql will need to perform the >> authentication, >> > and return a user without any password. >> > (According to Anthony, back in >> > 2008: >> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/029882.html >> ) >> > Only thing is I can't find any mention anywhere of how to re-generate >> the >> > user's password from the sip_auth variables in order to run it through >> my >> > one-way hash for comparison to the database. >> > It's got to be something to do with these: >> > sip_auth_nonce = 4d95dd9f-2247-474a-8496-aa7c08700fe7 >> > sip_auth_cnonce = a088c6b6ba18d1387a45998b6bfa842d >> > sip_auth_nc = 0000000a >> > sip_auth_response = 9edefab216a46ed75f1ed1297dd9c9d3 >> > Any ideas how to rebuild the original user's password? >> > Or is there a way to send the password through as part of the post? >> (maybe >> > using enable-post-var) >> > Cheers, >> > Fraser >> > >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/fc677f74/attachment-0001.html From msc at freeswitch.org Wed Nov 30 01:40:28 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Nov 2011 14:40:28 -0800 Subject: [Freeswitch-users] Mod_cdr_csv, no cdr_csv.conf.xml in standard installation. In-Reply-To: References: <1322506582708-7039930.post@n2.nabble.com> Message-ID: 2011/11/29 Henrik Aagaard S?rensen > Hi Michael. > > It did. What does make samples else do? I do not which to have FreeSwitch > installed with various sample-data, which I do not use. And that's what I > though/think make samples does. > "make samples" will install the default config files if they are not already there. It will never overwrite any existing file. In terms of a minimal FreeSWITCH install, I highly recommend starting with this file: conf/autoload_configs/modules.conf.xml Disable the autoloading of any modules you don't wish to use in production. That's like 99% of the work right there. If you truly care about that last 1% then go ahead and clear out all the configuration files for the modules that you choose not to load. Also, be sure to back up your conf directory when you're done so that you don't run into this scenario again. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/24806c59/attachment.html From curriegrad2004 at gmail.com Wed Nov 30 01:52:58 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 29 Nov 2011 14:52:58 -0800 Subject: [Freeswitch-users] Fwd: FS compile errors on Solaris 10 sparc In-Reply-To: References: Message-ID: Try filing this bug in the JIRA and if possible grant access to the Fs devs for this box as they mainly work on the Intel arch for fs. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/f7ac94d7/attachment.html From fraserredmond at gmail.com Wed Nov 30 02:30:10 2011 From: fraserredmond at gmail.com (Fraser Redmond) Date: Tue, 29 Nov 2011 18:30:10 -0500 Subject: [Freeswitch-users] xml_curl directory - doing authentication in cgi, how to recreate user's password? In-Reply-To: References: Message-ID: Thanks Vitalie, but the problem is that I don't have the password to concatenate into the string - I only have the result of a one-way hash. The equivalent would be I had a file and did an md5 hash on it, I now have the md5 hash, but don't have the original file. Freeswitch is doing a different type of hash on the password entered by the user into the phone, and sending the result of that, and now I have the different results of the two different algorithm's to compare, which will always be different, even though they were both based on the same input (the password string.) So what I need is for Freeswitch to send the original password through instead of only sending several different hashes of it. I tried sending sip_auth_password through using enable-post-var, but it's not available - can that be added to the possible variables? (sip_auth_username does work) Cheers, Fraser On 29 November 2011 17:29, Vitalie Colosov wrote: > This might solve your problem: > > http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#a1-hash > > In short, you should hash not only the "password", but the concatenation > of "username:domain:password" > > Then use xml_curl to return this hashed value and FS will do > the authentication for you. > > Please let me know if this helps. > > Regards, > Vitalie > > > 2011/11/29 Fraser Redmond > >> Thanks Randy... but I think either I don't understand you, or you don't >> understand me... >> >> The password stored in the database has been hashed using mysql's ENCRYPT >> function with a seed (because it's not good security policy to store a >> password in any recoverable format.) >> >> I think you're saying that the nonce is also a hashed version of the >> password that also can't be reverted back to the original password - is >> that right? >> >> Which means that I now have two hashes which have been generated using >> different methods, so there's no way to compare them - cant compare within >> the cgi, and can't send the Freeswitch format back for Freeswitch to >> compare. >> >> If that's the case (and I'd still like to be clear on that), is it >> possible to pass through the password in addition? (I'll be using https, so >> sending without hashing is ok.) >> >> Cheers, >> Fraser >> >> >> >> >> >> On 28 November 2011 23:59, Rendy wrote: >> >>> Hi, >>> Why don't you let your user authenticate using hashed password then in >>> php you return the user xml with the hashed password that is stored. >>> In that way, you will not have any issue. I don't think you can >>> rebuild the original password as what hash function is meant to be one >>> way only. >>> >>> >>> On Tue, Nov 29, 2011 at 11:45 AM, Fraser Redmond >>> wrote: >>> > I am setting up a connection to a database of users, whose passwords >>> have >>> > been saved as a one-way hash. >>> > That means that my xml_curl php/sql will need to perform the >>> authentication, >>> > and return a user without any password. >>> > (According to Anthony, back in >>> > 2008: >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/029882.html >>> ) >>> > Only thing is I can't find any mention anywhere of how to re-generate >>> the >>> > user's password from the sip_auth variables in order to run it through >>> my >>> > one-way hash for comparison to the database. >>> > It's got to be something to do with these: >>> > sip_auth_nonce = 4d95dd9f-2247-474a-8496-aa7c08700fe7 >>> > sip_auth_cnonce = a088c6b6ba18d1387a45998b6bfa842d >>> > sip_auth_nc = 0000000a >>> > sip_auth_response = 9edefab216a46ed75f1ed1297dd9c9d3 >>> > Any ideas how to rebuild the original user's password? >>> > Or is there a way to send the password through as part of the post? >>> (maybe >>> > using enable-post-var) >>> > Cheers, >>> > Fraser >>> > >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/b0b57764/attachment.html From vetali100 at gmail.com Wed Nov 30 03:03:05 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Tue, 29 Nov 2011 16:03:05 -0800 Subject: [Freeswitch-users] xml_curl directory - doing authentication in cgi, how to recreate user's password? In-Reply-To: References: Message-ID: My suggestion is based on the assumption that you can implement md5 hash method to store users credentials in the database. If so, then you can read further. When a user registers, Freeswitch is doing md5 on the following value - username:domain:password Domain can be hardcoded to any value in the internal profile: So, when a user will register, your xml_curl will need to respond with the value of a1-hash: When you create a user 1001, which has password 12345, you will need to store in the database the md5 hash value of the following string: 1001:some_domain:12345 In php you can do it by using - md5($user.":some_domain:".$password) Obviously, if you need to hash only the password (without user id and domain) than this approach will not work for you. However I don't see why you can't hash the whole string. Regards, Vitalie 2011/11/29 Fraser Redmond > Thanks Vitalie, but the problem is that I don't have the password to > concatenate into the string - I only have the result of a one-way hash. > > The equivalent would be I had a file and did an md5 hash on it, I now have > the md5 hash, but don't have the original file. Freeswitch is doing a > different type of hash on the password entered by the user into the phone, > and sending the result of that, and now I have the different results of the > two different algorithm's to compare, which will always be different, even > though they were both based on the same input (the password string.) > > So what I need is for Freeswitch to send the original password through > instead of only sending several different hashes of it. > > I tried sending sip_auth_password through using enable-post-var, but it's > not available - can that be added to the possible variables? (sip_auth_username > does work) > > Cheers, > Fraser > > > > > > On 29 November 2011 17:29, Vitalie Colosov wrote: > >> This might solve your problem: >> >> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#a1-hash >> >> In short, you should hash not only the "password", but the concatenation >> of "username:domain:password" >> >> Then use xml_curl to return this hashed value and FS will do >> the authentication for you. >> >> Please let me know if this helps. >> >> Regards, >> Vitalie >> >> >> 2011/11/29 Fraser Redmond >> >>> Thanks Randy... but I think either I don't understand you, or you don't >>> understand me... >>> >>> The password stored in the database has been hashed using mysql's >>> ENCRYPT function with a seed (because it's not good security policy to >>> store a password in any recoverable format.) >>> >>> I think you're saying that the nonce is also a hashed version of the >>> password that also can't be reverted back to the original password - is >>> that right? >>> >>> Which means that I now have two hashes which have been generated using >>> different methods, so there's no way to compare them - cant compare within >>> the cgi, and can't send the Freeswitch format back for Freeswitch to >>> compare. >>> >>> If that's the case (and I'd still like to be clear on that), is it >>> possible to pass through the password in addition? (I'll be using https, so >>> sending without hashing is ok.) >>> >>> Cheers, >>> Fraser >>> >>> >>> >>> >>> >>> On 28 November 2011 23:59, Rendy wrote: >>> >>>> Hi, >>>> Why don't you let your user authenticate using hashed password then in >>>> php you return the user xml with the hashed password that is stored. >>>> In that way, you will not have any issue. I don't think you can >>>> rebuild the original password as what hash function is meant to be one >>>> way only. >>>> >>>> >>>> On Tue, Nov 29, 2011 at 11:45 AM, Fraser Redmond >>>> wrote: >>>> > I am setting up a connection to a database of users, whose passwords >>>> have >>>> > been saved as a one-way hash. >>>> > That means that my xml_curl php/sql will need to perform the >>>> authentication, >>>> > and return a user without any password. >>>> > (According to Anthony, back in >>>> > 2008: >>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/029882.html >>>> ) >>>> > Only thing is I can't find any mention anywhere of how to re-generate >>>> the >>>> > user's password from the sip_auth variables in order to run it >>>> through my >>>> > one-way hash for comparison to the database. >>>> > It's got to be something to do with these: >>>> > sip_auth_nonce = 4d95dd9f-2247-474a-8496-aa7c08700fe7 >>>> > sip_auth_cnonce = a088c6b6ba18d1387a45998b6bfa842d >>>> > sip_auth_nc = 0000000a >>>> > sip_auth_response = 9edefab216a46ed75f1ed1297dd9c9d3 >>>> > Any ideas how to rebuild the original user's password? >>>> > Or is there a way to send the password through as part of the post? >>>> (maybe >>>> > using enable-post-var) >>>> > Cheers, >>>> > Fraser >>>> > >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/fe1c6ced/attachment-0001.html From fraserredmond at gmail.com Wed Nov 30 03:22:04 2011 From: fraserredmond at gmail.com (Fraser Redmond) Date: Tue, 29 Nov 2011 19:22:04 -0500 Subject: [Freeswitch-users] xml_curl directory - doing authentication in cgi, how to recreate user's password? In-Reply-To: References: Message-ID: Thanks Vitalie, you're right that would work. Unfortunately, I'm dealing with an existing set of users, and resetting everyone's passwords so as to hash them differently isn't going to be an option. I'm hoping that one of the freeswitch developers can make it possible to access the original sip_auth_password value (or a param in the xml_curl conf to tell it to send the password as plain text.) Cheers, Fraser On 29 November 2011 19:03, Vitalie Colosov wrote: > My suggestion is based on the assumption that you can implement md5 hash > method to store users credentials in the database. If so, then you can read > further. > > When a user registers, Freeswitch is doing md5 on the following value - > username:domain:password > > Domain can be hardcoded to any value in the internal profile: > > > > > > So, when a user will register, your xml_curl will need to respond with the > value of a1-hash: > > > > > > When you create a user 1001, which has password 12345, you will need to > store in the database the md5 hash value of the following > string: 1001:some_domain:12345 > > In php you can do it by using - md5($user.":some_domain:".$password) > > Obviously, if you need to hash only the password (without user id and > domain) than this approach will not work for you. > However I don't see why you can't hash the whole string. > > > Regards, > Vitalie > > > > > 2011/11/29 Fraser Redmond > >> Thanks Vitalie, but the problem is that I don't have the password to >> concatenate into the string - I only have the result of a one-way hash. >> >> The equivalent would be I had a file and did an md5 hash on it, I now >> have the md5 hash, but don't have the original file. Freeswitch is doing a >> different type of hash on the password entered by the user into the phone, >> and sending the result of that, and now I have the different results of the >> two different algorithm's to compare, which will always be different, even >> though they were both based on the same input (the password string.) >> >> So what I need is for Freeswitch to send the original password through >> instead of only sending several different hashes of it. >> >> I tried sending sip_auth_password through using enable-post-var, but >> it's not available - can that be added to the possible variables? (sip_auth_username >> does work) >> >> Cheers, >> Fraser >> >> >> >> >> >> On 29 November 2011 17:29, Vitalie Colosov wrote: >> >>> This might solve your problem: >>> >>> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#a1-hash >>> >>> In short, you should hash not only the "password", but the concatenation >>> of "username:domain:password" >>> >>> Then use xml_curl to return this hashed value and FS will do >>> the authentication for you. >>> >>> Please let me know if this helps. >>> >>> Regards, >>> Vitalie >>> >>> >>> 2011/11/29 Fraser Redmond >>> >>>> Thanks Randy... but I think either I don't understand you, or you don't >>>> understand me... >>>> >>>> The password stored in the database has been hashed using mysql's >>>> ENCRYPT function with a seed (because it's not good security policy to >>>> store a password in any recoverable format.) >>>> >>>> I think you're saying that the nonce is also a hashed version of the >>>> password that also can't be reverted back to the original password - is >>>> that right? >>>> >>>> Which means that I now have two hashes which have been generated using >>>> different methods, so there's no way to compare them - cant compare within >>>> the cgi, and can't send the Freeswitch format back for Freeswitch to >>>> compare. >>>> >>>> If that's the case (and I'd still like to be clear on that), is it >>>> possible to pass through the password in addition? (I'll be using https, so >>>> sending without hashing is ok.) >>>> >>>> Cheers, >>>> Fraser >>>> >>>> >>>> >>>> >>>> >>>> On 28 November 2011 23:59, Rendy wrote: >>>> >>>>> Hi, >>>>> Why don't you let your user authenticate using hashed password then in >>>>> php you return the user xml with the hashed password that is stored. >>>>> In that way, you will not have any issue. I don't think you can >>>>> rebuild the original password as what hash function is meant to be one >>>>> way only. >>>>> >>>>> >>>>> On Tue, Nov 29, 2011 at 11:45 AM, Fraser Redmond >>>>> wrote: >>>>> > I am setting up a connection to a database of users, whose passwords >>>>> have >>>>> > been saved as a one-way hash. >>>>> > That means that my xml_curl php/sql will need to perform the >>>>> authentication, >>>>> > and return a user without any password. >>>>> > (According to Anthony, back in >>>>> > 2008: >>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/029882.html >>>>> ) >>>>> > Only thing is I can't find any mention anywhere of how to >>>>> re-generate the >>>>> > user's password from the sip_auth variables in order to run it >>>>> through my >>>>> > one-way hash for comparison to the database. >>>>> > It's got to be something to do with these: >>>>> > sip_auth_nonce = 4d95dd9f-2247-474a-8496-aa7c08700fe7 >>>>> > sip_auth_cnonce = a088c6b6ba18d1387a45998b6bfa842d >>>>> > sip_auth_nc = 0000000a >>>>> > sip_auth_response = 9edefab216a46ed75f1ed1297dd9c9d3 >>>>> > Any ideas how to rebuild the original user's password? >>>>> > Or is there a way to send the password through as part of the post? >>>>> (maybe >>>>> > using enable-post-var) >>>>> > Cheers, >>>>> > Fraser >>>>> > >>>>> >>>> >>>> >>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/1ed496b8/attachment.html From rendyfrx at gmail.com Wed Nov 30 03:26:13 2011 From: rendyfrx at gmail.com (Rendy) Date: Wed, 30 Nov 2011 08:26:13 +0800 Subject: [Freeswitch-users] xml_curl directory - doing authentication in cgi, how to recreate user's password? In-Reply-To: References: Message-ID: Hi Fraser, What I mean is like this, when user trying to authenticate says via your application, can you hashed the password in the same manner before sending to Freeswitch (says MD5)? If yes, then in your php, you should return the XML with hashed user password that you retrieve from DB and let Freeswitch compare for you. You do not need to compare yourself. Hope I understand your problem correctly and this can solved it :) On Wed, Nov 30, 2011 at 12:19 AM, Fraser Redmond wrote: > Thanks Randy... but I think either I don't understand you, or you don't > understand me... > > The password stored in the database has been hashed using mysql's ENCRYPT > function with a seed (because it's not good security policy to store a > password in any recoverable format.) > > I think you're saying that the nonce is also a hashed version of the > password that also can't be reverted back to the original password - is that > right? > > Which means that I now have two hashes which have been generated using > different methods, so there's no way to compare them - cant compare within > the cgi, and can't send the Freeswitch format back for Freeswitch to > compare. > > If that's the case (and I'd still like to be clear on that), is it possible > to pass through the password in addition? (I'll be using https, so sending > without hashing is ok.) > > Cheers, > Fraser > > > > > > On 28 November 2011 23:59, Rendy wrote: >> >> Hi, >> Why don't you let your user authenticate using hashed password then in >> php you return the user xml with the hashed password that is stored. >> In that way, you will not have any issue. I don't think you can >> rebuild the original password as what hash function is meant to be one >> way only. >> >> >> On Tue, Nov 29, 2011 at 11:45 AM, Fraser Redmond >> wrote: >> > I am setting up a connection to a database of users, whose passwords >> > have >> > been saved as a one-way hash. >> > That means that my xml_curl php/sql will need to perform the >> > authentication, >> > and return a user without any password. >> > (According to Anthony, back in >> > >> > 2008:?http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/029882.html?) >> > Only thing is I can't find any mention anywhere of how to re-generate >> > the >> > user's password from the sip_auth variables in order to run it through >> > my >> > one-way hash for comparison to the database. >> > It's got to be something to do with these: >> > sip_auth_nonce = 4d95dd9f-2247-474a-8496-aa7c08700fe7 >> > sip_auth_cnonce = a088c6b6ba18d1387a45998b6bfa842d >> > sip_auth_nc = 0000000a >> > sip_auth_response = 9edefab216a46ed75f1ed1297dd9c9d3 >> > Any ideas how to rebuild the original user's password? >> > Or is there a way to send the password through as part of the post? >> > (maybe >> > using?enable-post-var) >> > Cheers, >> > Fraser >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 fraserredmond at gmail.com Wed Nov 30 03:33:31 2011 From: fraserredmond at gmail.com (Fraser Redmond) Date: Tue, 29 Nov 2011 19:33:31 -0500 Subject: [Freeswitch-users] xml_curl directory - doing authentication in cgi, how to recreate user's password? In-Reply-To: References: Message-ID: Thanks Rendy, but they're hashed differently, so I can't return the hash from the database, as it wouldn't match with the hash generated by Freeswitch. It really looks like I need to get Freeswitch to send the original password string to the cgi/application. Cheers, Fraser On 29 November 2011 19:26, Rendy wrote: > Hi Fraser, > What I mean is like this, when user trying to authenticate says via > your application, can you hashed the password in the same manner > before sending to Freeswitch (says MD5)? If yes, then in your php, you > should return the XML with hashed user password that you retrieve from > DB and let Freeswitch compare for you. You do not need to compare > yourself. > > Hope I understand your problem correctly and this can solved it :) > > > On Wed, Nov 30, 2011 at 12:19 AM, Fraser Redmond > wrote: > > Thanks Randy... but I think either I don't understand you, or you don't > > understand me... > > > > The password stored in the database has been hashed using mysql's ENCRYPT > > function with a seed (because it's not good security policy to store a > > password in any recoverable format.) > > > > I think you're saying that the nonce is also a hashed version of the > > password that also can't be reverted back to the original password - is > that > > right? > > > > Which means that I now have two hashes which have been generated using > > different methods, so there's no way to compare them - cant compare > within > > the cgi, and can't send the Freeswitch format back for Freeswitch to > > compare. > > > > If that's the case (and I'd still like to be clear on that), is it > possible > > to pass through the password in addition? (I'll be using https, so > sending > > without hashing is ok.) > > > > Cheers, > > Fraser > > > > > > > > > > > > On 28 November 2011 23:59, Rendy wrote: > >> > >> Hi, > >> Why don't you let your user authenticate using hashed password then in > >> php you return the user xml with the hashed password that is stored. > >> In that way, you will not have any issue. I don't think you can > >> rebuild the original password as what hash function is meant to be one > >> way only. > >> > >> > >> On Tue, Nov 29, 2011 at 11:45 AM, Fraser Redmond > >> wrote: > >> > I am setting up a connection to a database of users, whose passwords > >> > have > >> > been saved as a one-way hash. > >> > That means that my xml_curl php/sql will need to perform the > >> > authentication, > >> > and return a user without any password. > >> > (According to Anthony, back in > >> > > >> > 2008: > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/029882.html > ) > >> > Only thing is I can't find any mention anywhere of how to re-generate > >> > the > >> > user's password from the sip_auth variables in order to run it through > >> > my > >> > one-way hash for comparison to the database. > >> > It's got to be something to do with these: > >> > sip_auth_nonce = 4d95dd9f-2247-474a-8496-aa7c08700fe7 > >> > sip_auth_cnonce = a088c6b6ba18d1387a45998b6bfa842d > >> > sip_auth_nc = 0000000a > >> > sip_auth_response = 9edefab216a46ed75f1ed1297dd9c9d3 > >> > Any ideas how to rebuild the original user's password? > >> > Or is there a way to send the password through as part of the post? > >> > (maybe > >> > using enable-post-var) > >> > Cheers, > >> > Fraser > >> > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111129/0c3eb21a/attachment-0001.html From rendyfrx at gmail.com Wed Nov 30 05:29:18 2011 From: rendyfrx at gmail.com (Rendy) Date: Wed, 30 Nov 2011 10:29:18 +0800 Subject: [Freeswitch-users] xml_curl directory - doing authentication in cgi, how to recreate user's password? In-Reply-To: References: Message-ID: Hi Fraser, As pointed out in the link from Vitalie, you can use Basic type, not a1-hash, so there is no hashing by Freeswitch. However, if you cannot rehash user password input to the same like in your DB, then you will need to get Freeswitch to send over the password to your php. Anyway, I would like to know also how to get Freeswitch to send over the password and return XML format respectively. On Wed, Nov 30, 2011 at 8:33 AM, Fraser Redmond wrote: > Thanks Rendy, but they're hashed differently, so I can't return the hash > from the database, as it wouldn't match with the hash generated by > Freeswitch. > It really looks like I need to get Freeswitch to send the original password > string to the cgi/application. > Cheers, > Fraser > > > > > On 29 November 2011 19:26, Rendy wrote: >> >> Hi Fraser, >> What I mean is like this, when user trying to authenticate says via >> your application, can you hashed the password in the same manner >> before sending to Freeswitch (says MD5)? If yes, then in your php, you >> should return the XML with hashed user password that you retrieve from >> DB and let Freeswitch compare for you. You do not need to compare >> yourself. >> >> Hope I understand your problem correctly and this can solved it :) >> >> >> On Wed, Nov 30, 2011 at 12:19 AM, Fraser Redmond >> wrote: >> > Thanks Randy... but I think either I don't understand you, or you don't >> > understand me... >> > >> > The password stored in the database has been hashed using mysql's >> > ENCRYPT >> > function with a seed (because it's not good security policy to store a >> > password in any recoverable format.) >> > >> > I think you're saying that the nonce is also a hashed version of the >> > password that also can't be reverted back to the original password - is >> > that >> > right? >> > >> > Which means that I now have two hashes which have been generated using >> > different methods, so there's no way to compare them - cant compare >> > within >> > the cgi, and can't send the Freeswitch format back for Freeswitch to >> > compare. >> > >> > If that's the case (and I'd still like to be clear on that), is it >> > possible >> > to pass through the password in addition? (I'll be using https, so >> > sending >> > without hashing is ok.) >> > >> > Cheers, >> > Fraser >> > >> > >> > >> > >> > >> > On 28 November 2011 23:59, Rendy wrote: >> >> >> >> Hi, >> >> Why don't you let your user authenticate using hashed password then in >> >> php you return the user xml with the hashed password that is stored. >> >> In that way, you will not have any issue. I don't think you can >> >> rebuild the original password as what hash function is meant to be one >> >> way only. >> >> >> >> >> >> On Tue, Nov 29, 2011 at 11:45 AM, Fraser Redmond >> >> wrote: >> >> > I am setting up a connection to a database of users, whose passwords >> >> > have >> >> > been saved as a one-way hash. >> >> > That means that my xml_curl php/sql will need to perform the >> >> > authentication, >> >> > and return a user without any password. >> >> > (According to Anthony, back in >> >> > >> >> > >> >> > 2008:?http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/029882.html?) >> >> > Only thing is I can't find any mention anywhere of how to re-generate >> >> > the >> >> > user's password from the sip_auth variables in order to run it >> >> > through >> >> > my >> >> > one-way hash for comparison to the database. >> >> > It's got to be something to do with these: >> >> > sip_auth_nonce = 4d95dd9f-2247-474a-8496-aa7c08700fe7 >> >> > sip_auth_cnonce = a088c6b6ba18d1387a45998b6bfa842d >> >> > sip_auth_nc = 0000000a >> >> > sip_auth_response = 9edefab216a46ed75f1ed1297dd9c9d3 >> >> > Any ideas how to rebuild the original user's password? >> >> > Or is there a way to send the password through as part of the post? >> >> > (maybe >> >> > using?enable-post-var) >> >> > Cheers, >> >> > Fraser >> >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.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 >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 ddalton at shortbus.net Wed Nov 30 08:38:33 2011 From: ddalton at shortbus.net (Doug Dalton) Date: Tue, 29 Nov 2011 21:38:33 -0800 Subject: [Freeswitch-users] Asterisk TDM410 card and windows Message-ID: Does the asterisk TDM410 card work with windows 7 and freeswitch? Regards, Doug From farhan.husain at csebuet.org Wed Nov 30 10:14:23 2011 From: farhan.husain at csebuet.org (Farhan Husain) Date: Tue, 29 Nov 2011 23:14:23 -0800 Subject: [Freeswitch-users] Build error: undefined reference to `Curl_setopt' Message-ID: Hi all, I am trying to build the latest source from Git in an Ubuntu machine. The error I am getting is this: Making all in . quiet_libtool: link: gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DENABLE_SRTP -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -o .libs/freeswitch freeswitch-switch.o /usr/lib/libcurl.so -lm -lz ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -luuid -lrt -ldl -lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lncurses -pthread -Wl,-rpath -Wl,/usr/local/freeswitch/lib ./.libs/libfreeswitch.so: undefined reference to `Curl_setopt' collect2: ld returned 1 exit status make[2]: *** [freeswitch] Error 1 I have installed php-curl (sudo apt-get install php5-curl) but to no avail. Does anyone have any solution to this? Thanks, Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111129/36b2cad9/attachment.html From thomas at chaschperli.ch Wed Nov 30 11:52:08 2011 From: thomas at chaschperli.ch (Thomas Mueller) Date: Wed, 30 Nov 2011 09:52:08 +0100 Subject: [Freeswitch-users] Build error: undefined reference to `Curl_setopt' In-Reply-To: References: Message-ID: <4ED5EEB8.9030709@chaschperli.ch> On 30.11.2011 08:14, Farhan Husain wrote: > Hi all, > > I am trying to build the latest source from Git in an Ubuntu machine. > The error I am getting is this: > > Making all in . > quiet_libtool: link: gcc -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -g -O2 -pthread -DLINUX=2 -D_REENTRANT > -D_GNU_SOURCE -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DENABLE_SRTP > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -o .libs/freeswitch > freeswitch-switch.o /usr/lib/libcurl.so -lm -lz > ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -luuid -lrt -ldl > -lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto > -lncurses -pthread -Wl,-rpath -Wl,/usr/local/freeswitch/lib > > ./.libs/libfreeswitch.so: undefined reference to `Curl_setopt' > collect2: ld returned 1 exit status > make[2]: *** [freeswitch] Error 1 > > I have installed php-curl (sudo apt-get install php5-curl) but to no > avail. Does anyone have any solution to this? seems like the curl dev package is missing ( libcurl4-dev or libcurl3-dev) -> sudo aptitude install libcurl4-dev - Thomas From avi at avimarcus.net Wed Nov 30 15:10:10 2011 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 30 Nov 2011 14:10:10 +0200 Subject: [Freeswitch-users] Timezone with time conditions In-Reply-To: References: <91EA0D35-FF57-4A97-AA16-C6693B51BDA7@5ninesolutions.com> Message-ID: Thanks Anthony! I update the wiki at http://wiki.freeswitch.org/wiki/Tod if anyone else can test and confirm this feature (on git > 2011-11-08 / revision 65a756643a45a9462270a83238a396d742b7103f) The TOD doc page could be streamlined a bit, too.. -Avi On Tue, Nov 8, 2011 at 12:39 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I looked into what the code would require and decided to push a patch, now > you get to test and document it. > > if you get the variable tod_tz_offset set on the channel before it hits > the dialplan extension in question it will apply that offset to gmt so you > can set it to say, -6 for central time. > > Note, you must have it set first so you either need to use inline set or > maybe it will work if you have already defined it in your user directory. > > also on a per-tag basis, you can do tz-offset attr ... > > > > > On Mon, Nov 7, 2011 at 4:20 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> on multi home you are better off exploring xml_curl to do time of day >> stuff, what you suggest would require some new code for sure. >> >> >> On Mon, Nov 7, 2011 at 4:10 PM, Michael Collins wrote: >> >>> >>> On Mon, Nov 7, 2011 at 11:37 AM, Avi Marcus wrote: >>> >>>> Note: The dialplan TOD conditions do not account for TZ >>>> >>>> I think that's the point of this thread. Is there a point to re-invent >>>> the wheel doing TOD conditions manually? >>>> >>> >>> No, the issue is a matter of supply and demand. The skills needed to add >>> TZ directly into the dialplan are in short supply, whereas doing TZ stuff >>> manually in the dialplan is all but trivial and does not require C coding >>> skills. Also, the demand for TZ checking is relatively low. >>> >>> If it's really desired then a bounty should be put up. >>> >>> -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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111130/d512a66b/attachment-0001.html From boris at tagnet.ru Wed Nov 30 16:10:18 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Wed, 30 Nov 2011 19:10:18 +0600 Subject: [Freeswitch-users] T.38 is broken again Message-ID: <4ED62B3A.1010002@tagnet.ru> Hello! After some updates to FreeSwitch I found that T.38 faxes are broken again. My test enviroments are: Audiocodes MP114 <---> FreeSwitch <--> Cisco 5350 <--> PSTN Kapanga softphone <---> FreeSwitch <--> AudioCodes MP-114 I see that Kapanga softphone reports "Rejecting fax call. Enable fax ReINVITE to send faxes". I haven't changed Kapanga or Audiocodes configurations. Only upgraded Freeswitch to the latest git (FreeSWITCH Version 1.0.head (git-fafcc46 2011-11-30 15-19-34 +0800). Please help, faxes it very important to use. Please ask for more information as needed. -- Regards, Boris From boris at tagnet.ru Wed Nov 30 16:22:45 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Wed, 30 Nov 2011 19:22:45 +0600 Subject: [Freeswitch-users] T.38 is broken again In-Reply-To: <4ED62B3A.1010002@tagnet.ru> References: <4ED62B3A.1010002@tagnet.ru> Message-ID: <4ED62E25.6010302@tagnet.ru> Hello! Forgot to add - I use t38-passthru=true in profile. Have tried "once" without success. > Hello! > > After some updates to FreeSwitch I found that T.38 faxes are broken > again. My test enviroments are: > Audiocodes MP114<---> FreeSwitch<--> Cisco 5350<--> PSTN > Kapanga softphone<---> FreeSwitch<--> AudioCodes MP-114 > > I see that Kapanga softphone reports "Rejecting fax call. Enable fax > ReINVITE to send faxes". I haven't changed Kapanga or Audiocodes > configurations. Only upgraded Freeswitch to the latest git (FreeSWITCH > Version 1.0.head (git-fafcc46 2011-11-30 15-19-34 +0800). Please help, > faxes it very important to use. Please ask for more information as needed. > > -- Regards, Boris From boris at tagnet.ru Wed Nov 30 17:47:39 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Wed, 30 Nov 2011 20:47:39 +0600 Subject: [Freeswitch-users] T.38 is broken again In-Reply-To: <4ED62E25.6010302@tagnet.ru> References: <4ED62B3A.1010002@tagnet.ru> <4ED62E25.6010302@tagnet.ru> Message-ID: <4ED6420B.9000901@tagnet.ru> Oops, found that there was a "Do not send faxes" in Kapanga selected. Sorry, need to do more tests to confirm if bug with faxes is really present. > Hello! > > Forgot to add - I use t38-passthru=true in profile. Have tried > "once" without success. > >> Hello! >> >> After some updates to FreeSwitch I found that T.38 faxes are broken >> again. My test enviroments are: >> Audiocodes MP114<---> FreeSwitch<--> Cisco 5350<--> PSTN >> Kapanga softphone<---> FreeSwitch<--> AudioCodes MP-114 >> >> I see that Kapanga softphone reports "Rejecting fax call. Enable fax >> ReINVITE to send faxes". I haven't changed Kapanga or Audiocodes >> configurations. Only upgraded Freeswitch to the latest git (FreeSWITCH >> Version 1.0.head (git-fafcc46 2011-11-30 15-19-34 +0800). Please help, >> faxes it very important to use. Please ask for more information as needed. >> >> > -- Regards, Boris From brian at freeswitch.org Wed Nov 30 18:08:10 2011 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Nov 2011 09:08:10 -0600 Subject: [Freeswitch-users] T.38 is broken again In-Reply-To: <4ED6420B.9000901@tagnet.ru> References: <4ED62B3A.1010002@tagnet.ru> <4ED62E25.6010302@tagnet.ru> <4ED6420B.9000901@tagnet.ru> Message-ID: Yep was just about to call BULLSHIT on this. :P It works FLAWLESSLY now that we have fixed a few lingering issues that didn't always present itself in every scenario. /b On Nov 30, 2011, at 8:47 AM, Boris Kovalenko wrote: > Oops, found that there was a "Do not send faxes" in Kapanga selected. > Sorry, need to do more tests to confirm if bug with faxes is really present. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111130/b55b4962/attachment.html From admin at blindi.net Wed Nov 30 15:22:45 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Wed, 30 Nov 2011 13:22:45 +0100 (CET) Subject: [Freeswitch-users] Question Answertime variable? Message-ID: Hi all, Is there a possibility to determine the caller time for a channel at the end of a call? For example: in asterisk exists a variable: ${ANSWEREDTIME} I like to store the Answertime in a Database for example: Dailly minutes to call a Cellphone. Can your help plese? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From curriegrad2004 at gmail.com Wed Nov 30 18:14:24 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 30 Nov 2011 07:14:24 -0800 Subject: [Freeswitch-users] Asterisk TDM410 card and windows In-Reply-To: References: Message-ID: Short answer: No On Tue, Nov 29, 2011 at 9:38 PM, Doug Dalton wrote: > Does the asterisk TDM410 card work with windows 7 and freeswitch? > > Regards, > Doug > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 curriegrad2004 at gmail.com Wed Nov 30 18:15:27 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 30 Nov 2011 07:15:27 -0800 Subject: [Freeswitch-users] Question Answertime variable? In-Reply-To: References: Message-ID: billseconds or bill_msec would be the one to turn to. On Wed, Nov 30, 2011 at 4:22 AM, Thomas Hoellriegel wrote: > Hi all, > Is there a possibility to determine the caller time for a channel at > the end of a call? > > For example: in asterisk exists a variable: > ${ANSWEREDTIME} > I like to store the Answertime in a Database for example: > Dailly minutes to call a Cellphone. > Can your help plese? > Thanks. > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 admin at blindi.net Wed Nov 30 18:30:42 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Wed, 30 Nov 2011 16:30:42 +0100 (CET) Subject: [Freeswitch-users] Problem send_dtmf send only to a-leg In-Reply-To: References: Message-ID: Hi all, i have a acoount on a phonebankingservice. If I send the characters, then I only hear the sounds coming. But not the Phone Banking Service here is my extension: Can you see what is wrong please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From admin at blindi.net Wed Nov 30 18:37:15 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Wed, 30 Nov 2011 16:37:15 +0100 (CET) Subject: [Freeswitch-users] Question Answertime variable? In-Reply-To: References: Message-ID: Hi all, Is there a possibility to determine the caller time for a channel at the end of a call? For example: in asterisk exists a variable name: ${ANSWEREDTIME} I like to store the Answertime in a Database for example: Dailly minutes to call a Cellphone. Can your help plese? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From msanitariusz at gmail.com Wed Nov 30 17:13:19 2011 From: msanitariusz at gmail.com (Mariusz Sanitariusz) Date: Wed, 30 Nov 2011 15:13:19 +0100 Subject: [Freeswitch-users] Run few script commands after the user answers the call (during that call) Message-ID: Hi, I'm writing a script (in Python, but I'll be grateful for examples in any language) processing incoming calls and I'm stuck at the final point of my script. Simply, I don't have any idea how to achieve something like this: - at the end of my script I have to call (bridge, originate, forgive me my lack of knowledge) a group of users (one at the time and after the specified timeout, if that user doesn't answer, try another user from that group of users and so on) and when an user answers a call I have to get an information on which user answered and pass that information via HTTP request to a web service and it has to be done at the moment of some user answering, not after the call has ended The problem is that I don't know how to get information about which user has answered the phone and make a HTTP request just after the call has been answered. Could you please provide me witch some hints or code samples on how to achieve this behavior. Best regards Mariusz Sanitariusz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111130/0e9cee01/attachment.html From msc at freeswitch.org Wed Nov 30 19:14:49 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Nov 2011 08:14:49 -0800 Subject: [Freeswitch-users] Run few script commands after the user answers the call (during that call) In-Reply-To: References: Message-ID: Look at execute_on_answer channel variable: http://wiki.freeswitch.org/wiki/Channel_Variables#The_execute_on_family -MC On Wed, Nov 30, 2011 at 6:13 AM, Mariusz Sanitariusz wrote: > Hi, > I'm writing a script (in Python, but I'll be grateful for examples in any > language) processing incoming calls and I'm stuck at the final point of my > script. > Simply, I don't have any idea how to achieve something like this: > - at the end of my script I have to call (bridge, originate, forgive me my > lack of knowledge) a group of users (one at the time and after the > specified timeout, if that user doesn't answer, try another user from that > group of users and so on) and when an user answers a call I have to get an > information on which user answered and pass that information via HTTP > request to a web service and it has to be done at the moment of some user > answering, not after the call has ended > The problem is that I don't know how to get information about which user > has answered the phone and make a HTTP request just after the call has been > answered. > Could you please provide me witch some hints or code samples on how to > achieve this behavior. > Best regards > Mariusz Sanitariusz > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111130/f4cf71bc/attachment-0001.html From msc at freeswitch.org Wed Nov 30 19:16:46 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Nov 2011 08:16:46 -0800 Subject: [Freeswitch-users] Problem send_dtmf send only to a-leg In-Reply-To: References: Message-ID: You are sending the dtmf not queueing the dtmf. Try "queue_dtmf" instead of "send_dtmf". -MC On Wed, Nov 30, 2011 at 7:30 AM, Thomas Hoellriegel wrote: > Hi all, i have a acoount on a phonebankingservice. > If I send the characters, then I only hear the sounds coming. But not > the Phone Banking Service > > here is my extension: > > > > > > > > > > Can you see what is wrong please? > thanks. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111130/ce8fd505/attachment.html From msc at freeswitch.org Wed Nov 30 19:21:31 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Nov 2011 08:21:31 -0800 Subject: [Freeswitch-users] Question Answertime variable? In-Reply-To: References: Message-ID: I think you'll find all sorts of useful variables on this page: http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Channel_Variables You can get the answer time as a timestamp, epoch seconds, epoch milliseconds, and epoch microseconds. What kind of CDRs are you using - CSV or XML? -MC On Wed, Nov 30, 2011 at 4:22 AM, Thomas Hoellriegel wrote: > Hi all, > Is there a possibility to determine the caller time for a channel at > the end of a call? > > For example: in asterisk exists a variable: > ${ANSWEREDTIME} > I like to store the Answertime in a Database for example: > Dailly minutes to call a Cellphone. > Can your help plese? > Thanks. > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111130/43be154c/attachment.html From daveh at beachdognet.com Wed Nov 30 19:48:24 2011 From: daveh at beachdognet.com (Dave Horton) Date: Wed, 30 Nov 2011 11:48:24 -0500 Subject: [Freeswitch-users] is 100rel still a bad idea? Message-ID: <40941536-3617-4EA4-8C24-B5B441B19CE9@beachdognet.com> I saw a few threads from a couple years back indicating the enabling reliable provisional responses could cause a FS crash. In fact, I am working with a customer who was told some time ago (by "freeswitch support", not sure exactly who they are referring to) to turn off 100rel (this while they were troubleshooting an unrelated issue). Just looking for an update on this, because I prefer to run my SIP gear with 100rel supported --- is there still a known bug (or even, suspected issue) with 100rel support on FS? From msc at freeswitch.org Wed Nov 30 20:12:21 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Nov 2011 09:12:21 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Don't forget: Jean-Marc Valin from Mozilla will be in to discuss the OPUS codec today! Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_30 Talk to you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111130/eebb7633/attachment.html From steve.kurzeja at gmail.com Wed Nov 30 21:21:17 2011 From: steve.kurzeja at gmail.com (Steve Kurzeja) Date: Thu, 1 Dec 2011 07:21:17 +1300 Subject: [Freeswitch-users] In Bypass Mode Freeswitch Changes SDP? In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D69633169C807C@VA3DIAXVS351.RED001.local> References: <2BF7FB90DF25EA4485949F3AF2B9D69633169C807C@VA3DIAXVS351.RED001.local> Message-ID: Tony would know for certain but from my experience inbound-bypass-media still does some calls to the sofia SOA engine, which may not be handling the multiple codec list in the answer. You could also try setting "enable-soa" to false. That should disable any SDP calls to sofia and pass the SDP through without any changes. On Thu, Nov 24, 2011 at 8:08 AM, Jerry Richards wrote: > Hello,**** > > ** ** > > I noticed an interop issue when using a Bria softphone and inbound-bypass-media > is true in the sip_profile. When answering an inbound call, the Bria > softphone's SDP Answer includes the list of codecs it supports (audio and > video). I noticed that, even in bypass media mode, Freeswitch filters out > all of the codecs in the list after the first one. I think this is not > right. According to RFC 3264 (Section 7 Offerer Processing of the Answer), > it is valid to provide a list of codecs in an SDP Answer and the endpoint > SHOULD use the first one (but it might not).**** > > ** ** > > Why is Freeswitch altering the SDP in the 200 OK? I think it should send > the SDP unmodified.**** > > ** ** > > Thanks,**** > > Jerry**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111201/19c46bfd/attachment.html From ifoundthetao at gmail.com Wed Nov 30 21:26:12 2011 From: ifoundthetao at gmail.com (Timothy Bolton) Date: Wed, 30 Nov 2011 12:26:12 -0600 Subject: [Freeswitch-users] NORTEL IP Phone 1535 with Video Message-ID: <4ED67544.3020503@gmail.com> Does anyone have any experience in setting up video to work with these Nortel phones? I am completely lost. I've searched the wiki and went to the IRC channels too. Any help would be glorious! -- 'We who cut mere stones must always be envisioning cathedrals.' Quarry Worker's Creed From curriegrad2004 at gmail.com Wed Nov 30 22:54:41 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 30 Nov 2011 11:54:41 -0800 Subject: [Freeswitch-users] NORTEL IP Phone 1535 with Video In-Reply-To: <4ED67544.3020503@gmail.com> References: <4ED67544.3020503@gmail.com> Message-ID: mod_h323 would be the place I'd start looking at for setting up for video on those nortel/avaya phones. Referring to the manufacturers docs is also strongly recommended. On 2011-11-30 11:23 AM, "Timothy Bolton" wrote: > Does anyone have any experience in setting up video to work with these > Nortel phones? > > I am completely lost. I've searched the wiki and went to the IRC > channels too. Any help would be glorious! > > -- > 'We who cut mere stones must always be envisioning cathedrals.' > Quarry Worker's Creed > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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/20111130/8e974721/attachment-0001.html